Asterisk 1.4.6, Copyright (C) 1999 - 2007 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 1.4.6 currently running on asterisk-price (pid = 10024) asterisk-price*CLI> Verbosity is at least 4 Core debug is at least 4 asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) asterisk-price*CLI> SIP/709-0982c110 (None) Up Bridged Call(Zap/6-1) asterisk-price*CLI> Zap/6-1 900@from-zaptel:6 Up Dial(SIP/701&SIP/703&SIP/704&S SIP/707-097a4ca0 202@parkedcalls:1 Up ParkedCall(202) Zap/4-1 (None) Up Bridged Call(SIP/705-09731328) SIP/705-09731328 6375737@from-interna Up Dial(ZAP/g0/6375737|120|rTt) Zap/2-1 900@from-zaptel:6 Up Bridged Call(SIP/707-097a4ca0) Zap/3-1 SIP/717@park-dial:1 Up Parked Call() 7 active channels 3 active calls asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> BYE sip:6375737@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKb23aa4b172d8fecfFrom: "Main Office Conf Room" ;tag=6811ac3a28ca079dTo: ;tag=as633d59d0Proxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:6375737@10.200.26.202", nonce="0215bd2a", response="ffd7bc71e304be2c2afe2ca479f67bf2"Call-ID: 25d07742f420b27b@10.200.26.105CSeq: 40913 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.200.26.105 : 5060 (NAT) <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKb23aa4b172d8fecf;received=10.200.26.105From: "Main Office Conf Room" ;tag=6811ac3a28ca079dTo: ;tag=as633d59d0Call-ID: 25d07742f420b27b@10.200.26.105CSeq: 40913 BYEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> -- Hungup 'Zap/4-1' == Spawn extension (from-internal, 6375737, 1) exited non-zero on 'SIP/705-09731328' -- Executing [h@from-internal:1] Hangup("SIP/705-09731328", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/705-09731328' asterisk-price*CLI> Really destroying SIP dialog '25d07742f420b27b@10.200.26.105' Method: BYE asterisk-price*CLI> -- Stopped music on hold on Zap/3-1 -- Added extension 'SIP/717' priority 1 to park-dial == Timeout for Zap/3-1 parked on 201. Returning to park-dial,SIP/717,1 asterisk-price*CLI> -- Executing [SIP/717@park-dial:1] Dial("Zap/3-1", "SIP/717||t") in new stack asterisk-price*CLI> Audio is at 10.200.26.202 port 10876 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.117:5060: INVITE sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1a124f6a;rportFrom: "WIRELESS CALLER" ;tag=as4ec3b36bTo: Contact: Call-ID: 4b560a236b52eb1b768e3c3a48222ddf@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:23 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10876 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.127:5060: NOTIFY sip:727@10.200.26.127:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK255414aa;rportFrom: ;tag=as0fac1be1To: "Conference Room" ;tag=1e94d3e74ab057e7Contact: Call-ID: cc74970ec691e7e1@10.200.26.127CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: active asterisk-price*CLI> Content-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 727 Reliably Transmitting (NAT) to 10.200.26.131:5060: NOTIFY sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK36f94faa;rportFrom: ;tag=as44d1397bT asterisk-price*CLI> o: "Lobby" ;tag=99f4161d5df43d88Contact: Call-ID: 62279987519ceb08@10.200.26.131CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 731 Reliably Transmitting (NAT) to 10.200.26.130:5060: NOTIFY sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4c9aac2b;rportFrom: ;tag=as73346657To: "Foyer" ;tag=bee43bdcf28e33c2Contact: Call-ID: 0c17b027f44a0b16@10.200.26.130CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 730 Reliably Transmitting (NAT) to 10.200.26.129:5060: NOTIFY sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK077877da;rportFrom: ;tag=as0b80a390To: "Will Jaimez" ;tag=18b4544c3b75f992Contact: Call-ID: 46071ba670da2a07@10.200.26.129CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 729 Reliably Transmitting (NAT) to 10.200.26.117:5060: NOTIFY sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK21a06a47;rportFrom: ;tag=as5ab9c51aTo: "Ken Williams" ;tag=85b710412fc42025Contact: Call-ID: d416754dbbee5bda@10.200.26.117CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 717 Reliably Transmitting (NAT) to 10.200.26.126:5060: NOTIFY sip:726@10.200.26.126:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2b8ea14f;rportFrom: ;tag=as3f56cc30To: "Bob Teny" ;tag=94c4908c7746b742Contact: Call-ID: 8e17d427fa4a6426@10.200.26.126CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 726 Reliably Transmitting (NAT) to 10.200.26.125:5060: NOTIFY sip:725@10.200.26.125:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31c53314;rportFrom: ;tag=as38f74437To: "Front Desk" ;tag=1bd437ccfd6ebe92Contact: Call-ID: 62473a6832fbae08@10.200.26.125CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 asterisk-price*CLI> terminated --- Extension Changed 201 new state Idle for Notify User 725 Reliably Transmitting (NAT) to 10.200.26.124:5060: NOTIFY sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK680275d2;rportFrom: ;tag=as35039685To: "Bobby Houston" ;tag=79c4178cb056f172Contact: Call-ID: 8d27b397d8fca0f7@10.200.26.124CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK36f94faa;rportFrom: ;tag=as44d1397bTo: "Lobby" ;tag=99f4161d5df43d88Call-ID: 62279987519ceb08@10.200.26.131CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Extension Changed 201 new state Idle for Notify User 724 <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4c9aac2b;rportFrom: ;tag=as73346657To: "Foyer" ;tag=bee43bdcf28e33c2Call-ID: 0c17b027f44a0b16@10.200.26.130CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Reliably Transmitting (NAT) to 10.200.26.121:5060: NOTIFY sip:721@10.200.26.121:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0f6dec03;rportFrom: ;tag=as52c9f68fTo: "Danny D'Ambrosio" ;tag=91e4dffc190f5ae1Contact: Call-ID: 8c47d4781d5ccab6@10.200.26.121CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK077877da;rportFrom: ;tag=as0b80a390To: "Will Jaimez" ;tag=18b4544c3b75f992Call-ID: 46071ba670da2a07@10.200.26.129CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Extension Changed 201 new state Idle for Notify User 721 Reliably Transmitting (NAT) to 10.200.26.122:5060: NOTIFY sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK292efbcd;rportFrom: ;tag=as3bd2c35dTo: "Jake Erramouspe" ;tag=d0e49efc170f18e1Contact: Call-ID: a94732785b5c49b6@10.200.26.122CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 722 <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2b8ea14f;rportFrom: ;tag=as3f56cc30To: "Bob Teny" ;tag=94c4908c7746b742Call-ID: 8e17d427fa4a6426@10.200.26.126CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31c53314;rportFrom: ;tag=as38f74437To: "Front Desk" ;tag=1bd437ccfd6ebe92Call-ID: 62473a6832fbae08@10.200.26.125CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Reliably Transmitting (NAT) to 10.200.26.120:5060: NOTIFY sip:720@10.200.26.120:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK52411c0d;rportFrom: ;tag=as26caf817To: "Darek Martinez" ;tag=1ae4770db0f4d3b8Contact: Call-ID: 0067f749305becc6@10.200.26.120CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 720 Reliably Transmitting (NAT) to 10.200.26.119:5060: NOTIFY sip:719@10.200.26.119:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1488ac85;rportFrom: ;tag=as4d16fad1To: "Randy Leffler" ;tag=9e94dba4740eb5d1Contact: Call-ID: 2f273697fcfc87f7@10.200.26.119CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK680275d2;rportFrom: ;tag=as35039685To: "Bobby Houston" ;tag=79c4178cb056f172Call-ID: 8d27b397d8fca0f7@10.200.26.124CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Extension Changed 201 new state Idle for Notify User 719 Reliably Transmitting (NAT) to 10.200.26.118:5060: NOTIFY sip:718@10.200.26.118:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6c8dd27e;rportFrom: ;tag=as501b143eTo: "Chris / Ed" ;tag=f9a4973cd07533c2Contact: Call-ID: 8647dd68f4fb0f08@10.200.26.118CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 718 Reliably Transmitting (NAT) to 10.200.26.116:5060: NOTIFY sip:716@10.200.26.116:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0bae62a5;rportFrom: ;tag=as71101529To: "Shop Kitchen" ;tag=3ea41b3c727533c2Contact: Call-ID: 484791787a5c06b6@10.200.26.116CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- <--- SIP read from 10.200.26.121:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0f6dec03;rportFrom: ;tag=as52c9f68fTo: "Danny D'Ambrosio" ;tag=91e4dffc190f5ae1Call-ID: 8c47d4781d5ccab6@10.200.26.121CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Extension Changed 201 new state Idle for Notify User 716 <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1a124f6a;rportFrom: "WIRELESS CALLER" ;tag=as4ec3b36bTo: Call-ID: 4b560a236b52eb1b768e3c3a48222ddf@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK292efbcd;rportFrom: ;tag=as3bd2c35dTo: "Jake Erramouspe" ;tag=d0e49efc170f18e1Call-ID: a94732785b5c49b6@10.200.26.122CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.120:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK52411c0d;rportFrom: ;tag=as26caf817To: "Darek Martinez" ;tag=1ae4770db0f4d3b8Call-ID: 0067f749305becc6@10.200.26.120CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Reliably Transmitting (NAT) to 10.200.26.111:5060: NOTIFY sip:711@10.200.26.111:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2510b448;rportFrom: ;tag=as4dacd753To: "Center Shop" ;tag=cf7dedb2b53fa60fContact: Call-ID: 43ce35395af9fd6f@10.200.26.111CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- <--- SIP read from 10.200.26.119:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1488ac85;rportFrom: ;tag=as4d16fad1To: "Randy Leffler" ;tag=9e94dba4740eb5d1Call-ID: 2f273697fcfc87f7@10.200.26.119CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1a124f6a;rportFrom: "WIRELESS CALLER" ;tag=as4ec3b36bTo: ;tag=9dcc5ef335a53591Call-ID: 4b560a236b52eb1b768e3c3a48222ddf@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6c8dd27e;rportFrom: ;tag=as501b143eTo: "Chris / Ed" ;tag=f9a4973cd07533c2Call-ID: 8647dd68f4fb0f08@10.200.26.118CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> Extension Changed 201 new state Idle for Notify User 711 Reliably Transmitting (NAT) to 10.200.26.115:5060: NOTIFY sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2bcf42f6;rportFrom: ;tag=as039135acTo: "Jerry Wright" ;tag=34b4537c3d36fe42Contact: Call-ID: af47d878926c2027@10.200.26.115CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 715 Reliably Transmitting (NAT) to 10.200.26.113:5060: NOTIFY sip:713@10.200.26.113:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK444c558a;rportFrom: ;tag=as05b1e1acTo: "Front Shop" ;tag=37a152d9516ad0eeContact: Call-ID: 49e06e58c4d76003@10.200.26.113CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 713 Reliably Transmitting (NAT) to 10.200.26.112:5060: NOTIFY sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1eceb242;rportFrom: ;tag=as0b0c2072To: "Lorrie Blake" ;tag=d3947f943676b5a2Contact: Call-ID: 0d175327994ac226@10.200.26.112CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 712 Reliably Transmitting (NAT) to 10.200.26.110:5060: NOTIFY sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK053c25d7;rportFrom: ;tag=as3b8bd8b5To: "Shawn Norton" ;tag=d481dbf1f79a741fContact: Call-ID: 0ca0cf97a319ed9e@10.200.26.110CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 710 Reliably Transmitting (NAT) to 10.200.26.109:5060: NOTIFY sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK005cab0d;rportFrom: ;tag=as24dbb862To: "Alan Young" ;tag=597133f1324431aeContact: Call-ID: ace04268693825af@10.200.26.109CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 709 Reliably Transmitting (NAT) to 10.200.26.108:5060: NOTIFY sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2c7e0821;rportFrom: ;tag=as23529257To: "Jake Ori" ;tag=3b5457d35e7edd92Contact: Call-ID: ae07f2b6183b61b5@10.200.26.108CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 708 Reliably Transmitting (NAT) to 10.200.26.107:5060: NOTIFY sip:707@10.200.26.107:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7228dfcb;rportFrom: ;tag=as55ec459bTo: "Tom Akers" ;tag=b864d414bbf47a88Contact: Call-ID: 05271c87d29c4c08@10.200.26.107CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Extension Changed 201 new state Idle for Notify User 707 <--- SIP read from 10.200.26.116:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0bae62a5;rportFrom: ;tag=as71101529To: "Shop Kitchen" ;tag=3ea41b3c727533c2Call-ID: 484791787a5c06b6@10.200.26.116CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.127:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK255414aa;rportFrom: ;tag=as0fac1be1To: "Conference Room" ;tag=1e94d3e74ab057e7Call-ID: cc74970ec691e7e1@10.200.26.127CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2bcf42f6;rportFrom: ;tag=as039135acTo: "Jerry Wright" ;tag=34b4537c3d36fe42Call-ID: af47d878926c2027@10.200.26.115CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.113:5060 ---> S asterisk-price*CLI> IP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK444c558a;rportFrom: ;tag=as05b1e1acTo: "Front Shop" ;tag=37a152d9516ad0eeCall-ID: 49e06e58c4d76003@10.200.26.113CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.111:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2510b448;rportFrom: ;tag=as4dacd753To: "Center Shop" ;tag=cf7dedb2b53fa60fCall-ID: 43ce35395af9fd6f@10.200.26.111CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.2.23Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1eceb242;rportFrom: ;tag=as0b0c2072To: "Lorrie Blake" ;tag=d3947f943676b5a2Call-ID: 0d175327994ac226@10.200.26.112CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Reliably Transmitting (NAT) to 10.200.26.106:5060: NOTIFY sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0d67ed8d;rportFrom: ;tag=as12414dc8To: "Jenine Bentley" ;tag=19cef75d507ad439Contact: Call-ID: ba019788181b78b6@10.200.26.106CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 706 <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK053c25d7;rportFrom: ;tag=as3b8bd8b5To: "Shawn Norton" ;tag=d481dbf1f79a741fCall-ID: 0ca0cf97a319ed9e@10.200.26.110CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK005cab0d;rportFrom: ;tag=as24dbb862To: "Alan Young" ;tag=597133f1324431aeCall-ID: ace04268693825af@10.200.26.109CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Reliably Transmitting (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK316dcdd2;rportFrom: ;tag=as30d10f77To: "Main Office Conf Room" ;tag=5b5475d3ba7e3992Contact: Call-ID: a2171917ddd9a517@10.200.26.105CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2c7e0821;rportFrom: ;tag=as23529257To: "Jake Ori" ;tag=3b5457d35e7edd92Call-ID: ae07f2b6183b61b5@10.200.26.108CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> Extension Changed 201 new state Idle for Notify User 705 Reliably Transmitting (NAT) to 10.200.26.104:5060: NOTIFY sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK74a1cd6c;rportFrom: ;tag=as6afb6310To: "Suzy Iorg" ;tag=06f2f98b43a41526Contact: Call-ID: 97b4dff5782a54a2@10.200.26.104CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 704 Reliably Transmitting (NAT) to 10.200.26.103:5060: NOTIFY sip:703@10.200.26.103:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7db2b797;rportFrom: ;tag=as0f08e20bTo: "Shelli Marvidakis" ;tag=9984b794f0a6b218Contact: Call-ID: 0a5774e87eaceb46@10.200.26.103CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 703 Reliably Transmitting (NAT) to 10.200.26.102:5060: NOTIFY sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK304d8a80;rportFrom: ;tag=as414071ddTo: "Jason Worley" ;tag=5b84999412a69418Contact: Call-ID: 8a5755e830bc6ea6@10.200.26.102CSeq: 112 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Extension Changed 201 new state Idle for Notify User 702 <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7228dfcb;rportFrom: ;tag=as55ec459bTo: "Tom Akers" ;tag=b864d414bbf47a88Call-ID: 05271c87d29c4c08@10.200.26.107CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0d67ed8d;rportFrom: ;tag=as12414dc8To: "Jenine Bentley" ;tag=19cef75d507ad439Call-ID: ba019788181b78b6@10.200.26.106CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Reliably Transmitting (NAT) to 10.200.26.101:5060: NOTIFY sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK41306c29;rportFrom: ;tag=as15c13c13To: "John Houston" ;tag=1d74fa5434d55668Contact: Call-ID: 88479078785cc3b6@10.200.26.101CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 701 <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK316dcdd2;rportFrom: ;tag=as30d10f77To: "Main Office Conf Room" ;tag=5b5475d3ba7e3992Call-ID: a2171917ddd9a517@10.200.26.105CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK74a1cd6c;rportFrom: ;tag=as6afb6310To: "Suzy Iorg" ;tag=06f2f98b43a41526Call-ID: 97b4dff5782a54a2@10.200.26.104CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7db2b797;rportFrom: ;tag=as0f08e20bTo: "Shelli Marvidakis" ;tag=9984b794f0a6b218Call-ID: 0a5774e87eaceb46@10.200.26.103CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK304d8a80;rportFrom: ;tag=as414071ddTo: "Jason Worley" ;tag=5b84999412a69418Call-ID: 8a5755e830bc6ea6@10.200.26.102CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK41306c29;rportFrom: ;tag=as15c13c13To: "John Houston" ;tag=1d74fa5434d55668Call-ID: 88479078785cc3b6@10.200.26.101CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> -- Called 717 -- SIP/717-09731328 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK21a06a47;rportFrom: ;tag=as5ab9c51aTo: "Ken Williams" ;tag=85b710412fc42025Call-ID: d416754dbbee5bda@10.200.26.117CSeq: 112 NOTIFYUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1a124f6a;rportFrom: "WIRELESS CALLER" ;tag=as4ec3b36bTo: ;tag=9dcc5ef335a53591Call-ID: 4b560a236b52eb1b768e3c3a48222ddf@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=717 8000 8000 IN IP4 10.200.26.117s=SIP Callc=IN IP4 10.200.26.117t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.117:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.117:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.117, port 5060 Transmitting (NAT) to 10.200.26.117:5060: ACK sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK61f19148;rportFrom: "WIRELESS CALLER" ;tag=as4ec3b36bTo: ;tag=9dcc5ef335a53591Contact: Call-ID: 4b560a236b52eb1b768e3c3a48222ddf@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- -- SIP/717-09731328 answered Zap/3-1 asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) SIP/709-0982c110 (None) Up Bridged Call(Zap/6-1) Zap/6-1 900@from-zaptel:6 Up Dial(SIP/701&SIP/703&SIP/704&S SIP/707-097a4ca0 202@parkedcalls:1 Up ParkedCall(202) Zap/2-1 900@from-zaptel:6 Up Bridged Call(SIP/707-097a4ca0) Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) 6 active channels 3 active calls asterisk-price*CLI> show channelsquit Really destroying SIP dialog '7c32f55a6fc1350157048bbd5fb8f3e1@66.111.122.20' Method: INVITE asterisk-price*CLI> quitshow channelsparkedcallschannelsparkedcallsasterisk-price*CLI> Num Channel (Context Extension Pri ) Timeout 0 parked calls. asterisk-price*CLI> show parkedcallschannelsasterisk-price*CLI> Channel Location State Application(Data) asterisk-price*CLI> SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) asterisk-price*CLI> SIP/709-0982c110 (None) Up Bridged Call(Zap/6-1) asterisk-price*CLI> Zap/6-1 900@from-zaptel:6 Up Dial(SIP/701&SIP/703&SIP/704&S asterisk-price*CLI> SIP/707-097a4ca0 202@parkedcalls:1 Up ParkedCall(202) asterisk-price*CLI> Zap/2-1 900@from-zaptel:6 Up Bridged Call(SIP/707-097a4ca0) asterisk-price*CLI> Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) asterisk-price*CLI> 6 active channels asterisk-price*CLI> 3 active calls asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> INVITE sip:4356376360@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK8ccf51d219f251fcFrom: ;tag=9501098a44676bf1To: "DUGOUT CANYON M" ;tag=as2b9ed892Contact: Supported: replaces, timerCall-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 1031 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 253v=0o=709 8001 8001 IN IP4 10.200.26.109s=SIP Callc=IN IP4 10.200.26.109t=0 0m=audio 5006 RTP/AVP 0 8 4 18 3a=sendonlya=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:4 G723/8000a=rtpmap:18 G729/8000a=rtpmap:3 GSM/8000a=ptime:20 <-------------> --- (13 headers 13 lines) --- Sending to 10.200.26.109 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 3 Peer audio RTP is at port 10.200.26.109:5006 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G723 for ID 4 Found description format G729 for ID 18 Found description format GSM for ID 3 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.109:5006 Audio is at 10.200.26.202 port 10878 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP <--- Transmitting (NAT) to 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK8ccf51d219f251fc;received=10.200.26.109From: ;tag=9501098a44676bf1To: "DUGOUT CANYON M" ;tag=as2b9ed892Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 1031 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Type: application/sdpContent-Length: 209v=0o=root 10024 10025 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10878 RTP/AVP 0 3a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=silenceSupp:off - - - -a=ptime:20a=recvonly <------------> -- Started music on hold, class 'default', on Zap/6-1 asterisk-price*CLI> [Jul 20 12:10:48] WARNING[10482]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> ACK sip:4356376360@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK0b81cace9494b3c3From: ;tag=9501098a44676bf1To: "DUGOUT CANYON M" ;tag=as2b9ed892Contact: Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 1031 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> REFER sip:4356376360@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bKa5a64be943e084f7From: ;tag=9501098a44676bf1To: "DUGOUT CANYON M" ;tag=as2b9ed892Contact: Supported: replacesRefer-To: Referred-By: Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 1032 REFERUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (14 headers 0 lines) --- Call 759c303713d201317a4f437476698b2e@66.111.122.20 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 200@from-internal by 709@10.200.26.202 -- Stopped music on hold on Zap/6-1 == Spawn extension (from-zaptel, 900, 6) exited non-zero on 'Zap/6-1' -- Executing [h@from-zaptel:1] Hangup("Zap/6-1", "") in new stack == Spawn extension (from-zaptel, h, 1) exited non-zero on 'Zap/6-1' asterisk-price*CLI> -- Started music on hold, class 'default', on Zap/6-1 asterisk-price*CLI> [Jul 20 12:10:49] WARNING[10059]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 asterisk-price*CLI> == Parked Zap/6-1 on 201@parkedcalls. Will timeout back to extension [from-zaptel] 900, 6 in 75 seconds asterisk-price*CLI> -- Added extension '201' priority 1 to parkedcalls asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.127:5060: NOTIFY sip:727@10.200.26.127:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK649f2283;rportFrom: ;tag=as0fac1be1To: "Conference Room" ;tag=1e94d3e74ab057e7Contact: Call-ID: cc74970ec691e7e1@10.200.26.127CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: active asterisk-price*CLI> Content-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 727 Reliably Transmitting (NAT) to 10.200.26.131:5060: NOTIFY sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7e06df41;rportFrom: ;tag=as44d1397bT asterisk-price*CLI> o: "Lobby" ;tag=99f4161d5df43d88Contact: Call-ID: 62279987519ceb08@10.200.26.131CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- asterisk-price*CLI> Extension Changed 201 new state InUse for Notify User 731 Reliably Transmitting (NAT) to 10.200.26.130:5060: NOTIFY sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4e5493e7;rportFrom: ;tag=as73346657To: "Foyer" ;tag=bee43bdcf28e33c2Contact: Call-ID: 0c17b027f44a0b16@10.200.26.130CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 730 Reliably Transmitting (NAT) to 10.200.26.129:5060: NOTIFY sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK114fd2fe;rportFrom: ;tag=as0b80a390To: "Will Jaimez" ;tag=18b4544c3b75f992Contact: Call-ID: 46071ba670da2a07@10.200.26.129CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 < asterisk-price*CLI> state>confirmed --- Extension Changed 201 new state InUse for Notify User 729 Reliably Transmitting (NAT) to 10.200.26.117:5060: NOTIFY sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK02a507a1;rportFrom: ;tag=as5ab9c51aTo: "Ken Williams" ;tag=85b710412fc42025Contact: Call-ID: d416754dbbee5bda@10.200.26.117CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- asterisk-price*CLI> Extension Changed 201 new state InUse for Notify User 717 Reliably Transmitting (NAT) to 10.200.26.126:5060: NOTIFY sip:726@10.200.26.126:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK419edac5;rportFrom: ;tag=as3f56cc30To: "Bob Teny" ;tag=94c4908c7746b742Contact: Call-ID: 8e17d427fa4a6426@10.200.26.126CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 726 Reliably Transmitting (NAT) to 10.200.26.125:5060: NOTIFY sip:725@10.200.26.125:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6159099f;rportFrom: ;tag=as38f74437To: "Front Desk" ;tag=1bd437ccfd6ebe92Contact: Call-ID: 62473a6832fbae08@10.200.26.125CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 725 Reliably Transmitting (NAT) to 10.200.26.124:5060: NOTIFY sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK46736e3f;rportFrom: ;tag=as35039685To: "Bobby Houston" ;tag=79c4178cb056f172Contact: Call-ID: 8d27b397d8fca0f7@10.200.26.124CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 724 Reliably Transmitting (NAT) to 10.200.26.121:5060: NOTIFY sip:721@10.200.26.121:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK396489b5;rportFrom: ;tag=as52c9f68fTo: "Danny D'Ambrosio" ;tag=91e4dffc190f5ae1Contact: Call-ID: 8c47d4781d5ccab6@10.200.26.121CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- asterisk-price*CLI> Extension Changed 201 new state InUse for Notify User 721 Reliably Transmitting (NAT) to 10.200.26.122:5060: NOTIFY sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d3cb006;rportFrom: ;tag=as3bd2c35dTo: "Jake Erramouspe" ;tag=d0e49efc170f18e1Contact: Call-ID: a94732785b5c49b6@10.200.26.122CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 722 Reliably Transmitting (NAT) to 10.200.26.120:5060: NOTIFY sip:720@10.200.26.120:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK532a0e9a;rportF asterisk-price*CLI> rom: ;tag=as26caf817To: "Darek Martinez" ;tag=1ae4770db0f4d3b8Contact: Call-ID: 0067f749305becc6@10.200.26.120CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 720 Reliably Transmitting (NAT) to 10.200.26.119:5060: NOTIFY sip:719@10.200.26.119:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK44a66a87;rportFrom: ;tag=as4d16fad1To: "Randy Leffler" ;tag=9e94dba4740eb5d1Contact: Call-ID: 2f273697fcfc87f7@10.200.26.119CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 719 Reliably Transmitting (NAT) to 10.200.26.118:5060: NOTIFY sip:718@10.200.26.118:5060 SIP/2.0V asterisk-price*CLI> ia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK432e4b52;rportFrom: ;tag=as501b143eTo: "Chris / Ed" ;tag=f9a4973cd07533c2Contact: Call-ID: 8647dd68f4fb0f08@10.200.26.118CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 718 Reliably Transmitting (NAT) to 10.200.26.116:5060: NOTIFY sip:716@10.200.26.116:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK17ea14e0;rportFrom: ;tag=as71101529To: "Shop Kitchen" ;tag=3ea41b3c727533c2Contact: Call-ID: 484791787a5c06b6@10.200.26.116CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 716 Reliably Transmitting (NAT) to 10.200.26.111:5060: NOTIFY sip:711@10.200.26.111:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4be03bbb;rportFrom: ;tag=as4dacd753To: "Center Shop" ;tag=cf7dedb2b53fa60fContact: Call-ID: 43ce35395af9fd6f@10.200.26.111CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 711 Reliably Transmitting (NAT) to 10.200.26.115:5060: NOTIFY sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5dbe0431;rportFrom: ;tag=as039135acTo: "Jerry Wright" ;tag=34b4537c3d36fe42Contact: Call-ID: af47d878926c2027@10.200.26.115CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 715 Reliably Transmitting (NAT) to 10.200.26.113:5060: NOTIFY sip:713@10.200.26.113:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK29204a62;rportFrom: ;tag=as05b1e1acTo: "Front Shop" ;tag=37a152d9516ad0eeContact: Call-ID: 49e06e58c4d76003@10.200.26.113CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 713 Reliably Transmitting (NAT) to 10.200.26.112:5060: NOTIFY sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d9cc3fd;rportFrom: ;tag=as0b0c2072To: "Lorrie Blake" ;tag=d3947f943676b5a2Contact: Call-ID: 0d175327994ac226@10.200.26.112CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 712 Reliably Transmitting (NAT) to 10.200.26.110:5060: NOTIFY sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK508ef8c8;rportFrom: ;tag=as3b8bd8b5To: "Shawn Norton" ;tag=d481dbf1f79a741fContact: Call-ID: 0ca0cf97a319ed9e@10.200.26.110CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 710 Reliably Transmitting (NAT) to 10.200.26.109:5060: NOTIFY sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK21544c4f;rportFrom: ;tag=as24dbb862To: "Alan Young" ;tag=597133f1324431aeContact: Call-ID: ace04268693825af@10.200.26.109CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 709 Reliably Transmitting (NAT) to 10.200.26.108:5060: NOTIFY sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2025da7f;rportFrom: ;tag=as23529257To: "Jake Ori" ;tag=3b5457d35e7edd92Contact: Call-ID: ae07f2b6183b61b5@10.200.26.108CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 708 Reliably Transmitting (NAT) to 10.200.26.107:5060: NOTIFY sip:707@10.200.26.107:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK70c2d153;rportFrom: ;tag=as55ec459bTo: "Tom Akers" ;tag=b864d414bbf47a88Contact: Call-ID: 05271c87d29c4c08@10.200.26.107CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 707 Reliably Transmitting (NAT) to 10.200.26.106:5060: NOTIFY sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK466bd1fa;rportFrom: ;tag=as12414dc8To: "Jenine Bentley" ;tag=19cef75d507ad439Contact: Call-ID: ba019788181b78b6@10.200.26.106CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 706 Reliably Transmitting (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK633b0bcf;rportFrom: ;tag=as30d10f77To: "Main Office Conf Room" ;tag=5b5475d3ba7e3992Contact: Call-ID: a2171917ddd9a517@10.200.26.105CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 705 Reliably Transmitting (NAT) to 10.200.26.104:5060: NOTIFY sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3bcf978b;rportFrom: ;tag=as6afb6310 asterisk-price*CLI> To: "Suzy Iorg" ;tag=06f2f98b43a41526Contact: Call-ID: 97b4dff5782a54a2@10.200.26.104CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 704 Reliably Transmitting (NAT) to 10.200.26.103:5060: NOTIFY sip:703@10.200.26.103:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5f5e36a8;rportFrom: ;tag=as0f08e20bTo: "Shelli Marvidakis" ;tag=9984b794f0a6b218Contact: Call-ID: 0a5774e87eaceb46@10.200.26.103CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 703 Reliably Transmitting (NAT) to 10.200.26.102:5060: NOTIFY sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4a801b69;rportFrom: ;tag=as414071ddTo: "Jason Worley" ;tag=5b84999412a69418Contact: Call-ID: 8a5755e830bc6ea6@10.200.26.102CSeq: 113 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 702 Reliably Transmitting (NAT) to 10.200.26.101:5060: NOTIFY sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK769bc447;rportFrom: ;tag=as15c13c13To: "John Houston" ;tag=1d74fa5434d55668Contact: Call-ID: 88479078785cc3b6@10.200.26.101CSeq: 114 NOTIFYU asterisk-price*CLI> ser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 701 asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7e06df41;rportFrom: ;tag=as44d1397bTo: "Lobby" ;tag=99f4161d5df43d88Call-ID: 62279987519ceb08@10.200.26.131CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4e5493e7;rportFrom: ;tag=as73346657To: "Foyer" ;tag=bee43bdcf28e33c2Call-ID: 0c17b027f44a0b16@10.200.26.130CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK114fd2fe;rportFrom: ;tag=as0b80a390To: "Will Jaimez" ;tag=18b4544c3b75f992Call-ID: 46071ba670da2a07@10.200.26.129C asterisk-price*CLI> Seq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK419edac5;rportFrom: ;tag=as3f56cc30To: "Bob Teny" ;tag=94c4908c7746b742Call-ID: 8e17d427fa4a6426@10.200.26.126CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6159099f;rportFrom: ;tag=as38f74437To: "Front Desk" ;tag=1bd437ccfd6ebe92Call-ID: 62473a6832fbae08@10.200.26.125CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK46736e3f;rportFrom: ;tag=as35039685To: "Bobby Houston" ;tag=79c4178cb056f172Call-ID: 8d27b397d8fca0f7@10.200.26.124CSeq: 113 NOTIFY asterisk-price*CLI> User-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.121:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK396489b5;rportFrom: ;tag=as52c9f68fTo: "Danny D'Ambrosio" ;tag=91e4dffc190f5ae1Call-ID: 8c47d4781d5ccab6@10.200.26.121CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d3cb006;rportFrom: ;tag=as3bd2c35dTo: "Jake Erramouspe" ;tag=d0e49efc170f18e1Call-ID: a94732785b5c49b6@10.200.26.122CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.120:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK532a0e9a;rportFrom: ;tag=as26caf817To: "Darek Martinez" ;tag=1ae4770db0f4d3b8Call-ID: 0067f749305becc6@10.200.26.120CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.119:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK44a66a87;rportFrom: ;tag=as4d16fad1To: "Randy Leffler" ;tag=9e94dba4740eb5d1Call-ID: 2f273697fcfc87f7@10.200.26.119CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK432e4b52;rportFrom: ;tag=as501b143eTo: "Chris / Ed" ;tag=f9a4973cd07533c2Call-ID: 8647dd68f4fb0f08@10.200.26.118CSeq: 113 NOTIFY asterisk-price*CLI> User-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.116:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK17ea14e0;rportFrom: ;tag=as71101529To: "Shop Kitchen" ;tag=3ea41b3c727533c2Call-ID: 484791787a5c06b6@10.200.26.116CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5dbe0431;rportFrom: ;tag=as039135acTo: "Jerry Wright" ;tag=34b4537c3d36fe42Call-ID: af47d878926c2027@10.200.26.115CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.111:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4be03bbb;rportFrom: ;tag=as4dacd753To: "Center Shop" ;tag=cf7dedb2b53fa60fCall-ID: 43ce35395af9fd6f@10.200.26.111CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.2.23Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.113:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK29204a62;rportFrom: ;tag=as05b1e1acTo: "Front Shop" ;tag=37a152d9516ad0eeCall-ID: 49e06e58c4d76003@10.200.26.113CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d9cc3fd;rportFrom: ;tag=as0b0c2072To: "Lorrie Blake" ;tag=d3947f943676b5a2Call-ID: 0d175327994ac226@10.200.26.112CSeq: 114 NOTIFYU asterisk-price*CLI> ser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK508ef8c8;rportFrom: ;tag=as3b8bd8b5To: "Shawn Norton" ;tag=d481dbf1f79a741fCall-ID: 0ca0cf97a319ed9e@10.200.26.110CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK21544c4f;rportFrom: ;tag=as24dbb862To: "Alan Young" ;tag=597133f1324431aeCall-ID: ace04268693825af@10.200.26.109CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2025da7f;rportFrom: ;tag=as23529257To: "Jake Ori" ;tag=3b5457d35e7edd92Call-ID: ae07f2b6183b61b5@10.200.26.108CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK70c2d153;rportFrom: ;tag=as55ec459bTo: "Tom Akers" ;tag=b864d414bbf47a88Call-ID: 05271c87d29c4c08@10.200.26.107CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK466bd1fa;rportFrom: ;tag=as12414dc8To: "Jenine Bentley" ;tag=19cef75d507ad439Call-ID: ba019788181b78b6@10.200.26.106CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK633b0bcf;rportFrom: ;tag=as30d10f77To: "Main Office Conf Room" ;tag=5b5475d3ba7e3992Call-ID: a2171917ddd9a517@10.200.26.105CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.127:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK649f2283;rportFrom: ;tag=as0fac1be1To: "Conference Room" ;tag=1e94d3e74ab057e7Call-ID: cc74970ec691e7e1@10.200.26.127CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3bcf978b;rportFrom: ;tag=as6afb6310To: "Suzy Iorg" ;tag=06f2f98b43a41526Call-ID: 97b4dff5782a54a2@10.200.26.104CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5f5e36a8;rportFrom: ;tag=as0f08e20bTo: "Shelli Marvidakis" ;tag=9984b794f0a6b218Call-ID: 0a5774e87eaceb46@10.200.26.103CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4a801b69;rportFrom: ;tag=as414071ddTo: "Jason Worley" ;tag=5b84999412a69418Call-ID: 8a5755e830bc6ea6@10.200.26.102CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK769bc447;rportFrom: ;tag=as15c13c13To: "John Houston" ;tag=1d74fa5434d55668Call-ID: 88479078785cc3b6@10.200.26.101CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK02a507a1;rportFrom: ;tag=as5ab9c51aTo: "Ken Williams" ;tag=85b710412fc42025Call-ID: d416754dbbee5bda@10.200.26.117CSeq: 113 NOTIFYUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- Transmitting (NAT) to 10.200.26.109:5060 ---> SIP/2.0 202 AcceptedVia: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bKa5a64be943e084f7;received=10.200.26.109From: ;tag=9501098a44676bf1To: "DUGOUT CANYON M" ;tag=as2b9ed892Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 1032 REFERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> set_destination: Parsing for address/port to send to asterisk-price*CLI> set_destination: set destination to 10.200.26.109, port 5060 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.109:5060: NOTIFY sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18006479;rportFrom: "DUGOUT CANYON M" ;tag=as2b9ed892To: ;tag=9501098a44676bf1Contact: Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=1032Subscription-state: terminated;reason=noresourceContent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 16SIP/2.0 200 OK --- asterisk-price*CLI> Scheduling destruction of SIP dialog '759c303713d201317a4f437476698b2e@66.111.122.20' in 32000 ms (Method: REFER) asterisk-price*CLI> set_destination: Parsing for address/port to send to asterisk-price*CLI> set_destination: set destination to 10.200.26.109, port 5060 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.109:5060: BYE sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK27bb7139;rportFrom: "DUGOUT CANYON M" ;tag=as2b9ed892To: ;tag=9501098a44676bf1Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 104 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18006479;rportFrom: "DUGOUT CANYON M" ;tag=as2b9ed892To: ;tag=9501098a44676bf1Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK27bb7139;rportFrom: "DUGOUT CANYON M" ;tag=as2b9ed892To: ;tag=9501098a44676bf1Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 104 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived asterisk-price*CLI> show channels Retransmitting #1 (NAT) to 10.200.26.109:5060: NOTIFY sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18006479;rportFrom: "DUGOUT CANYON M" ;tag=as2b9ed892To: ;tag=9501098a44676bf1Contact: Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=1032Subscription-state: terminated;reason=noresourceContent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 16SIP/2.0 200 OK --- asterisk-price*CLI> show channels<--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18006479;rportFrom: "DUGOUT CANYON M" ;tag=as2b9ed892To: ;tag=9501098a44676bf1Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) asterisk-price*CLI> Zap/6-1 900@from-zaptel:6 Up Parked Call() asterisk-price*CLI> SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) asterisk-price*CLI> SIP/707-097a4ca0 202@parkedcalls:1 Up ParkedCall(202) asterisk-price*CLI> Zap/2-1 900@from-zaptel:6 Up Bridged Call(SIP/707-097a4ca0) asterisk-price*CLI> Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) asterisk-price*CLI> 5 active channels asterisk-price*CLI> 2 active calls asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> INVITE sip:760@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK71c05ec33e1d0e93From: "Alan Young" ;tag=be9ae2b508f4ae4bTo: Contact: Supported: replaces, timerCall-ID: 550ff287113961b9@10.200.26.109CSeq: 53401 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 253v=0o=709 8000 8000 IN IP4 10.200.26.109s=SIP Callc=IN IP4 10.200.26.109t=0 0m=audio 5004 RTP/AVP 0 8 4 18 3a=sendrecva=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:4 G723/8000a=rtpmap:18 G729/8000a=rtpmap:3 GSM/8000a=ptime:20 <-------------> --- (13 headers 13 lines) --- Sending to 10.200.26.109 : 5060 (NAT) Using INVITE request as basis request - 550ff287113961b9@10.200.26.109 <--- Reliably Transmitting (NAT) to 10.200.26.109:5060 ---> SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK71c05ec33e1d0e93;received=10.200.26.109From: "Alan Young" ;tag=be9ae2b508f4ae4bTo: ;tag=as2c408a4dCall-ID: 550ff287113961b9@10.200.26.109CSeq: 53401 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesProxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4cbf0cab"Content-Length: 0 <------------> Scheduling destruction of SIP dialog '550ff287113961b9@10.200.26.109' in 32000 ms (Method: INVITE) Found user '709' asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> ACK sip:760@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK71c05ec33e1d0e93From: "Alan Young" ;tag=be9ae2b508f4ae4bTo: ;tag=as2c408a4dContact: Call-ID: 550ff287113961b9@10.200.26.109CSeq: 53401 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> INVITE sip:760@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK045ab35407f09d11From: "Alan Young" ;tag=be9ae2b508f4ae4bTo: Contact: Supported: replaces, timerProxy-Authorization: Digest username="709", realm="asterisk", algorithm=MD5, uri="sip:760@10.200.26.202", nonce="4cbf0cab", response="4baab248bfc2eca51ebeaf722e9c8d25"Call-ID: 550ff287113961b9@10.200.26.109CSeq: 53402 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 253v=0o=709 8000 8001 IN IP4 10.200.26.109s=SIP Callc=IN IP4 10.200.26.109t=0 0m=audio 5004 RTP/AVP 0 8 4 18 3a=sendrecva=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:4 G723/8000a=rtpmap:18 G729/8000a=rtpmap:3 GSM/8000a=ptime:20 <-------------> asterisk-price*CLI> --- (14 headers 13 lines) --- Sending to 10.200.26.109 : 5060 (NAT) Using INVITE request as basis request - 550ff287113961b9@10.200.26.109 Found user '709' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 3 Peer audio RTP is at port 10.200.26.109:5004 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G723 for ID 4 Found description format G729 for ID 18 Found description format GSM for ID 3 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.109:5004 Looking for 760 in from-internal (domain 10.200.26.202) list_route: hop: <--- Transmitting (NAT) to 10.200.26.109:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK045ab35407f09d11;received=10.200.26.109From: "Alan Young" ;tag=be9ae2b508f4ae4bTo: Call-ID: 550ff287113961b9@10.200.26.109CSeq: 53402 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> -- Executing [760@from-internal:1] GotoIf("SIP/709-09864628", "0?parkedcalls|200|1") in new stack asterisk-price*CLI> -- Executing [760@from-internal:2] SIPAddHeader("SIP/709-09864628", "Call-Info: answer-after=0") in new stack asterisk-price*CLI> -- Executing [760@from-internal:3] Set("SIP/709-09864628", "TIMEOUT(absolute)=10") in new stack asterisk-price*CLI> -- Channel will hangup at 2007-07-20 18:11:00 UTC. asterisk-price*CLI> -- Executing [760@from-internal:4] Page("SIP/709-09864628", "SIP/701&SIP/702&SIP/703&SIP/704&SIP/705&SIP/706&SIP/707&SIP/708&SIP/709&SIP/710&SIP/711&SIP/712&SIP/713&SIP/715&SIP/716&SIP/717&SIP/718&SIP/719&SIP/720&SIP/721&SIP/722&SIP/724&SIP/725&SIP/726&SIP/727&SIP/728&SIP/729&SIP/730&SIP/731&SIP/739|10") in new stack asterisk-price*CLI> Audio is at 10.200.26.202 port 10812 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.101:5060: INVITE sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK04100a20;rportFrom: "Alan Young" ;tag=as1d686695To: Contact: Call-ID: 6513fda91e8fc2874126d99c0ea7b0cf@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10812 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 701 asterisk-price*CLI> Audio is at 10.200.26.202 port 10288 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK04100a20;rportFrom: "Alan Young" ;tag=as1d686695To: Call-ID: 6513fda91e8fc2874126d99c0ea7b0cf@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK04100a20;rportFrom: "Alan Young" ;tag=as1d686695To: ;tag=037e0c420eea0a74Call-ID: 6513fda91e8fc2874126d99c0ea7b0cf@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.102:5060: INVITE sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4aefc467;rportFrom: "Alan Young" ;tag=as1ec98049To: Contact: Call-ID: 3bd61041520db5657513234b5f239ed5@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10288 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 702 asterisk-price*CLI> Audio is at 10.200.26.202 port 10748 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4aefc467;rportFrom: "Alan Young" ;tag=as1ec98049To: Call-ID: 3bd61041520db5657513234b5f239ed5@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4aefc467;rportFrom: "Alan Young" ;tag=as1ec98049To: ;tag=6f9f38c7676f62b5Call-ID: 3bd61041520db5657513234b5f239ed5@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.103:5060: INVITE sip:703@10.200.26.103:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK613678b1;rportFrom: "Alan Young" ;tag=as143a5b04To: Contact: Call-ID: 689c857d3beef73c11a1d45c32aabb9e@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10748 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- SIP/701-0983b4b0 is ringing asterisk-price*CLI> -- SIP/702-0984a588 is ringing asterisk-price*CLI> -- Called 703 asterisk-price*CLI> Audio is at 10.200.26.202 port 10090 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK613678b1;rportFrom: "Alan Young" ;tag=as143a5b04To: Call-ID: 689c857d3beef73c11a1d45c32aabb9e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK613678b1;rportFrom: "Alan Young" ;tag=as143a5b04To: ;tag=e19f9cb70ffeef2fCall-ID: 689c857d3beef73c11a1d45c32aabb9e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.104:5060: INVITE sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK70f6e16a;rportFrom: "Alan Young" ;tag=as0cfcd90cTo: Contact: Call-ID: 714546c31d85d1115a4e14477e76abda@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10090 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 704 asterisk-price*CLI> Audio is at 10.200.26.202 port 10576 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK70f6e16a;rportFrom: "Alan Young" ;tag=as0cfcd90cTo: Call-ID: 714546c31d85d1115a4e14477e76abda@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK70f6e16a;rportFrom: "Alan Young" ;tag=as0cfcd90cTo: ;tag=aa6163e6854126a5Call-ID: 714546c31d85d1115a4e14477e76abda@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.105:5060: INVITE sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3a79c47f;rportFrom: "Alan Young" ;tag=as2fdc543eTo: Contact: Call-ID: 37cfd5a204fd6f931bd631513ff8ea83@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10576 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 705 asterisk-price*CLI> Audio is at 10.200.26.202 port 10690 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3a79c47f;rportFrom: "Alan Young" ;tag=as2fdc543eTo: Call-ID: 37cfd5a204fd6f931bd631513ff8ea83@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3a79c47f;rportFrom: "Alan Young" ;tag=as2fdc543eTo: ;tag=3eacf011728a3dfcCall-ID: 37cfd5a204fd6f931bd631513ff8ea83@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.106:5060: INVITE sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1f49c3c4;rportFrom: "Alan Young" ;tag=as2721ab0bTo: Contact: Call-ID: 249244dc7ed95ae0342e8452157ca0db@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10690 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 706 asterisk-price*CLI> -- SIP/703-09819ef0 is ringing asterisk-price*CLI> -- SIP/704-097bc9a8 is ringing asterisk-price*CLI> -- SIP/705-0981e7e0 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1f49c3c4;rportFrom: "Alan Young" ;tag=as2721ab0bTo: Call-ID: 249244dc7ed95ae0342e8452157ca0db@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1f49c3c4;rportFrom: "Alan Young" ;tag=as2721ab0bTo: ;tag=27ebfade1f689e20Call-ID: 249244dc7ed95ae0342e8452157ca0db@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Audio is at 10.200.26.202 port 10436 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.107:5060: INVITE sip:707@10.200.26.107:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0f931b01;rportFrom: "Alan Young" ;tag=as6cc9ba3cTo: Contact: Call-ID: 695c12866313cbed2cfa4b56268463ec@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10436 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 707 asterisk-price*CLI> Audio is at 10.200.26.202 port 10234 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0f931b01;rportFrom: "Alan Young" ;tag=as6cc9ba3cTo: Call-ID: 695c12866313cbed2cfa4b56268463ec@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 486 BusyVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0f931b01;rportFrom: "Alan Young" ;tag=as6cc9ba3cTo: ;tag=as1219f5daCall-ID: 695c12866313cbed2cfa4b56268463ec@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- Got SIP response 486 "Busy" back from 10.200.26.107 Transmitting (NAT) to 10.200.26.107:5060: ACK sip:707@10.200.26.107:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0f931b01;rportFrom: "Alan Young" ;tag=as6cc9ba3cTo: ;tag=as1219f5daContact: Call-ID: 695c12866313cbed2cfa4b56268463ec@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.108:5060: INVITE sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK22644f71;rportFrom: "Alan Young" ;tag=as3c427729To: Contact: Call-ID: 126226a4412420b512d29c055f4ea93e@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10234 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 708 asterisk-price*CLI> Audio is at 10.200.26.202 port 10080 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK22644f71;rportFrom: "Alan Young" ;tag=as3c427729To: Call-ID: 126226a4412420b512d29c055f4ea93e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK22644f71;rportFrom: "Alan Young" ;tag=as3c427729To: ;tag=ea8ff457861f4bb4Call-ID: 126226a4412420b512d29c055f4ea93e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.110:5060: INVITE sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5d1be9ac;rportFrom: "Alan Young" ;tag=as70368615To: Contact: Call-ID: 4a9868a0074005112c751dbc5ab30b2c@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10080 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 710 asterisk-price*CLI> Audio is at 10.200.26.202 port 10094 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5d1be9ac;rportFrom: "Alan Young" ;tag=as70368615To: Call-ID: 4a9868a0074005112c751dbc5ab30b2c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5d1be9ac;rportFrom: "Alan Young" ;tag=as70368615To: ;tag=22af9bc5ab6fe4b5Call-ID: 4a9868a0074005112c751dbc5ab30b2c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.111:5060: INVITE sip:711@10.200.26.111:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5b056a8d;rportFrom: "Alan Young" ;tag=as01a3e519To: Contact: Call-ID: 724452b11d09bded08f602c50389f86c@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10094 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- SIP/706-0983d9a8 is ringing asterisk-price*CLI> -- SIP/707-0985fc90 is busy asterisk-price*CLI> -- SIP/708-098667e8 is ringing asterisk-price*CLI> -- SIP/710-0980ef48 is ringing asterisk-price*CLI> -- Called 711 asterisk-price*CLI> Audio is at 10.200.26.202 port 10390 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.112:5060: INVITE sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7a3cd000;rportFrom: "Alan Young" ;tag=as18f45065To: Contact: Call-ID: 12f02e0e0d9dadf17d5e51204c429325@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10390 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> <--- SIP read from 10.200.26.111:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5b056a8d;rportFrom: "Alan Young" ;tag=as01a3e519To: Call-ID: 724452b11d09bded08f602c50389f86c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.2.23Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> -- Called 712 asterisk-price*CLI> <--- SIP read from 10.200.26.111:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5b056a8d;rportFrom: "Alan Young" ;tag=as01a3e519To: ;tag=8f1bf4a301431f73Call-ID: 724452b11d09bded08f602c50389f86c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.2.23Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7a3cd000;rportFrom: "Alan Young" ;tag=as18f45065To: Call-ID: 12f02e0e0d9dadf17d5e51204c429325@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Audio is at 10.200.26.202 port 10778 asterisk-price*CLI> <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7a3cd000;rportFrom: "Alan Young" ;tag=as18f45065To: ;tag=24af7ec5007fa7e5Call-ID: 12f02e0e0d9dadf17d5e51204c429325@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.113:5060: INVITE sip:713@10.200.26.113:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK54023d91;rportFrom: "Alan Young" ;tag=as18cb2af3To: Contact: Call-ID: 174030d3780e24b66a7decf50f9870d3@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10778 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 713 asterisk-price*CLI> Audio is at 10.200.26.202 port 10554 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.113:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK54023d91;rportFrom: "Alan Young" ;tag=as18cb2af3To: Call-ID: 174030d3780e24b66a7decf50f9870d3@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.113:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK54023d91;rportFrom: "Alan Young" ;tag=as18cb2af3To: ;tag=03bfde36e18e8126Call-ID: 174030d3780e24b66a7decf50f9870d3@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.115:5060: INVITE sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK534c5808;rportFrom: "Alan Young" ;tag=as4cd9b896To: Contact: Call-ID: 11a2dee210a47ccc69f42efd41ba8d09@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10554 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 715 asterisk-price*CLI> -- SIP/711-09852018 is ringing asterisk-price*CLI> -- SIP/712-09823068 is ringing asterisk-price*CLI> -- SIP/713-097b6138 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK534c5808;rportFrom: "Alan Young" ;tag=as4cd9b896To: Call-ID: 11a2dee210a47ccc69f42efd41ba8d09@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK534c5808;rportFrom: "Alan Young" ;tag=as4cd9b896To: ;tag=a2affcc58e6f47b5Call-ID: 11a2dee210a47ccc69f42efd41ba8d09@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Really destroying SIP dialog '695c12866313cbed2cfa4b56268463ec@66.111.122.20' Method: INVITE asterisk-price*CLI> Audio is at 10.200.26.202 port 10470 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.116:5060: INVITE sip:716@10.200.26.116:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5311b8dc;rportFrom: "Alan Young" ;tag=as07888ab9To: Contact: Call-ID: 6d1316122b84406105cb9c00200be223@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10470 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 716 asterisk-price*CLI> Audio is at 10.200.26.202 port 10122 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.116:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5311b8dc;rportFrom: "Alan Young" ;tag=as07888ab9To: Call-ID: 6d1316122b84406105cb9c00200be223@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.116:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5311b8dc;rportFrom: "Alan Young" ;tag=as07888ab9To: ;tag=6eff81d55ad62abfCall-ID: 6d1316122b84406105cb9c00200be223@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.117:5060: INVITE sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6e9a807e;rportFrom: "Alan Young" ;tag=as27042008To: Contact: Call-ID: 31b1bcff6e11ad215ea0219664ab36e6@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10122 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 717 asterisk-price*CLI> Audio is at 10.200.26.202 port 10620 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.118:5060: INVITE sip:718@10.200.26.118:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18b7c3c1;rportFrom: "Alan Young" ;tag=as16470356To: Contact: Call-ID: 0e0a99cd7345a333057af7f03e34b098@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10620 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 718 asterisk-price*CLI> Audio is at 10.200.26.202 port 10504 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18b7c3c1;rportFrom: "Alan Young" ;tag=as16470356To: Call-ID: 0e0a99cd7345a333057af7f03e34b098@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18b7c3c1;rportFrom: "Alan Young" ;tag=as16470356To: ;tag=edaf18d549df4cdeCall-ID: 0e0a99cd7345a333057af7f03e34b098@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.119:5060: INVITE sip:719@10.200.26.119:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK13abc844;rportFrom: "Alan Young" ;tag=as5bc14d38To: Contact: Call-ID: 1ec04c233f204c127252e80604d42d38@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10504 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 719 asterisk-price*CLI> <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6e9a807e;rportFrom: "Alan Young" ;tag=as27042008To: Call-ID: 31b1bcff6e11ad215ea0219664ab36e6@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> -- SIP/715-09857028 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.119:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK13abc844;rportFrom: "Alan Young" ;tag=as5bc14d38To: Call-ID: 1ec04c233f204c127252e80604d42d38@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> -- SIP/716-0984ca68 is ringing asterisk-price*CLI> -- SIP/718-097cc9e8 is ringing asterisk-price*CLI> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.119:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK13abc844;rportFrom: "Alan Young" ;tag=as5bc14d38To: ;tag=cb20cd735727282eCall-ID: 1ec04c233f204c127252e80604d42d38@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6e9a807e;rportFrom: "Alan Young" ;tag=as27042008To: ;tag=559704474a96dbe7Call-ID: 31b1bcff6e11ad215ea0219664ab36e6@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Audio is at 10.200.26.202 port 10296 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.120:5060: INVITE sip:720@10.200.26.120:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK59ecc0db;rportFrom: "Alan Young" ;tag=as6ae36f4cTo: Contact: Call-ID: 2e243def0785365a0d5b4c816de46dd4@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10296 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 720 asterisk-price*CLI> Audio is at 10.200.26.202 port 10974 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.120:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK59ecc0db;rportFrom: "Alan Young" ;tag=as6ae36f4cTo: Call-ID: 2e243def0785365a0d5b4c816de46dd4@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.120:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK59ecc0db;rportFrom: "Alan Young" ;tag=as6ae36f4cTo: ;tag=4d0c8e6472167712Call-ID: 2e243def0785365a0d5b4c816de46dd4@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.121:5060: INVITE sip:721@10.200.26.121:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7821bfb2;rportFrom: "Alan Young" ;tag=as48eb6f2eTo: Contact: Call-ID: 0aa4c3ae40f6f454003f27bf43115396@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10974 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 721 asterisk-price*CLI> Audio is at 10.200.26.202 port 10416 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.121:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7821bfb2;rportFrom: "Alan Young" ;tag=as48eb6f2eTo: Call-ID: 0aa4c3ae40f6f454003f27bf43115396@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.121:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7821bfb2;rportFrom: "Alan Young" ;tag=as48eb6f2eTo: ;tag=802d037f5d126fb9Call-ID: 0aa4c3ae40f6f454003f27bf43115396@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.122:5060: INVITE sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3839fd2b;rportFrom: "Alan Young" ;tag=as3ccfab7bTo: Contact: Call-ID: 39c3390b7098186024cb48d625410c71@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10416 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 722 asterisk-price*CLI> <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK04100a20;rportFrom: "Alan Young" ;tag=as1d686695To: ;tag=037e0c420eea0a74Call-ID: 6513fda91e8fc2874126d99c0ea7b0cf@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=701 8000 8000 IN IP4 10.200.26.101s=SIP Callc=IN IP4 10.200.26.101t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> asterisk-price*CLI> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.101:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.101:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.101, port 5060 asterisk-price*CLI> Transmitting (NAT) to 10.200.26.101:5060: ACK sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2abfabf8;rportFrom: "Alan Young" ;tag=as1d686695To: ;tag=037e0c420eea0a74Contact: Call-ID: 6513fda91e8fc2874126d99c0ea7b0cf@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3839fd2b;rportFrom: "Alan Young" ;tag=as3ccfab7bTo: Call-ID: 39c3390b7098186024cb48d625410c71@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> -- SIP/701-0983b4b0 answered -- Created MeetMe conference 1023 for conference '200414677d' asterisk-price*CLI> <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3839fd2b;rportFrom: "Alan Young" ;tag=as3ccfab7bTo: ;tag=2741a01662bcebddCall-ID: 39c3390b7098186024cb48d625410c71@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Audio is at 10.200.26.202 port 10060 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.124:5060: INVITE sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK466318ee;rportFrom: "Alan Young" ;tag=as6f990c4cTo: Contact: Call-ID: 1ffd322c0b0cd4ee609a64a545c06530@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10060 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4aefc467;rportFrom: "Alan Young" ;tag=as1ec98049To: ;tag=6f9f38c7676f62b5Call-ID: 3bd61041520db5657513234b5f239ed5@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=702 8000 8000 IN IP4 10.200.26.102s=SIP Callc=IN IP4 10.200.26.102t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> asterisk-price*CLI> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.102:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.102:5004 asterisk-price*CLI> list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.102, port 5060 Transmitting (NAT) to 10.200.26.102:5060: ACK sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK051caa78;rportFrom: "Alan Young" ;tag=as1ec98049To: ;tag=6f9f38c7676f62b5Contact: Call-ID: 3bd61041520db5657513234b5f239ed5@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/702-0984a588 answered asterisk-price*CLI> -- Called 724 asterisk-price*CLI> -- SIP/719-09813b98 is ringing asterisk-price*CLI> -- SIP/717-0985fc90 is ringing asterisk-price*CLI> -- SIP/720-0982e138 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK466318ee;rportFrom: "Alan Young" ;tag=as6f990c4cTo: Call-ID: 1ffd322c0b0cd4ee609a64a545c06530@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> -- SIP/721-09832d00 is ringing asterisk-price*CLI> -- SIP/722-09842f88 is ringing asterisk-price*CLI> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK613678b1;rportFrom: "Alan Young" ;tag=as143a5b04To: ;tag=e19f9cb70ffeef2fCall-ID: 689c857d3beef73c11a1d45c32aabb9e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=703 8000 8000 IN IP4 10.200.26.103s=SIP Callc=IN IP4 10.200.26.103t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.103:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.103:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.103, port 5060 Transmitting (NAT) to 10.200.26.103:5060: ACK sip:703@10.200.26.103:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK61176c48;rportFrom: "Alan Young" ;tag=as143a5b04To: ;tag=e19f9cb70ffeef2fContact: Call-ID: 689c857d3beef73c11a1d45c32aabb9e@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK466318ee;rportFrom: "Alan Young" ;tag=as6f990c4cTo: ;tag=8bcf36b6473fea35Call-ID: 1ffd322c0b0cd4ee609a64a545c06530@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> -- SIP/703-09819ef0 answered asterisk-price*CLI> Audio is at 10.200.26.202 port 10902 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK70f6e16a;rportFrom: "Alan Young" ;tag=as0cfcd90cTo: ;tag=aa6163e6854126a5Call-ID: 714546c31d85d1115a4e14477e76abda@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=704 8000 8000 IN IP4 10.200.26.104s=SIP Callc=IN IP4 10.200.26.104t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.104:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.104:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.104, port 5060 Transmitting (NAT) to 10.200.26.104:5060: ACK sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1d21a4c7;rportFrom: "Alan Young" ;tag=as0cfcd90cTo: ;tag=aa6163e6854126a5Contact: Call-ID: 714546c31d85d1115a4e14477e76abda@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/704-097bc9a8 answered asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.125:5060: INVITE sip:725@10.200.26.125:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK366f0df7;rportFrom: "Alan Young" ;tag=as7febef1aTo: Contact: Call-ID: 4804b5d95d50c9b0263c38df5c3abe20@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10902 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 725 asterisk-price*CLI> Audio is at 10.200.26.202 port 10458 asterisk-price*CLI> <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK366f0df7;rportFrom: "Alan Young" ;tag=as7febef1aTo: Call-ID: 4804b5d95d50c9b0263c38df5c3abe20@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK366f0df7;rportFrom: "Alan Young" ;tag=as7febef1aTo: ;tag=ecaf56d587df6bdeCall-ID: 4804b5d95d50c9b0263c38df5c3abe20@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3a79c47f;rportFrom: "Alan Young" ;tag=as2fdc543eTo: ;tag=3eacf011728a3dfcCall-ID: 37cfd5a204fd6f931bd631513ff8ea83@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=705 8000 8000 IN IP4 10.200.26.105s=SIP Callc=IN IP4 10.200.26.105t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> asterisk-price*CLI> --- (12 headers 9 lines) --- asterisk-price*CLI> Found RTP audio format 0 asterisk-price*CLI> Peer audio RTP is at port 10.200.26.105:5004 asterisk-price*CLI> Found description format PCMU for ID 0 asterisk-price*CLI> Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) asterisk-price*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) asterisk-price*CLI> Peer audio RTP is at port 10.200.26.105:5004 asterisk-price*CLI> list_route: hop: asterisk-price*CLI> set_destination: Parsing for address/port to send to asterisk-price*CLI> set_destination: set destination to 10.200.26.105, port 5060 asterisk-price*CLI> Transmitting (NAT) to 10.200.26.105:5060: ACK sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK35afde80;rportFrom: "Alan Young" ;tag=as2fdc543eTo: ;tag=3eacf011728a3dfcContact: Call-ID: 37cfd5a204fd6f931bd631513ff8ea83@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/705-0981e7e0 answered asterisk-price*CLI> <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1f49c3c4;rportFrom: "Alan Young" ;tag=as2721ab0bTo: ;tag=27ebfade1f689e20Call-ID: 249244dc7ed95ae0342e8452157ca0db@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=706 8000 8000 IN IP4 10.200.26.106s=SIP Callc=IN IP4 10.200.26.106t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> asterisk-price*CLI> --- (12 headers 9 lines) --- asterisk-price*CLI> Found RTP audio format 0 asterisk-price*CLI> Peer audio RTP is at port 10.200.26.106:5004 asterisk-price*CLI> Found description format PCMU for ID 0 asterisk-price*CLI> Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) asterisk-price*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) asterisk-price*CLI> Peer audio RTP is at port 10.200.26.106:5004 asterisk-price*CLI> list_route: hop: asterisk-price*CLI> set_destination: Parsing for address/port to send to asterisk-price*CLI> set_destination: set destination to 10.200.26.106, port 5060 asterisk-price*CLI> Transmitting (NAT) to 10.200.26.106:5060: ACK sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK55356423;rportFrom: "Alan Young" ;tag=as2721ab0bTo: ;tag=27ebfade1f689e20Contact: Call-ID: 249244dc7ed95ae0342e8452157ca0db@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/706-0983d9a8 answered asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.126:5060: INVITE sip:726@10.200.26.126:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK61ede77f;rportFrom: "Alan Young" ;tag=as734b74e6To: Contact: Call-ID: 2c5441d61cec5a3741b7a30d66f8af7f@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10458 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 726 asterisk-price*CLI> Audio is at 10.200.26.202 port 10586 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK61ede77f;rportFrom: "Alan Young" ;tag=as734b74e6To: Call-ID: 2c5441d61cec5a3741b7a30d66f8af7f@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK22644f71;rportFrom: "Alan Young" ;tag=as3c427729To: ;tag=ea8ff457861f4bb4Call-ID: 126226a4412420b512d29c055f4ea93e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=708 8000 8000 IN IP4 10.200.26.108s=SIP Callc=IN IP4 10.200.26.108t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.108:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.108:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.108, port 5060 Transmitting (NAT) to 10.200.26.108:5060: ACK sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7aacf8c8;rportFrom: "Alan Young" ;tag=as3c427729To: ;tag=ea8ff457861f4bb4Contact: Call-ID: 126226a4412420b512d29c055f4ea93e@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK61ede77f;rportFrom: "Alan Young" ;tag=as734b74e6To: ;tag=0951617643fe6c9eCall-ID: 2c5441d61cec5a3741b7a30d66f8af7f@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> --- (10 headers 0 lines) --- -- SIP/708-098667e8 answered asterisk-price*CLI> <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5d1be9ac;rportFrom: "Alan Young" ;tag=as70368615To: ;tag=22af9bc5ab6fe4b5Call-ID: 4a9868a0074005112c751dbc5ab30b2c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=710 8000 8000 IN IP4 10.200.26.110s=SIP Callc=IN IP4 10.200.26.110t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.110:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.110:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.110, port 5060 Transmitting (NAT) to 10.200.26.110:5060: ACK sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6e61002f;rportFrom: "Alan Young" ;tag=as70368615To: ;tag=22af9bc5ab6fe4b5Contact: Call-ID: 4a9868a0074005112c751dbc5ab30b2c@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/710-0980ef48 answered asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.127:5062: INVITE sip:727@10.200.26.127:5062;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK334b08fe;rportFrom: "Alan Young" ;tag=as3623e17bTo: Contact: Call-ID: 61b60e040258966a5db77c1223f9d107@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10586 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 727 asterisk-price*CLI> Audio is at 10.200.26.202 port 10034 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK73a4e886;rportFrom: "Alan Young" ;tag=as58d0690dTo: Contact: Call-ID: 11de51510e007aef4b1998e10871de92@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10034 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 728 asterisk-price*CLI> -- SIP/724-09871b08 is ringing asterisk-price*CLI> -- SIP/725-09885cf0 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.111:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5b056a8d;rportFrom: "Alan Young" ;tag=as01a3e519To: ;tag=8f1bf4a301431f73Call-ID: 724452b11d09bded08f602c50389f86c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.2.23Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE asterisk-price*CLI> Content-Type: application/sdpSupported: replaces, timerContent-Length: 212v=0o=711 8000 8000 IN IP4 10.200.26.111s=SIP Callc=IN IP4 10.200.26.111t=0 0m=audio 5004 RTP/AVP 0 101a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-11 <-------------> asterisk-price*CLI> -- SIP/726-098a1af8 is ringing asterisk-price*CLI> --- (12 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.200.26.111:5004 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.200.26.111:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.111, port 5060 Transmitting (NAT) to 10.200.26.111:5060: ACK sip:711@10.200.26.111:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1ba3a6bb;rportFrom: "Alan Young" ;tag=as01a3e519To: ;tag=8f1bf4a301431f73Contact: Call-ID: 724452b11d09bded08f602c50389f86c@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7a3cd000;rportFrom: "Alan Young" ;tag=as18f45065To: ;tag=24af7ec5007fa7e5Call-ID: 12f02e0e0d9dadf17d5e51204c429325@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=712 8000 8000 IN IP4 10.200.26.112s=SIP Callc=IN IP4 10.200.26.112t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.112:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) asterisk-price*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.112:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.112, port 5060 Transmitting (NAT) to 10.200.26.112:5060: ACK sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6a3d4572;rportFrom: "Alan Young" ;tag=as18f45065To: ;tag=24af7ec5007fa7e5Contact: Call-ID: 12f02e0e0d9dadf17d5e51204c429325@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.127:5062 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK334b08fe;rportFrom: "Alan Young" ;tag=as3623e17bTo: C asterisk-price*CLI> all-ID: 61b60e040258966a5db77c1223f9d107@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.127:5062 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK334b08fe;rportFrom: "Alan Young" ;tag=as3623e17bTo: ;tag=5da5dd91851bd0a4Call-ID: 61b60e040258966a5db77c1223f9d107@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/711-09852018 answered asterisk-price*CLI> -- SIP/712-09823068 answered asterisk-price*CLI> <--- SIP read from 10.200.26.113:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK54023d91;rportFrom: "Alan Young" ;tag=as18cb2af3To: ;tag=03bfde36e18e8126Call-ID: 174030d3780e24b66a7decf50f9870d3@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=713 8000 8000 IN IP4 10.200.26.113s=SIP Callc=IN IP4 10.200.26.113t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- asterisk-price*CLI> Found RTP audio format 0 Peer audio RTP is at port 10.200.26.113:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.113:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.113, port 5060 Transmitting (NAT) to 10.200.26.113:5060: ACK sip:713@10.200.26.113:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0d8a8fb3;rportFrom: "Alan Young" ;tag=as18cb2af3To: ;tag=03bfde36e18e8126Contact: Call-ID: 174030d3780e24b66a7decf50f9870d3@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/713-097b6138 answered asterisk-price*CLI> -- SIP/727-098c12e0 is ringing asterisk-price*CLI> Audio is at 10.200.26.202 port 10126 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK534c5808;rportFrom: "Alan Young" ;tag=as4cd9b896To: ;tag=a2affcc58e6f47b5Call-ID: 11a2dee210a47ccc69f42efd41ba8d09@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=715 8000 8000 IN IP4 10.200.26.115s=SIP Callc=IN IP4 10.200.26.115t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> asterisk-price*CLI> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.115:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.115:5004 list_route: hop: asterisk-price*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.115, port 5060 Transmitting (NAT) to 10.200.26.115:5060: ACK sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5162530b;rportFrom: "Alan Young" ;tag=as4cd9b896To: ;tag=a2affcc58e6f47b5Contact: Call-ID: 11a2dee210a47ccc69f42efd41ba8d09@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/715-09857028 answered asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.129:5060: INVITE sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK020c8abe;rportFrom: "Alan Young" ;tag=as3fb97a89To: Contact: Call-ID: 74c246834a4bc359006fa4d47c53eb7f@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10126 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 729 asterisk-price*CLI> <--- SIP read from 10.200.26.116:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5311b8dc;rportFrom: "Alan Young" ;tag=as07888ab9To: ;tag=6eff81d55ad62abfCall-ID: 6d1316122b84406105cb9c00200be223@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=716 8000 8000 IN IP4 10.200.26.116s=SIP Callc=IN IP4 10.200.26.116t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.116:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.116:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.116, port 5060 Transmitting (NAT) to 10.200.26.116:5060: ACK sip:716@10.200.26.116:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK504925a3;rportFrom: "Alan Young" ;tag=as07888ab9To: ;tag=6eff81d55ad62abfContact: Call-ID: 6d1316122b84406105cb9c00200be223@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK020c8abe;rportFrom: "Alan Young" ;tag=as3fb97a89To: Call-ID: 74c246834a4bc359006fa4d47c53eb7f@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK020c8abe;rportFrom: "Alan Young" ;tag=as3fb97a89To: ;tag=eaaf35d566df0bdeCall-ID: 74c246834a4bc359006fa4d47c53eb7f@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> --- (10 headers 0 lines) --- asterisk-price*CLI> -- SIP/716-0984ca68 answered asterisk-price*CLI> Audio is at 10.200.26.202 port 10546 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.130:5060: INVITE sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3d7be483;rportFrom: "Alan Young" ;tag=as663caf72To: Contact: Call-ID: 5aaca74a674048a455c3118f2dba3864@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10546 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 730 asterisk-price*CLI> Audio is at 10.200.26.202 port 10558 asterisk-price*CLI> <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3d7be483;rportFrom: "Alan Young" ;tag=as663caf72To: Call-ID: 5aaca74a674048a455c3118f2dba3864@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18b7c3c1;rportFrom: "Alan Young" ;tag=as16470356To: ;tag=edaf18d549df4cdeCall-ID: 0e0a99cd7345a333057af7f03e34b098@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=718 8000 8000 IN IP4 10.200.26.118s=SIP Callc=IN IP4 10.200.26.118t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.118:5004 Found description format PCMU for ID 0 asterisk-price*CLI> Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.118:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.118, port 5060 Transmitting (NAT) to 10.200.26.118:5060: ACK sip:718@10.200.26.118:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK53aa5e83;rportFrom: "Alan Young" ;tag=as16470356To: ;tag=edaf18d549df4cdeContact: Call-ID: 0e0a99cd7345a333057af7f03e34b098@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3d7be483;rportFrom: "Alan Young" ;tag=as663caf72To: ;tag=81314985ea2fe7c4Call-ID: 5aaca74a674048a455c3118f2dba3864@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> -- SIP/718-097cc9e8 answered <--- SIP read from 10.200.26.119:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK13abc844;rportFrom: "Alan Young" ;tag=as5bc14d38To: ;tag=cb20cd735727282eCall-ID: 1ec04c233f204c127252e80604d42d38@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=719 8000 8000 IN IP4 10.200.26.119s=SIP Callc=IN IP4 10.200.26.119t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.119:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.119:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.119, port 5060 Transmitting (NAT) to 10.200.26.119:5060: ACK sip:719@10.200.26.119:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1e085c5c;rportFrom: "Alan Young" ;tag=as5bc14d38To: ;tag=cb20cd735727282eContact: Call-ID: 1ec04c233f204c127252e80604d42d38@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- -- SIP/719-09813b98 answered asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.131:5060: INVITE sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5411d294;rportFrom: "Alan Young" ;tag=as35b5aa8bTo: Contact: Call-ID: 5d131a4253c5d7ca00f45ebc216e5a94@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10558 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 731 asterisk-price*CLI> -- SIP/729-098fbc00 is ringing asterisk-price*CLI> -- SIP/730-09913c00 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5411d294;rportFrom: "Alan Young" ;tag=as35b5aa8bTo: Call-ID: 5d131a4253c5d7ca00f45ebc216e5a94@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5411d294;rportFrom: "Alan Young" ;tag=as35b5aa8bTo: ;tag=18716aff6d3f0115Call-ID: 5d131a4253c5d7ca00f45ebc216e5a94@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> --- (10 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.120:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK59ecc0db;rportFrom: "Alan Young" ;tag=as6ae36f4cTo: ;tag=4d0c8e6472167712Call-ID: 2e243def0785365a0d5b4c816de46dd4@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=720 8000 8000 IN IP4 10.200.26.120s=SIP Callc=IN IP4 10.200.26.120t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> asterisk-price*CLI> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.120:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.120:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.120, port 5060 asterisk-price*CLI> Transmitting (NAT) to 10.200.26.120:5060: ACK sip:720@10.200.26.120:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK56d71dfd;rportFrom: "Alan Young" ;tag=as6ae36f4cTo: ;tag=4d0c8e6472167712Contact: Call-ID: 2e243def0785365a0d5b4c816de46dd4@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/720-0982e138 answered asterisk-price*CLI> <--- SIP read from 10.200.26.121:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7821bfb2;rportFrom: "Alan Young" ;tag=as48eb6f2eTo: ;tag=802d037f5d126fb9Call-ID: 0aa4c3ae40f6f454003f27bf43115396@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=721 8000 8000 IN IP4 10.200.26.121s=SIP Callc=IN IP4 10.200.26.121t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.121:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.121:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.121, port 5060 Transmitting (NAT) to 10.200.26.121:5060: ACK sip:721@10.200.26.121:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0d72aa40;rportFrom: "Alan Young" ;tag=as48eb6f2eTo: ;tag=802d037f5d126fb9Contact: Call-ID: 0aa4c3ae40f6f454003f27bf43115396@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/721-09832d00 answered asterisk-price*CLI> -- SIP/731-09924198 is ringing asterisk-price*CLI> Retransmitting #2 (NAT) to 10.200.26.109:5060: NOTIFY sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18006479;rportFrom: "DUGOUT CANYON M" ;tag=as2b9ed892To: ;tag=9501098a44676bf1Contact: Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=1032Subscription-state: terminated;reason=noresourceContent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 16SIP/2.0 200 OK --- asterisk-price*CLI> <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3839fd2b;rportFrom: "Alan Young" ;tag=as3ccfab7bTo: ;tag=2741a01662bcebddCall-ID: 39c3390b7098186024cb48d625410c71@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=722 8000 8000 IN IP4 10.200.26.122s=SIP Callc=IN IP4 10.200.26.122t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.122:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.122:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.122, port 5060 Transmitting (NAT) to 10.200.26.122:5060: ACK sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK47ee90e7;rportFrom: "Alan Young" ;tag=as3ccfab7bTo: ;tag=2741a01662bcebddContact: Call-ID: 39c3390b7098186024cb48d625410c71@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/722-09842f88 answered <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18006479;rportFrom: "DUGOUT CANYON M" ;tag=as2b9ed892To: ;tag=9501098a44676bf1Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> Audio is at 10.200.26.202 port 10986 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.139:5060: INVITE sip:739@10.200.26.139:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3538d6ca;rportFrom: "Alan Young" ;tag=as7c00d669To: Contact: Call-ID: 46b8da124c1f2d144f4dca7a5030a15e@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY asterisk-price*CLI> Supported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10986 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK466318ee;rportFrom: "Alan Young" ;tag=as6f990c4cTo: ;tag=8bcf36b6473fea35Call-ID: 1ffd322c0b0cd4ee609a64a545c06530@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=724 8000 8000 IN IP4 10.200.26.124s=SIP Callc=IN IP4 10.200.26.124t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> asterisk-price*CLI> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.124:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.124:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.124, port 5060 Transmitting (NAT) to 10.200.26.124:5060: ACK sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK559ccc97;rportFrom: "Alan Young" ;tag=as6f990c4cTo: ;tag=8bcf36b6473fea35Contact: Call-ID: 1ffd322c0b0cd4ee609a64a545c06530@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 asterisk-price*CLI> --- asterisk-price*CLI> -- SIP/724-09871b08 answered asterisk-price*CLI> -- Called 739 asterisk-price*CLI> <--- SIP read from 10.200.26.139:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3538d6ca;rportFrom: "Alan Young" ;tag=as7c00d669To: Call-ID: 46b8da124c1f2d144f4dca7a5030a15e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.139:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3538d6ca;rportFrom: "Alan Young" ;tag=as7c00d669To: ;tag=84a4553e99ebda62Call-ID: 46b8da124c1f2d144f4dca7a5030a15e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 < asterisk-price*CLI> -------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Audio is at 10.200.26.202 port 10380 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- Transmitting (NAT) to 10.200.26.109:5060 ---> SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK045ab35407f09d11;received=10.200.26.109From: "Alan Young" ;tag=be9ae2b508f4ae4bTo: ;tag=as0908d4c5Call-ID: 550ff287113961b9@10.200.26.109CSeq: 53402 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Type: application/sdpContent-Length: 209v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10380 RTP/AVP 0 3a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=silenceSupp:off - - - -a=ptime:20a=sendrecv <------------> asterisk-price*CLI> -- Playing 'beep' (language 'en') asterisk-price*CLI> -- SIP/739-0993bca0 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK366f0df7;rportFrom: "Alan Young" ;tag=as7febef1aTo: ;tag=ecaf56d587df6bdeCall-ID: 4804b5d95d50c9b0263c38df5c3abe20@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=725 8000 8000 IN IP4 10.200.26.125s=SIP Callc=IN IP4 10.200.26.125t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> asterisk-price*CLI> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.125:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.125:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.125, port 5060 Transmitting (NAT) to 10.200.26.125:5060: ACK sip:725@10.200.26.125:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK63d4fd32;rportFrom: "Alan Young" ;tag=as7febef1aTo: ;tag=ecaf56d587df6bdeContact: Call-ID: 4804b5d95d50c9b0263c38df5c3abe20@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 asterisk-price*CLI> --- asterisk-price*CLI> -- SIP/725-09885cf0 answered asterisk-price*CLI> <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK61ede77f;rportFrom: "Alan Young" ;tag=as734b74e6To: ;tag=0951617643fe6c9eCall-ID: 2c5441d61cec5a3741b7a30d66f8af7f@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=726 8000 8000 IN IP4 10.200.26.126s=SIP Callc=IN IP4 10.200.26.126t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 asterisk-price*CLI> Peer audio RTP is at port 10.200.26.126:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.126:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.126, port 5060 Transmitting (NAT) to 10.200.26.126:5060: ACK sip:726@10.200.26.126:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK25910ce7;rportFrom: "Alan Young" ;tag=as734b74e6To: ;tag=0951617643fe6c9eContact: Call-ID: 2c5441d61cec5a3741b7a30d66f8af7f@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/726-098a1af8 answered asterisk-price*CLI> <--- SIP read from 10.200.26.127:5062 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK334b08fe;rportFrom: "Alan Young" ;tag=as3623e17bTo: ;tag=5da5dd91851bd0a4Call-ID: 61b60e040258966a5db77c1223f9d107@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE asterisk-price*CLI> Content-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=727 8000 8000 IN IP4 10.200.26.127s=SIP Callc=IN IP4 10.200.26.127t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.127:5004 asterisk-price*CLI> Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.127:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.127, port 5062 Transmitting (NAT) to 10.200.26.127:5062: ACK sip:727@10.200.26.127:5062;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK07ae4fec;rportFrom: "Alan Young" ;tag=as3623e17bTo: ;tag=5da5dd91851bd0a4Contact: Call-ID: 61b60e040258966a5db77c1223f9d107@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/727-098c12e0 answered asterisk-price*CLI> <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK020c8abe;rportFrom: "Alan Young" ;tag=as3fb97a89To: ;tag=eaaf35d566df0bdeCall-ID: 74c246834a4bc359006fa4d47c53eb7f@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=729 8000 8000 IN IP4 10.200.26.129s=SIP Callc=IN IP4 10.200.26.129t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> asterisk-price*CLI> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.129:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.129:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.129, port 5060 Transmitting (NAT) to 10.200.26.129:5060: ACK sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31138142;rportFrom: "Alan Young" ;tag=as3fb97a89To: ;tag=eaaf35d566df0bdeContact: Call-ID: 74c246834a4bc359006fa4d47c53eb7f@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 asterisk-price*CLI> --- asterisk-price*CLI> -- SIP/729-098fbc00 answered asterisk-price*CLI> <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3d7be483;rportFrom: "Alan Young" ;tag=as663caf72To: ;tag=81314985ea2fe7c4Call-ID: 5aaca74a674048a455c3118f2dba3864@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=730 8000 8000 IN IP4 10.200.26.130s=SIP Callc=IN IP4 10.200.26.130t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.130:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) asterisk-price*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.130:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.130, port 5060 Transmitting (NAT) to 10.200.26.130:5060: ACK sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2a9b8c91;rportFrom: "Alan Young" ;tag=as663caf72To: ;tag=81314985ea2fe7c4Contact: Call-ID: 5aaca74a674048a455c3118f2dba3864@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/730-09913c00 answered asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5411d294;rportFrom: "Alan Young" ;tag=as35b5aa8bTo: ;tag=18716aff6d3f0115Call-ID: 5d131a4253c5d7ca00f45ebc216e5a94@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=731 8000 8000 IN IP4 10.200.26.131s=SIP Callc=IN IP4 10.200.26.131t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.131:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.131:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.131, port 5060 Transmitting (NAT) to 10.200.26.131:5060: ACK sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK749c4578;rportFrom: "Alan Young" ;tag=as35b5aa8bTo: ;tag=18716aff6d3f0115Contact: Call-ID: 5d131a4253c5d7ca00f45ebc216e5a94@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/731-09924198 answered asterisk-price*CLI> <--- SIP read from 10.200.26.139:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3538d6ca;rportFrom: "Alan Young" ;tag=as7c00d669To: ;tag=84a4553e99ebda62Call-ID: 46b8da124c1f2d144f4dca7a5030a15e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 212v=0o=739 8000 8000 IN IP4 10.200.26.139s=SIP Callc=IN IP4 10.200.26.139t=0 0m=audio 5004 RTP/AVP 0 101a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-11 <-------------> --- (12 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.200.26.139:5004 Found description format PCMU for ID 0 asterisk-price*CLI> Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.200.26.139:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.139, port 5060 Transmitting (NAT) to 10.200.26.139:5060: ACK sip:739@10.200.26.139:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6c174a9c;rportFrom: "Alan Young" ;tag=as7c00d669To: ;tag=84a4553e99ebda62Contact: Call-ID: 46b8da124c1f2d144f4dca7a5030a15e@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/739-0993bca0 answered asterisk-price*CLI> Audio is at 10.200.26.202 port 10380 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- Reliably Transmitting (NAT) to 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK045ab35407f09d11;received=10.200.26.109From: "Alan Young" ;tag=be9ae2b508f4ae4bTo: ;tag=as0908d4c5Call-ID: 550ff287113961b9@10.200.26.109CSeq: 53402 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Type: application/sdpContent-Length: 209v=0o=root 10024 10025 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10380 RTP/AVP 0 3a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=silenceSupp:off - - - -a=ptime:20a=sendrecv <------------> asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> ACK sip:760@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK36f0f44b438802ccFrom: "Alan Young" ;tag=be9ae2b508f4ae4bTo: ;tag=as0908d4c5Contact: Proxy-Authorization: Digest username="709", realm="asterisk", algorithm=MD5, uri="sip:760@10.200.26.202", nonce="4cbf0cab", response="63655d368c61c3769973e1eab1bca612"Call-ID: 550ff287113961b9@10.200.26.109CSeq: 53402 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (12 headers 0 lines) --- asterisk-price*CLI> Retransmitting #1 (NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK73a4e886;rportFrom: "Alan Young" ;tag=as58d0690dTo: Contact: Call-ID: 11de51510e007aef4b1998e10871de92@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10034 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> Retransmitting #2 (NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK73a4e886;rportFrom: "Alan Young" ;tag=as58d0690dTo: Contact: Call-ID: 11de51510e007aef4b1998e10871de92@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10034 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> Retransmitting #3 (NAT) to 10.200.26.109:5060: NOTIFY sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18006479;rportFrom: "DUGOUT CANYON M" ;tag=as2b9ed892To: ;tag=9501098a44676bf1Contact: Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=1032Subscription-state: terminated;reason=noresourceContent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 16SIP/2.0 200 OK --- asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18006479;rportFrom: "DUGOUT CANYON M" ;tag=as2b9ed892To: ;tag=9501098a44676bf1Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> show channels Retransmitting #3 (NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK73a4e886;rportFrom: "Alan Young" ;tag=as58d0690dTo: Contact: Call-ID: 11de51510e007aef4b1998e10871de92@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:10:51 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10034 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> show channels<--- SIP read from 10.200.26.109:5060 ---> BYE sip:760@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bKe4adbaffa35042b7From: "Alan Young" ;tag=be9ae2b508f4ae4bTo: ;tag=as0908d4c5Proxy-Authorization: Digest username="709", realm="asterisk", algorithm=MD5, uri="sip:760@10.200.26.202", nonce="4cbf0cab", response="77961d4b462daafc2f0c3f11567d7f9f"Call-ID: 550ff287113961b9@10.200.26.109CSeq: 53403 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.200.26.109 : 5060 (NAT) <--- Transmitting (NAT) to 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bKe4adbaffa35042b7;received=10.200.26.109From: "Alan Young" ;tag=be9ae2b508f4ae4bTo: ;tag=as0908d4c5Call-ID: 550ff287113961b9@10.200.26.109CSeq: 53403 BYEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> show channels Scheduling destruction of SIP dialog '6513fda91e8fc2874126d99c0ea7b0cf@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.101, port 5060 Reliably Transmitting (NAT) to 10.200.26.101:5060: BYE sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4cdd42ae;rportFrom: "Alan Young" ;tag=as1d686695To: ;tag=037e0c420eea0a74Call-ID: 6513fda91e8fc2874126d99c0ea7b0cf@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '3bd61041520db5657513234b5f239ed5@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.102, port 5060 Reliably Transmitting (NAT) to 10.200.26.102:5060: BYE sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK70c437e4;rportFrom: "Alan Young" ;tag=as1ec98049To: ;tag=6f9f38c7676f62b5Call-ID: 3bd61041520db5657513234b5f239ed5@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '689c857d3beef73c11a1d45c32aabb9e@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.103, port 5060 Reliably Transmitting (NAT) to 10.200.26.103:5060: BYE sip:703@10.200.26.103:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6403168b;rportFrom: "Alan Young" ;tag=as143a5b04To: ;tag=e19f9cb70ffeef2fCall-ID: 689c857d3beef73c11a1d45c32aabb9e@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4cdd42ae;rportFrom: "Alan Young" ;tag=as1d686695To: ;tag=037e0c420eea0a74Call-ID: 6513fda91e8fc2874126d99c0ea7b0cf@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 asterisk-price*CLI> show channels <-------------> --- (11 headers 0 lines) --- Scheduling destruction of SIP dialog '714546c31d85d1115a4e14477e76abda@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.104, port 5060 Reliably Transmitting (NAT) to 10.200.26.104:5060: BYE sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK23b45242;rportFrom: "Alan Young" ;tag=as0cfcd90cTo: ;tag=aa6163e6854126a5Call-ID: 714546c31d85d1115a4e14477e76abda@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '37cfd5a204fd6f931bd631513ff8ea83@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.105, port 5060 Reliably Transmitting (NAT) to 10.200.26.105:5060: BYE sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5976664b;rportFrom: "Alan Young" ;tag=as2fdc543eTo: ;tag=3eacf011728a3dfcCall-ID: 37cfd5a204fd6f931bd631513ff8ea83@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '249244dc7ed95ae0342e8452157ca0db@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.106, port 5060 Reliably Transmitting (NAT) to 10.200.26.106:5060: BYE sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK305c5a49;rportFrom: "Alan Young" ;tag=as2721ab0bTo: ;tag=27ebfade1f689e20Call-ID: 249244dc7ed95ae0342e8452157ca0db@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- -- Hungup 'Zap/pseudo-553814494' <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK70c437e4;rportFrom: "Alan Young" ;tag=as1ec98049To: ;tag=6f9f38c7676f62b5Call-ID: 3bd61041520db5657513234b5f239ed5@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6403168b;rportFrom: "Alan Young" ;tag=as143a5b04To: ;tag=e19f9cb70ffeef2fCall-ID: 689c857d3beef73c11a1d45c32aabb9e@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '689c857d3beef73c11a1d45c32aabb9e@66.111.122.20' Method: INVITE Really destroying SIP dialog '3bd61041520db5657513234b5f239ed5@66.111.122.20' Method: INVITE Really destroying SIP dialog '6513fda91e8fc2874126d99c0ea7b0cf@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK23b45242;rportFrom: "Alan Young" ;tag=as0cfcd90cTo: ;tag=aa6163e6854126a5Call-ID: 714546c31d85d1115a4e14477e76abda@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5976664b;rportFrom: "Alan Young" ;tag=as2fdc543eTo: ;tag=3eacf011728a3dfcCall-ID: 37cfd5a204fd6f931bd631513ff8ea83@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK305c5a49;rportFrom: "Alan Young" ;tag=as2721ab0bTo: ;tag=27ebfade1f689e20Call-ID: 249244dc7ed95ae0342e8452157ca0db@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '249244dc7ed95ae0342e8452157ca0db@66.111.122.20' Method: INVITE Really destroying SIP dialog '37cfd5a204fd6f931bd631513ff8ea83@66.111.122.20' Method: INVITE Really destroying SIP dialog '714546c31d85d1115a4e14477e76abda@66.111.122.20' Method: INVITE Scheduling destruction of SIP dialog '126226a4412420b512d29c055f4ea93e@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.108, port 5060 Reliably Transmitting (NAT) to 10.200.26.108:5060: BYE sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK24c49015;rportFrom: "Alan Young" ;tag=as3c427729To: ;tag=ea8ff457861f4bb4Call-ID: 126226a4412420b512d29c055f4ea93e@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '4a9868a0074005112c751dbc5ab30b2c@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.110, port 5060 Reliably Transmitting (NAT) to 10.200.26.110:5060: BYE sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3dda7ee0;rportFrom: "Alan Young" ;tag=as70368615To: ;tag=22af9bc5ab6fe4b5Call-ID: 4a9868a0074005112c751dbc5ab30b2c@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '724452b11d09bded08f602c50389f86c@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.111, port 5060 Reliably Transmitting (NAT) to 10.200.26.111:5060: BYE sip:711@10.200.26.111:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK444509e8;rportFrom: "Alan Young" ;tag=as01a3e519To: ;tag=8f1bf4a301431f73Call-ID: 724452b11d09bded08f602c50389f86c@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '12f02e0e0d9dadf17d5e51204c429325@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.112, port 5060 Reliably Transmitting (NAT) to 10.200.26.112:5060: BYE sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK44858347;rportFrom: "Alan Young" ;tag=as18f45065To: ;tag=24af7ec5007fa7e5Call-ID: 12f02e0e0d9dadf17d5e51204c429325@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '174030d3780e24b66a7decf50f9870d3@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK24c49015;rportFrom: "Alan Young" ;tag=as3c427729To: ;tag=ea8ff457861f4bb4Call-ID: 126226a4412420b512d29c055f4ea93e@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3dda7ee0;rportFrom: "Alan Young" ;tag=as70368615To: ;tag=22af9bc5ab6fe4b5Call-ID: 4a9868a0074005112c751dbc5ab30b2c@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK44858347;rportFrom: "Alan Young" ;tag=as18f45065To: ;tag=24af7ec5007fa7e5Call-ID: 12f02e0e0d9dadf17d5e51204c429325@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- set_destination: set destination to 10.200.26.113, port 5060 Reliably Transmitting (NAT) to 10.200.26.113:5060: BYE sip:713@10.200.26.113:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31451755;rportFrom: "Alan Young" ;tag=as18cb2af3To: ;tag=03bfde36e18e8126Call-ID: 174030d3780e24b66a7decf50f9870d3@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '12f02e0e0d9dadf17d5e51204c429325@66.111.122.20' Method: INVITE Really destroying SIP dialog '4a9868a0074005112c751dbc5ab30b2c@66.111.122.20' Method: INVITE Really destroying SIP dialog '126226a4412420b512d29c055f4ea93e@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.111:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK444509e8;rportFrom: "Alan Young" ;tag=as01a3e519To: ;tag=8f1bf4a301431f73Call-ID: 724452b11d09bded08f602c50389f86c@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.2.23Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.113:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31451755;rportFrom: "Alan Young" ;tag=as18cb2af3To: ;tag=03bfde36e18e8126Call-ID: 174030d3780e24b66a7decf50f9870d3@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '174030d3780e24b66a7decf50f9870d3@66.111.122.20' Method: INVITE asterisk-price*CLI> show channels Really destroying SIP dialog '724452b11d09bded08f602c50389f86c@66.111.122.20' Method: INVITE Scheduling destruction of SIP dialog '11a2dee210a47ccc69f42efd41ba8d09@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.115, port 5060 Reliably Transmitting (NAT) to 10.200.26.115:5060: BYE sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK273a0196;rportFrom: "Alan Young" ;tag=as4cd9b896To: ;tag=a2affcc58e6f47b5Call-ID: 11a2dee210a47ccc69f42efd41ba8d09@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '6d1316122b84406105cb9c00200be223@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.116, port 5060 Reliably Transmitting (NAT) to 10.200.26.116:5060: BYE sip:716@10.200.26.116:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2666e5ca;rportFrom: "Alan Young" ;tag=as07888ab9To: ;tag=6eff81d55ad62abfCall-ID: 6d1316122b84406105cb9c00200be223@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK273a0196;rportFrom: "Alan Young" ;tag=as4cd9b896To: ;tag=a2affcc58e6f47b5Call-ID: 11a2dee210a47ccc69f42efd41ba8d09@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Scheduling destruction of SIP dialog '31b1bcff6e11ad215ea0219664ab36e6@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.117:5060: CANCEL sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6e9a807e;rportFrom: "Alan Young" ;tag=as27042008To: Call-ID: 31b1bcff6e11ad215ea0219664ab36e6@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '31b1bcff6e11ad215ea0219664ab36e6@66.111.122.20' in 32000 ms (Method: INVITE) Really destroying SIP dialog '11a2dee210a47ccc69f42efd41ba8d09@66.111.122.20' Method: INVITE Scheduling destruction of SIP dialog '0e0a99cd7345a333057af7f03e34b098@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.118, port 5060 Reliably Transmitting (NAT) to 10.200.26.118:5060: BYE sip:718@10.200.26.118:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2b9f67ba;rportFrom: "Alan Young" ;tag=as16470356To: ;tag=edaf18d549df4cdeCall-ID: 0e0a99cd7345a333057af7f03e34b098@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '1ec04c233f204c127252e80604d42d38@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.119, port 5060 Reliably Transmitting (NAT) to 10.200.26.119:5060: BYE sip:719@10.200.26.119:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4016974d;rportFrom: "Alan Young" ;tag=as5bc14d38To: ;tag=cb20cd735727282eCall-ID: 1ec04c233f204c127252e80604d42d38@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.116:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2666e5ca;rportFrom: "Alan Young" ;tag=as07888ab9To: ;tag=6eff81d55ad62abfCall-ID: 6d1316122b84406105cb9c00200be223@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> Scheduling destruction of SIP dialog '2e243def0785365a0d5b4c816de46dd4@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.120, port 5060 Reliably Transmitting (NAT) to 10.200.26.120:5060: BYE sip:720@10.200.26.120:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7c9211a6;rportFrom: "Alan Young" ;tag=as6ae36f4cTo: ;tag=4d0c8e6472167712Call-ID: 2e243def0785365a0d5b4c816de46dd4@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '0aa4c3ae40f6f454003f27bf43115396@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.121, port 5060 Reliably Transmitting (NAT) to 10.200.26.121:5060: BYE sip:721@10.200.26.121:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK372dd36a;rportFrom: "Alan Young" ;tag=as48eb6f2eTo: ;tag=802d037f5d126fb9Call-ID: 0aa4c3ae40f6f454003f27bf43115396@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '39c3390b7098186024cb48d625410c71@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.122, port 5060 Reliably Transmitting (NAT) to 10.200.26.122:5060: BYE sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK75897263;rportFrom: "Alan Young" ;tag=as3ccfab7bTo: ;tag=2741a01662bcebddCall-ID: 39c3390b7098186024cb48d625410c71@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2b9f67ba;rportFrom: "Alan Young" ;tag=as16470356To: ;tag=edaf18d549df4cdeCall-ID: 0e0a99cd7345a333057af7f03e34b098@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.119:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4016974d;rportFrom: "Alan Young" ;tag=as5bc14d38To: ;tag=cb20cd735727282eCall-ID: 1ec04c233f204c127252e80604d42d38@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '1ec04c233f204c127252e80604d42d38@66.111.122.20' Method: INVITE Really destroying SIP dialog '0e0a99cd7345a333057af7f03e34b098@66.111.122.20' Method: INVITE Really destroying SIP dialog '6d1316122b84406105cb9c00200be223@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.120:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7c9211a6;rportFrom: "Alan Young" ;tag=as6ae36f4cTo: ;tag=4d0c8e6472167712Call-ID: 2e243def0785365a0d5b4c816de46dd4@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.121:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK372dd36a;rportFrom: "Alan Young" ;tag=as48eb6f2eTo: ;tag=802d037f5d126fb9Call-ID: 0aa4c3ae40f6f454003f27bf43115396@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6e9a807e;rportFrom: "Alan Young" ;tag=as27042008To: ;tag=559704474a96dbe7Call-ID: 31b1bcff6e11ad215ea0219664ab36e6@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.4.18Supported: replaces, timerContent-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '0aa4c3ae40f6f454003f27bf43115396@66.111.122.20' Method: INVITE Really destroying SIP dialog '2e243def0785365a0d5b4c816de46dd4@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK75897263;rportFrom: "Alan Young" ;tag=as3ccfab7bTo: ;tag=2741a01662bcebddCall-ID: 39c3390b7098186024cb48d625410c71@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Scheduling destruction of SIP dialog '1ffd322c0b0cd4ee609a64a545c06530@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.124, port 5060 Reliably Transmitting (NAT) to 10.200.26.124:5060: BYE sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2cbc7001;rportFrom: "Alan Young" ;tag=as6f990c4cTo: ;tag=8bcf36b6473fea35Call-ID: 1ffd322c0b0cd4ee609a64a545c06530@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '4804b5d95d50c9b0263c38df5c3abe20@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.125, port 5060 Reliably Transmitting (NAT) to 10.200.26.125:5060: BYE sip:725@10.200.26.125:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK26eb1499;rportFrom: "Alan Young" ;tag=as7febef1aTo: ;tag=ecaf56d587df6bdeCall-ID: 4804b5d95d50c9b0263c38df5c3abe20@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '2c5441d61cec5a3741b7a30d66f8af7f@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.126, port 5060 Reliably Transmitting (NAT) to 10.200.26.126:5060: BYE sip:726@10.200.26.126:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK311cb8c2;rportFrom: "Alan Young" ;tag=as734b74e6To: ;tag=0951617643fe6c9eCall-ID: 2c5441d61cec5a3741b7a30d66f8af7f@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '61b60e040258966a5db77c1223f9d107@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.127, port 5062 Reliably Transmitting (NAT) to 10.200.26.127:5062: BYE sip:727@10.200.26.127:5062;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK63101a68;rportFrom: "Alan Young" ;tag=as3623e17bTo: ;tag=5da5dd91851bd0a4Call-ID: 61b60e040258966a5db77c1223f9d107@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 asterisk-price*CLI> show channels --- <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6e9a807e;rportFrom: "Alan Young" ;tag=as27042008To: ;tag=559704474a96dbe7Call-ID: 31b1bcff6e11ad215ea0219664ab36e6@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.117:5060: ACK sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6e9a807e;rportFrom: "Alan Young" ;tag=as27042008To: ;tag=559704474a96dbe7Contact: Call-ID: 31b1bcff6e11ad215ea0219664ab36e6@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '39c3390b7098186024cb48d625410c71@66.111.122.20' Method: INVITE Really destroying SIP dialog '31b1bcff6e11ad215ea0219664ab36e6@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2cbc7001;rportFrom: "Alan Young" ;tag=as6f990c4cTo: ;tag=8bcf36b6473fea35Call-ID: 1ffd322c0b0cd4ee609a64a545c06530@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK26eb1499;rportFrom: "Alan Young" ;tag=as7febef1aTo: ;tag=ecaf56d587df6bdeCall-ID: 4804b5d95d50c9b0263c38df5c3abe20@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK311cb8c2;rportFrom: "Alan Young" ;tag=as734b74e6To: ;tag=0951617643fe6c9eCall-ID: 2c5441d61cec5a3741b7a30d66f8af7f@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '2c5441d61cec5a3741b7a30d66f8af7f@66.111.122.20' Method: INVITE Really destroying SIP dialog '4804b5d95d50c9b0263c38df5c3abe20@66.111.122.20' Method: INVITE Really destroying SIP dialog '1ffd322c0b0cd4ee609a64a545c06530@66.111.122.20' Method: INVITE Scheduling destruction of SIP dialog '11de51510e007aef4b1998e10871de92@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '74c246834a4bc359006fa4d47c53eb7f@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.129, port 5060 Reliably Transmitting (NAT) to 10.200.26.129:5060: BYE sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK07c16736;rportFrom: "Alan Young" ;tag=as3fb97a89To: ;tag=eaaf35d566df0bdeCall-ID: 74c246834a4bc359006fa4d47c53eb7f@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '5aaca74a674048a455c3118f2dba3864@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.130, port 5060 Reliably Transmitting (NAT) to 10.200.26.130:5060: BYE sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK490f923f;rportFrom: "Alan Young" ;tag=as663caf72To: ;tag=81314985ea2fe7c4Call-ID: 5aaca74a674048a455c3118f2dba3864@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '5d131a4253c5d7ca00f45ebc216e5a94@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.131, port 5060 Reliably Transmitting (NAT) to 10.200.26.131:5060: BYE sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0309daf5;rportFrom: "Alan Young" ;tag=as35b5aa8bTo: ;tag=18716aff6d3f0115Call-ID: 5d131a4253c5d7ca00f45ebc216e5a94@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '46b8da124c1f2d144f4dca7a5030a15e@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.139, port 5060 Reliably Transmitting (NAT) to 10.200.26.139:5060: BYE sip:739@10.200.26.139:5060 SIP/2.0V asterisk-price*CLI> show channels ia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0d4d7b67;rportFrom: "Alan Young" ;tag=as7c00d669To: ;tag=84a4553e99ebda62Call-ID: 46b8da124c1f2d144f4dca7a5030a15e@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- == Spawn extension (from-internal, 760, 4) exited non-zero on 'SIP/709-09864628' -- Executing [h@from-internal:1] Hangup("SIP/709-09864628", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/709-09864628' <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK07c16736;rportFrom: "Alan Young" ;tag=as3fb97a89To: ;tag=eaaf35d566df0bdeCall-ID: 74c246834a4bc359006fa4d47c53eb7f@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK490f923f;rportFrom: "Alan Young" ;tag=as663caf72To: ;tag=81314985ea2fe7c4Call-ID: 5aaca74a674048a455c3118f2dba3864@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.127:5062 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK63101a68;rportFrom: "Alan Young" ;tag=as3623e17bTo: ;tag=5da5dd91851bd0a4Call-ID: 61b60e040258966a5db77c1223f9d107@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '5aaca74a674048a455c3118f2dba3864@66.111.122.20' Method: INVITE Really destroying SIP dialog '74c246834a4bc359006fa4d47c53eb7f@66.111.122.20' Method: INVITE Really destroying SIP dialog '61b60e040258966a5db77c1223f9d107@66.111.122.20' Method: INVITE Really destroying SIP dialog '550ff287113961b9@10.200.26.109' Method: BYE <--- SIP read from 10.200.26.139:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0d4d7b67;rportFrom: "Alan Young" ;tag=as7c00d669To: ;tag=84a4553e99ebda62Call-ID: 46b8da124c1f2d144f4dca7a5030a15e@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0309daf5;rportFrom: "Alan Young" ;tag=as35b5aa8bTo: ;tag=18716aff6d3f0115Call-ID: 5d131a4253c5d7ca00f45ebc216e5a94@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '46b8da124c1f2d144f4dca7a5030a15e@66.111.122.20' Method: INVITE Really destroying SIP dialog '5d131a4253c5d7ca00f45ebc216e5a94@66.111.122.20' Method: INVITE asterisk-price*CLI> show channels Retransmitting #4 (NAT) to 10.200.26.109:5060: NOTIFY sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18006479;rportFrom: "DUGOUT CANYON M" ;tag=as2b9ed892To: ;tag=9501098a44676bf1Contact: Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=1032Subscription-state: terminated;reason=noresourceContent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 16SIP/2.0 200 OK --- asterisk-price*CLI> show channels<--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18006479;rportFrom: "DUGOUT CANYON M" ;tag=as2b9ed892To: ;tag=9501098a44676bf1Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) Zap/6-1 900@from-zaptel:6 Up Parked Call() SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) SIP/707-097a4ca0 202@parkedcalls:1 Up ParkedCall(202) Zap/2-1 900@from-zaptel:6 Up Bridged Call(SIP/707-097a4ca0) Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) 5 active channels 2 active calls asterisk-price*CLI> Retransmitting #5 (NAT) to 10.200.26.109:5060: NOTIFY sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18006479;rportFrom: "DUGOUT CANYON M" ;tag=as2b9ed892To: ;tag=9501098a44676bf1Contact: Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=1032Subscription-state: terminated;reason=noresourceContent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 16SIP/2.0 200 OK --- asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18006479;rportFrom: "DUGOUT CANYON M" ;tag=as2b9ed892To: ;tag=9501098a44676bf1Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.122:5060 ---> INVITE sip:709@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.122:5060;branch=z9hG4bK66e93b9ff73892d0From: "Jake Erramouspe" ;tag=6b1f101ca51707efTo: Contact: Supported: replaces, timerCall-ID: fd4df747f847f904@10.200.26.122CSeq: 55852 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 156v=0o=722 8000 8000 IN IP4 10.200.26.122s=SIP Callc=IN IP4 10.200.26.122t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> asterisk-price*CLI> --- (13 headers 9 lines) --- Sending to 10.200.26.122 : 5060 (NAT) Using INVITE request as basis request - fd4df747f847f904@10.200.26.122 <--- Reliably Transmitting (NAT) to 10.200.26.122:5060 ---> SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 10.200.26.122:5060;branch=z9hG4bK66e93b9ff73892d0;received=10.200.26.122From: "Jake Erramouspe" ;tag=6b1f101ca51707efTo: ;tag=as0cd8d147Call-ID: fd4df747f847f904@10.200.26.122CSeq: 55852 INVITE asterisk-price*CLI> User-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesProxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2b2b3c08"Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'fd4df747f847f904@10.200.26.122' in 32000 ms (Method: INVITE) Found user '722' asterisk-price*CLI> <--- SIP read from 10.200.26.122:5060 ---> ACK sip:709@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.122:5060;branch=z9hG4bK66e93b9ff73892d0From: "Jake Erramouspe" ;tag=6b1f101ca51707efTo: ;tag=as0cd8d147Contact: Call-ID: fd4df747f847f904@10.200.26.122CSeq: 55852 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.122:5060 ---> INVITE sip:709@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.122:5060;branch=z9hG4bKa1d9216a19fba8cbFrom: "Jake Erramouspe" ;tag=6b1f101ca51707efTo: Contact: Supported: replaces, timerProxy-Authorization: Digest username="722", realm="asterisk", algorithm=MD5, uri="sip:709@10.200.26.202", nonce="2b2b3c08", response="bb81429652e694844958ceeff5595334"Call-ID: fd4df747f847f904@10.200.26.122CSeq: 55853 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 156v=0o=722 8000 8001 IN IP4 10.200.26.122s=SIP Callc=IN IP4 10.200.26.122t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (14 headers 9 lines) --- asterisk-price*CLI> Sending to 10.200.26.122 : 5060 (NAT) Using INVITE request as basis request - fd4df747f847f904@10.200.26.122 Found user '722' Found RTP audio format 0 Peer audio RTP is at port 10.200.26.122:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.122:5004 Looking for 709 in from-internal (domain 10.200.26.202) list_route: hop: <--- Transmitting (NAT) to 10.200.26.122:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.122:5060;branch=z9hG4bKa1d9216a19fba8cb;received=10.200.26.122From: "Jake Erramouspe" ;tag=6b1f101ca51707efTo: Call-ID: fd4df747f847f904@10.200.26.122CSeq: 55853 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> -- Executing [709@from-internal:1] SIPAddHeader("SIP/722-0984a588", "Call-Info: answer-after=0") in new stack asterisk-price*CLI> -- Executing [709@from-internal:2] Dial("SIP/722-0984a588", "SIP/709|20|Tt") in new stack asterisk-price*CLI> Audio is at 10.200.26.202 port 10726 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.109:5060: INVITE sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7e691117;rportFrom: "Jake Erramouspe" ;tag=as53f6a942To: Contact: Call-ID: 2838eeb65b57fab22d0cfd1971038498@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:11:01 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10726 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 709 asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7e691117;rportFrom: "Jake Erramouspe" ;tag=as53f6a942To: Call-ID: 2838eeb65b57fab22d0cfd1971038498@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7e691117;rportFrom: "Jake Erramouspe" ;tag=as53f6a942To: ;tag=b59cf46dfd1c7247Call-ID: 2838eeb65b57fab22d0cfd1971038498@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> -- SIP/709-09864628 is ringing asterisk-price*CLI> <--- Transmitting (NAT) to 10.200.26.122:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.122:5060;branch=z9hG4bKa1d9216a19fba8cb;received=10.200.26.122From: "Jake Erramouspe" ;tag=6b1f101ca51707efTo: ;tag=as5851220cCall-ID: fd4df747f847f904@10.200.26.122CSeq: 55853 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 < asterisk-price*CLI> ------------> asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7e691117;rportFrom: "Jake Erramouspe" ;tag=as53f6a942To: ;tag=b59cf46dfd1c7247Call-ID: 2838eeb65b57fab22d0cfd1971038498@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=709 8000 8000 IN IP4 10.200.26.109s=SIP Callc=IN IP4 10.200.26.109t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.109:5004 Found description format PCMU for ID 0 asterisk-price*CLI> Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.109:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.109, port 5060 Transmitting (NAT) to 10.200.26.109:5060: ACK sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4c01864b;rportFrom: "Jake Erramouspe" ;tag=as53f6a942To: ;tag=b59cf46dfd1c7247Contact: Call-ID: 2838eeb65b57fab22d0cfd1971038498@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/709-09864628 answered SIP/722-0984a588 Audio is at 10.200.26.202 port 10200 Adding codec 0x4 (ulaw) to SDP <--- Reliably Transmitting (NAT) to 10.200.26.122:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.122:5060;branch=z9hG4bKa1d9216a19fba8cb;received=10.200.26.122From: "Jake Erramouspe" ;tag=6b1f101ca51707efTo: ;tag=as5851220cCall-ID: fd4df747f847f904@10.200.26.122CSeq: 55853 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Type: application/sdpContent-Length: 186v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10200 RTP/AVP 0a=rtpmap:0 PCMU/8000a=silenceSupp:off - - - -a=ptime:20a=sendrecv <------------> asterisk-price*CLI> <--- SIP read from 10.200.26.122:5060 ---> ACK sip:709@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.122:5060;branch=z9hG4bKee8a4e206fcf638aFrom: "Jake Erramouspe" ;tag=6b1f101ca51707efTo: ;tag=as5851220cContact: Proxy-Authorization: Digest username="722", realm="asterisk", algorithm=MD5, uri="sip:709@10.200.26.202", nonce="2b2b3c08", response="e67ef8fcb0d4513103a2ef8b172e6111"Call-ID: fd4df747f847f904@10.200.26.122C asterisk-price*CLI> Seq: 55853 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (12 headers 0 lines) --- asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) asterisk-price*CLI> SIP/709-09864628 (None) Up Bridged Call(SIP/722-0984a588) asterisk-price*CLI> SIP/722-0984a588 709@from-internal:2 Up Dial(SIP/709|20|Tt) asterisk-price*CLI> Zap/6-1 900@from-zaptel:6 Up Parked Call() asterisk-price*CLI> SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) asterisk-price*CLI> SIP/707-097a4ca0 202@parkedcalls:1 Up ParkedCall(202) asterisk-price*CLI> Zap/2-1 900@from-zaptel:6 Up Bridged Call(SIP/707-097a4ca0) asterisk-price*CLI> Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) asterisk-price*CLI> 7 active channels asterisk-price*CLI> 3 active calls asterisk-price*CLI> Retransmitting #6 (NAT) to 10.200.26.109:5060: NOTIFY sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18006479;rportFrom: "DUGOUT CANYON M" ;tag=as2b9ed892To: ;tag=9501098a44676bf1Contact: Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=1032Subscription-state: terminated;reason=noresourceContent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 16SIP/2.0 200 OK --- asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18006479;rportFrom: "DUGOUT CANYON M" ;tag=as2b9ed892To: ;tag=9501098a44676bf1Call-ID: 759c303713d201317a4f437476698b2e@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.122:5060 ---> BYE sip:709@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.122:5060;branch=z9hG4bK5c4b1ad170e0026cFrom: "Jake Erramouspe" ;tag=6b1f101ca51707efTo: ;tag=as5851220cProxy-Authorization: Digest username="722", realm="asterisk", algorithm=MD5, uri="sip:709@10.200.26.202", nonce="2b2b3c08", response="dc12ec6b3b3aff63e43901e7730420e4"Call-ID: fd4df747f847f904@10.200.26.122CSeq: 55854 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.200.26.122 : 5060 (NAT) <--- Transmitting (NAT) to 10.200.26.122:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.122:5060;branch=z9hG4bK5c4b1ad170e0026c;received=10.200.26.122From: "Jake Erramouspe" ;tag=6b1f101ca51707efTo: ;tag=as5851220cCall-ID: fd4df747f847f904@10.200.26.122CSeq: 55854 BYEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2838eeb65b57fab22d0cfd1971038498@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.109, port 5060 Reliably Transmitting (NAT) to 10.200.26.109:5060: BYE sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK27c86f45;rportFrom: "Jake Erramouspe" ;tag=as53f6a942To: ;tag=b59cf46dfd1c7247Call-ID: 2838eeb65b57fab22d0cfd1971038498@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- == Spawn extension (from-internal, 709, 2) exited non-zero on 'SIP/722-0984a588' -- Executing [h@from-internal:1] Hangup("SIP/722-0984a588", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/722-0984a588' asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK27c86f45;rportFrom: "Jake Erramouspe" ;tag=as53f6a942To: ;tag=b59cf46dfd1c7247Call-ID: 2838eeb65b57fab22d0cfd1971038498@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> Really destroying SIP dialog '2838eeb65b57fab22d0cfd1971038498@66.111.122.20' Method: INVITE Really destroying SIP dialog 'fd4df747f847f904@10.200.26.122' Method: BYE asterisk-price*CLI> [Jul 20 12:11:09] WARNING[10061]: chan_sip.c:1920 retrans_pkt: Maximum retries exceeded on transmission 759c303713d201317a4f437476698b2e@66.111.122.20 for seqno 103 (Non-critical Request) asterisk-price*CLI> Really destroying SIP dialog '759c303713d201317a4f437476698b2e@66.111.122.20' Method: REFER asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> INVITE sip:201@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK1d5a00060a44f575From: "Alan Young" ;tag=4eda7630ae1a8b89To: Contact: Supported: replaces, timerCall-ID: 0fe8fbfefb31e9a9@10.200.26.109CSeq: 30779 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 253v=0o=709 8000 8000 IN IP4 10.200.26.109s=SIP Callc=IN IP4 10.200.26.109t=0 0m=audio 5004 RTP/AVP 0 8 4 18 3a=sendrecva=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:4 G723/8000a=rtpmap:18 G729/8000a=rtpmap:3 GSM/8000a=ptime:20 <-------------> --- (13 headers 13 lines) --- Sending to 10.200.26.109 : 5060 (NAT) Using INVITE request as basis request - 0fe8fbfefb31e9a9@10.200.26.109 <--- Reliably Transmitting (NAT) to 10.200.26.109:5060 ---> SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK1d5a00060a44f575;received=10.200.26.109From: "Alan Young" ;tag=4eda7630ae1a8b89To: ;tag=as28e5ef50Call-ID: 0fe8fbfefb31e9a9@10.200.26.109CSeq: 30779 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesProxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7898c210"Content-Length: 0 <------------> Scheduling destruction of SIP dialog '0fe8fbfefb31e9a9@10.200.26.109' in 32000 ms (Method: INVITE) Found user '709' asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> ACK sip:201@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK1d5a00060a44f575From: "Alan Young" ;tag=4eda7630ae1a8b89To: ;tag=as28e5ef50Contact: Call-ID: 0fe8fbfefb31e9a9@10.200.26.109CSeq: 30779 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> INVITE sip:201@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bKb99d0bfba9bcfa9cFrom: "Alan Young" ;tag=4eda7630ae1a8b89To: Contact: Supported: replaces, timerProxy-Authorization: Digest username="709", realm="asterisk", algorithm=MD5, uri="sip:201@10.200.26.202", nonce="7898c210", response="9d38dcc3f894e084eece9bc5257ff5a3"Call-ID: 0fe8fbfefb31e9a9@10.200.26.109CSeq: 30780 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 253v=0o=709 8000 8001 IN IP4 10.200.26.109s=SIP Callc=IN IP4 10.200.26.109t=0 0m=audio 5004 RTP/AVP 0 8 4 18 3a=sendrecva=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:4 G723/8000a=rtpmap:18 G729/8000a=rtpmap:3 GSM/8000a=ptime:20 <-------------> asterisk-price*CLI> --- (14 headers 13 lines) --- Sending to 10.200.26.109 : 5060 (NAT) Using INVITE request as basis request - 0fe8fbfefb31e9a9@10.200.26.109 Found user '709' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 3 Peer audio RTP is at port 10.200.26.109:5004 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G723 for ID 4 Found description format G729 for ID 18 Found description format GSM for ID 3 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.109:5004 Looking for 201 in from-internal (domain 10.200.26.202) list_route: hop: <--- Transmitting (NAT) to 10.200.26.109:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bKb99d0bfba9bcfa9c;received=10.200.26.109From: "Alan Young" ;tag=4eda7630ae1a8b89To: Call-ID: 0fe8fbfefb31e9a9@10.200.26.109CSeq: 30780 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> -- Executing [201@from-internal:1] Goto("SIP/709-0982c0e0", "parkedcalls|201|1") in new stack asterisk-price*CLI> -- Goto (parkedcalls,201,1) asterisk-price*CLI> -- Executing [201@parkedcalls:1] ParkedCall("SIP/709-0982c0e0", "201") in new stack asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.127:5060: NOTIFY sip:727@10.200.26.127:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0e5e9302;rportFrom: ;tag=as0fac1be1To: "Conference Room" ;tag=1e94d3e74ab057e7Contact: Call-ID: cc74970ec691e7e1@10.200.26.127CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: active asterisk-price*CLI> Content-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 727 Reliably Transmitting (NAT) to 10.200.26.131:5060: NOTIFY sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK372cc51e;rportFrom: ;tag=as44d1397bT asterisk-price*CLI> o: "Lobby" ;tag=99f4161d5df43d88Contact: Call-ID: 62279987519ceb08@10.200.26.131CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 731 Reliably Transmitting (NAT) to 10.200.26.130:5060: NOTIFY sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK52b10ac9;rportFrom: ;tag=as73346657To: "Foyer" ;tag=bee43bdcf28e33c2Contact: Call-ID: 0c17b027f44a0b16@10.200.26.130CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 730 Reliably Transmitting (NAT) to 10.200.26.129:5060: NOTIFY sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6d839a7f;rportFrom: ;tag=as0b80a390To: "Will Jaimez" ;tag=18b4544c3b75f992Contact: Call-ID: 46071ba670da2a07@10.200.26.129CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 729 Reliably Transmitting (NAT) to 10.200.26.117:5060: NOTIFY sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK29200917;rportFrom: ;tag=as5ab9c51aTo: "Ken Williams" ;tag=85b710412fc42025Contact: Call-ID: d416754dbbee5bda@10.200.26.117CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 717 Reliably Transmitting (NAT) to 10.200.26.126:5060: NOTIFY sip:726@10.200.26.126:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK29ca7d59;rportFrom: ;tag=as3f56cc30To: "Bob Teny" ;tag=94c4908c7746b742Contact: Call-ID: 8e17d427fa4a6426@10.200.26.126CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 726 Reliably Transmitting (NAT) to 10.200.26.125:5060: NOTIFY sip:725@10.200.26.125:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3ff5f3a6;rportFrom: ;tag=as38f74437To: "Front Desk" ;tag=1bd437ccfd6ebe92Contact: Call-ID: 62473a6832fbae08@10.200.26.125CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 725 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.124:5060: NOTIFY sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK12afd49c;rportFrom: ;tag=as35039685To: "Bobby Houston" ;tag=79c4178cb056f172Contact: Call-ID: 8d27b397d8fca0f7@10.200.26.124CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 724 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.121:5060: NOTIFY sip:721@10.200.26.121:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK74757a69;rportFrom: ;tag=as52c9f68fTo: "Danny D'Ambrosio" ;tag=91e4dffc190f5ae1Contact: Call-ID: 8c47d4781d5ccab6@10.200.26.121CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK372cc51e;rportFrom: ;tag=as44d1397bTo: "Lobby" ;tag=99f4161d5df43d88Call-ID: 62279987519ceb08@10.200.26.131CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK52b10ac9;rportFrom: ;tag=as73346657To: "Foyer" ;tag=bee43bdcf28e33c2Call-ID: 0c17b027f44a0b16@10.200.26.130CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 721 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.122:5060: NOTIFY sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK660c0ee8;rportFrom: ;tag=as3bd2c35dTo: "Jake Erramouspe" ;tag=d0e49efc170f18e1Contact: Call-ID: a94732785b5c49b6@10.200.26.122CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 722 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.120:5060: NOTIFY sip:720@10.200.26.120:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK065fc5d3;rportFrom: ;tag=as26caf817To: "Darek Martinez" ;tag=1ae4770db0f4d3b8Contact: Call-ID: 0067f749305becc6@10.200.26.120CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 720 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.119:5060: NOTIFY sip:719@10.200.26.119:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4b732ef4;rportFrom: ;tag=as4d16fad1To: "Randy Leffler" ;tag=9e94dba4740eb5d1Contact: Call-ID: 2f273697fcfc87f7@10.200.26.119CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6d839a7f;rportFrom: ;tag=as0b80a390To: "Will Jaimez" ;tag=18b4544c3b75f992Call-ID: 46071ba670da2a07@10.200.26.129CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 719 Reliably Transmitting (NAT) to 10.200.26.118:5060: NOTIFY sip:718@10.200.26.118:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK01136daa;rportFrom: ;tag=as501b143eTo: "Chris / Ed" ;tag=f9a4973cd07533c2Contact: Call-ID: 8647dd68f4fb0f08@10.200.26.118CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 718 Reliably Transmitting (NAT) to 10.200.26.116:5060: NOTIFY sip:716@10.200.26.116:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2cb7f76e;rportFrom: ;tag=as71101529To: "Shop Kitchen" ;tag=3ea41b3c727533c2Contact: Call-ID: 484791787a5c06b6@10.200.26.116CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 716 Reliably Transmitting (NAT) to 10.200.26.111:5060: NOTIFY sip:711@10.200.26.111:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4b7df682;rportFrom: ;tag=as4dacd753To: "Center Shop" ;tag=cf7dedb2b53fa60fContact: Call-ID: 43ce35395af9fd6f@10.200.26.111CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 711 Reliably Transmitting (NAT) to 10.200.26.115:5060: NOTIFY sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4895856e;rportFrom: ;tag=as039135acTo: "Jerry Wright" ;tag=34b4537c3d36fe42Contact: Call-ID: af47d878926c2027@10.200.26.115CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 715 Reliably Transmitting (NAT) to 10.200.26.113:5060: NOTIFY sip:713@10.200.26.113:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK50c99c81;rportFrom: ;tag=as05b1e1acTo: "Front Shop" ;tag=37a152d9516ad0eeContact: Call-ID: 49e06e58c4d76003@10.200.26.113CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 713 Reliably Transmitting (NAT) to 10.200.26.112:5060: NOTIFY sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7d207eaa;rportFrom: ;tag=as0b0c2072To: "Lorrie Blake" ;tag=d3947f943676b5a2Contact: Call-ID: 0d175327994ac226@10.200.26.112CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 712 Reliably Transmitting (NAT) to 10.200.26.110:5060: NOTIFY sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK44de24cd;rportFrom: ;tag=as3b8bd8b5To: "Shawn Norton" ;tag=d481dbf1f79a741fContact: Call-ID: 0ca0cf97a319ed9e@10.200.26.110CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 710 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.109:5060: NOTIFY sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2ab237dc;rportFrom: ;tag=as24dbb862To: "Alan Young" ;tag=597133f1324431aeContact: Call-ID: ace04268693825af@10.200.26.109CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 709 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.108:5060: NOTIFY sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d6577f1;rportFrom: ;tag=as23529257To: "Jake Ori" ;tag=3b5457d35e7edd92Contact: Call-ID: ae07f2b6183b61b5@10.200.26.108CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 708 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.107:5060: NOTIFY sip:707@10.200.26.107:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16db7b0f;rportFrom: ;tag=as55ec459bTo: "Tom Akers" ;tag=b864d414bbf47a88Contact: Call-ID: 05271c87d29c4c08@10.200.26.107CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 707 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.106:5060: NOTIFY sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3499aa59;rportFrom: ;tag=as12414dc8To: "Jenine Bentley" ;tag=19cef75d507ad439Contact: Call-ID: ba019788181b78b6@10.200.26.106CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 706 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK20fc1f42;rportFrom: ;tag=as30d10f77To: "Main Office Conf Room" ;tag=5b5475d3ba7e3992Contact: Call-ID: a2171917ddd9a517@10.200.26.105CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 705 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.104:5060: NOTIFY sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3fdc9014;rportFrom: ;tag=as6afb6310To: "Suzy Iorg" ;tag=06f2f98b43a41526Contact: Call-ID: 97b4dff5782a54a2@10.200.26.104CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 704 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.103:5060: NOTIFY sip:703@10.200.26.103:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK677c66c4;rportFrom: ;tag=as0f08e20bTo: "Shelli Marvidakis" ;tag=9984b794f0a6b218Contact: Call-ID: 0a5774e87eaceb46@10.200.26.103CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeC asterisk-price*CLI> ontent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 703 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.102:5060: NOTIFY sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK29cd971d;rportFrom: ;tag=as414071ddTo: "Jason Worley" ;tag=5b84999412a69418Contact: Call-ID: 8a5755e830bc6ea6@10.200.26.102CSeq: 114 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 702 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.101:5060: NOTIFY sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK12af7d05;rportFrom: ;tag=as15c13c13To: "John Houston" ;tag=1d74fa5434d55668Contact: Call-ID: 88479078785cc3b6@10.200.26.101CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> Extension Changed 201 new state Idle for Notify User 701 asterisk-price*CLI> <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK29ca7d59;rportFrom: ;tag=as3f56cc30To: "Bob Teny" ;tag=94c4908c7746b742Call-ID: 8e17d427fa4a6426@10.200.26.126CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3ff5f3a6;rportFrom: ;tag=as38f74437To: "Front Desk" ;tag=1bd437ccfd6ebe92Call-ID: 62473a6832fbae08@10.200.26.125CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK12afd49c;rportFrom: ;tag=as35039685To: "Bobby Houston" ;tag=79c4178cb056f172Call-ID: 8d27b397d8fca0f7@10.200.26.124CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.121:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK74757a69;rportFrom: ;tag=as52c9f68fTo: "Danny D'Ambrosio" ;tag=91e4dffc190f5ae1Call-ID: 8c47d4781d5ccab6@10.200.26.121CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK660c0ee8;rportFrom: ;tag=as3bd2c35dTo: "Jake Erramouspe" ;tag=d0e49efc170f18e1Call-ID: a94732785b5c49b6@10.200.26.122CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.120:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK065fc5d3;rportFrom: ;tag=as26caf817To: "Darek Martinez" ;tag=1ae4770db0f4d3b8Call-ID: 0067f749305becc6@10.200.26.120CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.119:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4b732ef4;rportFrom: ;tag=as4d16fad1To: "Randy Leffler" ;tag=9e94dba4740eb5d1Call-ID: 2f273697fcfc87f7@10.200.26.119CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK01136daa;rportFrom: ;tag=as501b143eTo: "Chris / Ed" ;tag=f9a4973cd07533c2Call-ID: 8647dd68f4fb0f08@10.200.26.118CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.116:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2cb7f76e;rportFrom: ;tag=as71101529To: "Shop Kitchen" ;tag=3ea41b3c727533c2Call-ID: 484791787a5c06b6@10.200.26.116CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4895856e;rportFrom: ;tag=as039135acTo: "Jerry Wright" ;tag=34b4537c3d36fe42Call-ID: af47d878926c2027@10.200.26.115CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.113:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK50c99c81;rportFrom: ;tag=as05b1e1acTo: "Front Shop" ;tag=37a152d9516ad0eeCall-ID: 49e06e58c4d76003@10.200.26.113CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.111:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4b7df682;rportFrom: ;tag=as4dacd753To: "Center Shop" ;tag=cf7dedb2b53fa60fCall-ID: 43ce35395af9fd6f@10.200.26.111CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.2.23Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.127:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0e5e9302;rportFrom: ;tag=as0fac1be1To: "Conference Room" ;tag=1e94d3e74ab057e7Call-ID: cc74970ec691e7e1@10.200.26.127CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7d207eaa;rportFrom: ;tag=as0b0c2072To: "Lorrie Blake" ;tag=d3947f943676b5a2Call-ID: 0d175327994ac226@10.200.26.112CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK44de24cd;rportFrom: ;tag=as3b8bd8b5To: "Shawn Norton" ;tag=d481dbf1f79a741fCall-ID: 0ca0cf97a319ed9e@10.200.26.110CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2ab237dc;rportFrom: ;tag=as24dbb862To: "Alan Young" ;tag=597133f1324431aeCall-ID: ace04268693825af@10.200.26.109CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d6577f1;rportFrom: ;tag=as23529257To: "Jake Ori" ;tag=3b5457d35e7edd92Call-ID: ae07f2b6183b61b5@10.200.26.108CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK29200917;rportFrom: ;tag=as5ab9c51aTo: "Ken Williams" ;tag=85b710412fc42025Call-ID: d416754dbbee5bda@10.200.26.117CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16db7b0f;rportFrom: ;tag=as55ec459bTo: "Tom Akers" ;tag=b864d414bbf47a88Call-ID: 05271c87d29c4c08@10.200.26.107CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3499aa59;rportFrom: ;tag=as12414dc8To: "Jenine Bentley" ;tag=19cef75d507ad439Call-ID: ba019788181b78b6@10.200.26.106CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK20fc1f42;rportFrom: ;tag=as30d10f77To: "Main Office Conf Room" ;tag=5b5475d3ba7e3992Call-ID: a2171917ddd9a517@10.200.26.105CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3fdc9014;rportFrom: ;tag=as6afb6310To: "Suzy Iorg" ;tag=06f2f98b43a41526Call-ID: 97b4dff5782a54a2@10.200.26.104CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK677c66c4;rportFrom: ;tag=as0f08e20bTo: "Shelli Marvidakis" ;tag=9984b794f0a6b218Call-ID: 0a5774e87eaceb46@10.200.26.103CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK29cd971d;rportFrom: ;tag=as414071ddTo: "Jason Worley" ;tag=5b84999412a69418Call-ID: 8a5755e830bc6ea6@10.200.26.102CSeq: 114 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK12af7d05;rportFrom: ;tag=as15c13c13To: "John Houston" ;tag=1d74fa5434d55668Call-ID: 88479078785cc3b6@10.200.26.101CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> Audio is at 10.200.26.202 port 10146 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- Reliably Transmitting (NAT) to 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bKb99d0bfba9bcfa9c;received=10.200.26.109From: "Alan Young" ;tag=4eda7630ae1a8b89To: ;tag=as05587317Call-ID: 0fe8fbfefb31e9a9@10.200.26.109CSeq: 30780 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Type: application/sdpContent-Length: 209v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10146 RTP/AVP 0 3a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=silenceSupp:off - - - -a=ptime:20a=sendrecv <------------> asterisk-price*CLI> -- Stopped music on hold on Zap/6-1 asterisk-price*CLI> -- Channel SIP/709-0982c0e0 connected to parked call 201 asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> ACK sip:201@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK1f1f0b45294201edFrom: "Alan Young" ;tag=4eda7630ae1a8b89To: ;tag=as05587317Contact: Proxy-Authorization: Digest username="709", realm="asterisk", algorithm=MD5, uri="sip:201@10.200.26.202", nonce="7898c210", response="a42abe2aa9e3d234e2b590da89f93509"Call-ID: 0fe8fbfefb31e9a9@10.200.26.109CSeq: 30780 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> asterisk-price*CLI> --- (12 headers 0 lines) --- asterisk-price*CLI> Really destroying SIP dialog '11de51510e007aef4b1998e10871de92@66.111.122.20' Method: INVITE asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> BYE sip:201@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK3d8b67bcc2794b8cFrom: "Alan Young" ;tag=4eda7630ae1a8b89To: ;tag=as05587317Proxy-Authorization: Digest username="709", realm="asterisk", algorithm=MD5, uri="sip:201@10.200.26.202", nonce="7898c210", response="75816a93184942d3e89d2f6fb824f5c0"Call-ID: 0fe8fbfefb31e9a9@10.200.26.109CSeq: 30781 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.200.26.109 : 5060 (NAT) <--- Transmitting (NAT) to 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK3d8b67bcc2794b8c;received=10.200.26.109From: "Alan Young" ;tag=4eda7630ae1a8b89To: ;tag=as05587317Call-ID: 0fe8fbfefb31e9a9@10.200.26.109CSeq: 30781 BYEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> -- Hungup 'Zap/6-1' == Spawn extension (parkedcalls, 201, 1) exited non-zero on 'SIP/709-0982c0e0' asterisk-price*CLI> Really destroying SIP dialog '0fe8fbfefb31e9a9@10.200.26.109' Method: BYE asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> INVITE sip:6508554@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK8e6e40c521a37eeeFrom: "Alan Young" ;tag=7daae6070f6f74d2To: Contact: Supported: replaces, timerCall-ID: feb9b68b57d3f901@10.200.26.109CSeq: 44405 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 253v=0o=709 8000 8000 IN IP4 10.200.26.109s=SIP Callc=IN IP4 10.200.26.109t=0 0m=audio 5004 RTP/AVP 0 8 4 18 3a=sendrecva=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:4 G723/8000a=rtpmap:18 G729/8000a=rtpmap:3 GSM/8000a=ptime:20 <-------------> --- (13 headers 13 lines) --- Sending to 10.200.26.109 : 5060 (NAT) Using INVITE request as basis request - feb9b68b57d3f901@10.200.26.109 <--- Reliably Transmitting (NAT) to 10.200.26.109:5060 ---> SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK8e6e40c521a37eee;received=10.200.26.109From: "Alan Young" ;tag=7daae6070f6f74d2To: ;tag=as03db33c4Call-ID: feb9b68b57d3f901@10.200.26.109CSeq: 44405 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesProxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4192a1df"Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'feb9b68b57d3f901@10.200.26.109' in 32000 ms (Method: INVITE) Found user '709' asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> ACK sip:6508554@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK8e6e40c521a37eeeFrom: "Alan Young" ;tag=7daae6070f6f74d2To: ;tag=as03db33c4Contact: Call-ID: feb9b68b57d3f901@10.200.26.109CSeq: 44405 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> INVITE sip:6508554@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK70a75e29638d92abFrom: "Alan Young" ;tag=7daae6070f6f74d2To: Contact: Supported: replaces, timerProxy-Authorization: Digest username="709", realm="asterisk", algorithm=MD5, uri="sip:6508554@10.200.26.202", nonce="4192a1df", response="73316b8eea6e731b99c01c9b543cd42d"Call-ID: feb9b68b57d3f901@10.200.26.109CSeq: 44406 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 253v=0o=709 8000 8001 IN IP4 10.200.26.109s=SIP Callc=IN IP4 10.200.26.109t=0 0m=audio 5004 RTP/AVP 0 8 4 18 3a=sendrecva=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:4 G723/8000a=rtpmap:18 G729/8000a=rtpmap:3 GSM/8000a=ptime:20 <-------------> --- (14 headers 13 lines) --- Sending to 10.200.26.109 : 5060 (NAT) Using INVITE request as basis request - feb9b68b57d3f901@10.200.26.109 Found user '709' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 3 Peer audio RTP is at port 10.200.26.109:5004 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G723 for ID 4 Found description format G729 for ID 18 Found description format GSM for ID 3 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.109:5004 Looking for 6508554 in from-internal (domain 10.200.26.202) list_route: hop: <--- Transmitting (NAT) to 10.200.26.109:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK70a75e29638d92ab;received=10.200.26.109From: "Alan Young" ;tag=7daae6070f6f74d2To: Call-ID: feb9b68b57d3f901@10.200.26.109CSeq: 44406 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> -- Executing [6508554@from-internal:1] Dial("SIP/709-09864628", "ZAP/g0/6508554|120|rTt") in new stack asterisk-price*CLI> -- Called g0/6508554 <--- Transmitting (NAT) to 10.200.26.109:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK70a75e29638d92ab;received=10.200.26.109From: "Alan Young" ;tag=7daae6070f6f74d2To: ;tag=as04e22e10Call-ID: feb9b68b57d3f901@10.200.26.109CSeq: 44406 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> -- Zap/4-1 answered SIP/709-09864628 Audio is at 10.200.26.202 port 10048 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP <--- Reliably Transmitting (NAT) to 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK70a75e29638d92ab;received=10.200.26.109From: "Alan Young" ;tag=7daae6070f6f74d2To: ;tag=as04e22e10Call-ID: feb9b68b57d3f901@10.200.26.109CSeq: 44406 INVITEUser-Agent: Asterisk PBX asterisk-price*CLI> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Type: application/sdpContent-Length: 209v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10048 RTP/AVP 0 3a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=silenceSupp:off - - - -a=ptime:20a=sendrecv <------------> asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> ACK sip:6508554@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK0eb9ec608ddead05From: "Alan Young" ;tag=7daae6070f6f74d2To: ;tag=as04e22e10Contact: Proxy-Authorization: Digest username="709", realm="asterisk", algorithm=MD5, uri="sip:6508554@10.200.26.202", nonce="4192a1df", response="b77dbaa6c55a2fc44328672b567b3a4c"Call-ID: feb9b68b57d3f901@10.200.26.109CSeq: 44406 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (12 headers 0 lines) --- asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) asterisk-price*CLI> Zap/4-1 (None) Up Bridged Call(SIP/709-09864628) asterisk-price*CLI> SIP/709-09864628 6508554@from-interna Up Dial(ZAP/g0/6508554|120|rTt) asterisk-price*CLI> SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) asterisk-price*CLI> SIP/707-097a4ca0 202@parkedcalls:1 Up ParkedCall(202) asterisk-price*CLI> Zap/2-1 900@from-zaptel:6 Up Bridged Call(SIP/707-097a4ca0) asterisk-price*CLI> Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) asterisk-price*CLI> 6 active channels asterisk-price*CLI> 3 active calls asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> INVITE sip:8884476@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKdc2e6867b1972749From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: Contact: Supported: replaces, timerCall-ID: 3f936ac11193096f@10.200.26.105CSeq: 58891 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 280v=0o=705 8000 8000 IN IP4 10.200.26.105s=SIP Callc=IN IP4 10.200.26.105t=0 0m=audio 5004 RTP/AVP 0 8 4 18 2 3a=sendrecva=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:4 G723/8000a=rtpmap:18 G729/8000a=rtpmap:2 G726-32/8000a=rtpmap:3 GSM/8000a=ptime:20 <-------------> --- (13 headers 14 lines) --- Sending to 10.200.26.105 : 5060 (NAT) Using INVITE request as basis request - 3f936ac11193096f@10.200.26.105 <--- Reliably Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKdc2e6867b1972749;received=10.200.26.105From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as0cc01b5eCall-ID: 3f936ac11193096f@10.200.26.105CSeq: 58891 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesProxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2e618678"Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3f936ac11193096f@10.200.26.105' in 32000 ms (Method: INVITE) Found user '705' asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> ACK sip:8884476@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKdc2e6867b1972749From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as0cc01b5eContact: Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58891 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> INVITE sip:8884476@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKf8743eb14007545eFrom: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: Contact: Supported: replaces, timerProxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:8884476@10.200.26.202", nonce="2e618678", response="6baba15c056330cc1eebc4cd669f2573"Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58892 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 280v=0o=705 8000 8001 IN IP4 10.200.26.105s=SIP Callc=IN IP4 10.200.26.105t=0 0m=audio 5004 RTP/AVP 0 8 4 18 2 3a=sendrecva=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:4 G723/8000a=rtpmap:18 G729/8000a=rtpmap:2 G726-32/8000a=rtpmap:3 GSM/8000a=ptime:20 <-------------> --- (14 headers 14 lines) --- Sending to 10.200.26.105 : 5060 (NAT) Using INVITE request as basis request - 3f936ac11193096f@10.200.26.105 Found user '705' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 3 Peer audio RTP is at port 10.200.26.105:5004 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G723 for ID 4 Found description format G729 for ID 18 Found description format G726-32 for ID 2 Found description format GSM for ID 3 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.105:5004 Looking for 8884476 in from-internal (domain 10.200.26.202) list_route: hop: <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKf8743eb14007545e;received=10.200.26.105From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58892 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> -- Executing [8884476@from-internal:1] Dial("SIP/705-0982c0e0", "ZAP/g0/8884476|120|rTt") in new stack asterisk-price*CLI> -- Called g0/8884476 asterisk-price*CLI> <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKf8743eb14007545e;received=10.200.26.105From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58892 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) Zap/6-1 8884476@from-zaptel: Dialing AppDial((Outgoing Line)) SIP/705-0982c0e0 8884476@from-interna Ring Dial(ZAP/g0/8884476|120|rTt) Zap/4-1 (None) Up Bridged Call(SIP/709-09864628) SIP/709-09864628 6508554@from-interna Up Dial(ZAP/g0/6508554|120|rTt) SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) SIP/707-097a4ca0 202@parkedcalls:1 Up ParkedCall(202) Zap/2-1 900@from-zaptel:6 Up Bridged Call(SIP/707-097a4ca0) Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) 8 active channels 4 active calls asterisk-price*CLI> -- Zap/6-1 answered SIP/705-0982c0e0 Audio is at 10.200.26.202 port 10640 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP <--- Reliably Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKf8743eb14007545e;received=10.200.26.105From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58892 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Type: application/sdpContent-Length: 209v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10640 RTP/AVP 0 3a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=silenceSupp:off - - - -a=ptime:20a=sendrecv <------------> asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> ACK sip:8884476@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKf6dd634aa8b0117dFrom: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Contact: Proxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:8884476@10.200.26.202", nonce="2e618678", response="9fd982afdf564568d4567b256a129723"Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58892 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (12 headers 0 lines) --- asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) asterisk-price*CLI> Zap/6-1 (None) Up Bridged Call(SIP/705-0982c0e0) asterisk-price*CLI> SIP/705-0982c0e0 8884476@from-interna Up Dial(ZAP/g0/8884476|120|rTt) asterisk-price*CLI> Zap/4-1 (None) Up Bridged Call(SIP/709-09864628) asterisk-price*CLI> SIP/709-09864628 6508554@from-interna Up Dial(ZAP/g0/6508554|120|rTt) asterisk-price*CLI> SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) asterisk-price*CLI> SIP/707-097a4ca0 202@parkedcalls:1 Up ParkedCall(202) asterisk-price*CLI> Zap/2-1 900@from-zaptel:6 Up Bridged Call(SIP/707-097a4ca0) asterisk-price*CLI> Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) asterisk-price*CLI> 8 active channels asterisk-price*CLI> 4 active calls asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> INFO sip:8884476@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK7e5b581045b44f4bFrom: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Contact: Proxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:8884476@10.200.26.202", nonce="2e618678", response="918830d551a7675744ef05b07f10ee0d"Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58893 INFOUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/dtmf-relayContent-Length: 22Signal=3Duration=640 <-------------> --- (13 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 3 <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK7e5b581045b44f4b;received=10.200.26.105From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58893 INFOUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> INFO sip:8884476@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKdb8973b41b8c9921From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Contact: Proxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:8884476@10.200.26.202", nonce="2e618678", response="918830d551a7675744ef05b07f10ee0d"Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58894 INFOUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/dtmf-relayContent-Length: 22Signal=1Duration=960 <-------------> --- (13 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 1 <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKdb8973b41b8c9921;received=10.200.26.105From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58894 INFOUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> INFO sip:8884476@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK9f1a42a78f6c5d4bFrom: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Contact: Proxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:8884476@10.200.26.202", nonce="2e618678", response="918830d551a7675744ef05b07f10ee0d"Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58895 INFOUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/dtmf-relayContent-Length: 22Signal=0Duration=800 <-------------> --- (13 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 0 <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK9f1a42a78f6c5d4b;received=10.200.26.105From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58895 INFOUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> INFO sip:8884476@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK94cba7a06aa604f9From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Contact: Proxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:8884476@10.200.26.202", nonce="2e618678", response="918830d551a7675744ef05b07f10ee0d"Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58896 INFOUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/dtmf-relayContent-Length: 22Signal=0Duration=800 <-------------> asterisk-price*CLI> --- (13 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 0 <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK94cba7a06aa604f9;received=10.200.26.105From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58896 INFOUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> INFO sip:8884476@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKae293ff21b4dd9a0From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Contact: Proxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:8884476@10.200.26.202", nonce="2e618678", response="918830d551a7675744ef05b07f10ee0d"Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58897 INFOUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/dtmf-relayContent-Length: 23Signal=10Duration=800 <-------------> --- (13 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: * <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKae293ff21b4dd9a0;received=10.200.26.105From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58897 INFOUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> -- Starting simple switch on 'Zap/7-1' asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> INFO sip:8884476@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK102af5e8285ca782From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Contact: Proxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:8884476@10.200.26.202", nonce="2e618678", response="918830d551a7675744ef05b07f10ee0d"Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58898 INFOUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/dtmf-relayContent-Length: 23Signal=7Duration=1120 <-------------> asterisk-price*CLI> --- (13 headers 2 lines) --- asterisk-price*CLI> Receiving INFO! asterisk-price*CLI> * DTMF-relay event received: 7 asterisk-price*CLI> <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK102af5e8285ca782;received=10.200.26.105From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58898 INFOUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> -- Executing [s@from-zaptel:1] Goto("Zap/7-1", "900|1") in new stack -- Goto (from-zaptel,900,1) -- Executing [900@from-zaptel:1] NoOp("Zap/7-1", "SIP/701&SIP/703&SIP/704&SIP/705&SIP/706&SIP/708&SIP/709&SIP/712&SIP/715&SIP/717&SIP/718&SIP/722&SIP/724&SIP/725&SIP/726&SIP/727&SIP/728&SIP/729&SIP/730&SIP/731") in new stack -- Executing [900@from-zaptel:2] Set("Zap/7-1", "CURRENTDIALEXTENSIONS1=SIP/701&SIP/703&SIP/704&SIP/705&SIP/706&SIP/708&SIP/709&SIP/712&SIP/715&SIP/717&SIP/718&SIP/722&SIP/724&SIP/725&SIP/726&SIP/727&SIP/728&SIP/729&SIP/730&SIP/731&SIP/710") in new stack -- Executing [900@from-zaptel:3] Set("Zap/7-1", "CURRENTDIALEXTENSIONS2=SIP/701&SIP/703&SIP/704&SIP/705&SIP/706&SIP/708&SIP/709&SIP/712&SIP/715&SIP/717&SIP/718&SIP/722&SIP/724&SIP/725&SIP/726&SIP/727&SIP/728&SIP/729&SIP/730&SIP/731&SIP/710&SIP/702") in new stack -- Executing [900@from-zaptel:4] Set("Zap/7-1", "CURRENTDIALEXTENSIONS3=SIP/701&SIP/703&SIP/704&SIP/705&SIP/706&SIP/708&SIP/709&SIP/712&SIP/715&SIP/717&SIP/718&SIP/722&SIP/724&SIP/725&SIP/726&SIP/727&SIP/728&SIP/729&SIP/730&SIP/731&SIP/710&SIP/702&SIP/707") in new stack -- Executing [900@from-zaptel:5] NoOp("Zap/7-1", "SIP/701&SIP/703&SIP/704&SIP/705&SIP/706&SIP/708&SIP/709&SIP/712&SIP/715&SIP/717&SIP/718&SIP/722&SIP/724&SIP/725&SIP/726&SIP/727&SIP/728&SIP/729&SIP/730&SIP/731&SIP/710&SIP/702&SIP/707") in new stack -- Executing [900@from-zaptel:6] Dial("Zap/7-1", "SIP/701&SIP/703&SIP/704&SIP/705&SIP/706&SIP/708&SIP/709&SIP/712&SIP/715&SIP/717&SIP/718&SIP/722&SIP/724&SIP/725&SIP/726&SIP/727&SIP/728&SIP/729&SIP/730&SIP/731&SIP/710&SIP/702&SIP/707|20|Tt") in new stack asterisk-price*CLI> Audio is at 10.200.26.202 port 10958 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.101:5060: INVITE sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5203d1e0;rportFrom: "3046567257" ;tag=as733535c1To: Contact: Call-ID: 72e6faa94ccfbe6f654e78bb6aa6ebd2@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10958 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 701 asterisk-price*CLI> Audio is at 10.200.26.202 port 10234 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.103:5060: INVITE sip:703@10.200.26.103:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK45fc296b;rportFrom: "3046567257" ;tag=as3b7850e1To: Contact: Call-ID: 4d6082a65d35c7191f72ee115ad0d1a0@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10234 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5203d1e0;rportFrom: "3046567257" ;tag=as733535c1To: Call-ID: 72e6faa94ccfbe6f654e78bb6aa6ebd2@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> -- Called 703 asterisk-price*CLI> Audio is at 10.200.26.202 port 10868 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5203d1e0;rportFrom: "3046567257" ;tag=as733535c1To: ;tag=aa65fdf93915721bCall-ID: 72e6faa94ccfbe6f654e78bb6aa6ebd2@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK45fc296b;rportFrom: "3046567257" ;tag=as3b7850e1To: Call-ID: 4d6082a65d35c7191f72ee115ad0d1a0@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK45fc296b;rportFrom: "3046567257" ;tag=as3b7850e1To: ;tag=96065dd2c5f82b67Call-ID: 4d6082a65d35c7191f72ee115ad0d1a0@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.104:5060: INVITE sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0b5a6a2e;rportFrom: "3046567257" ;tag=as6a83f71bTo: Contact: Call-ID: 499efbc501c14c32650ed8a6494391e8@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10868 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 704 asterisk-price*CLI> Audio is at 10.200.26.202 port 10570 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0b5a6a2e;rportFrom: "3046567257" ;tag=as6a83f71bTo: Call-ID: 499efbc501c14c32650ed8a6494391e8@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.105:5060: INVITE sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2f183300;rportFrom: "3046567257" ;tag=as781846c7To: Contact: Call-ID: 4b2eec007e7924a10b393fff40324a13@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10570 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 705 asterisk-price*CLI> Audio is at 10.200.26.202 port 10936 asterisk-price*CLI> <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0b5a6a2e;rportFrom: "3046567257" ;tag=as6a83f71bTo: ;tag=0c8931e65edf3a59Call-ID: 499efbc501c14c32650ed8a6494391e8@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2f183300;rportFrom: "3046567257" ;tag=as781846c7To: Call-ID: 4b2eec007e7924a10b393fff40324a13@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2f183300;rportFrom: "3046567257" ;tag=as781846c7To: ;tag=9691a877e837c03aCall-ID: 4b2eec007e7924a10b393fff40324a13@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.106:5060: INVITE sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4f4416d2;rportFrom: "3046567257" ;tag=as3c42d731To: Contact: Call-ID: 02fde9142b708f205511e3c57c27023b@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10936 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 706 Audio is at 10.200.26.202 port 10558 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4f4416d2;rportFrom: "3046567257" ;tag=as3c42d731To: Call-ID: 02fde9142b708f205511e3c57c27023b@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.108:5060: INVITE sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7032ed3b;rportFrom: "3046567257" ;tag=as1fcefe51To: Contact: Call-ID: 17335b417afd035a471b530f71e69cfe@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10558 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 708 asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4f4416d2;rportFrom: "3046567257" ;tag=as3c42d731To: ;tag=e83660f93bd1cdf5Call-ID: 02fde9142b708f205511e3c57c27023b@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7032ed3b;rportFrom: "3046567257" ;tag=as1fcefe51To: Call-ID: 17335b417afd035a471b530f71e69cfe@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Audio is at 10.200.26.202 port 10248 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7032ed3b;rportFrom: "3046567257" ;tag=as1fcefe51To: ;tag=a185805521c5cea5Call-ID: 17335b417afd035a471b530f71e69cfe@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.109:5060: INVITE sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7127e680;rportFrom: "3046567257" ;tag=as3633515bTo: Contact: Call-ID: 2cdac56c5c6c55e417e461ef409d1b89@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10248 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 709 asterisk-price*CLI> Audio is at 10.200.26.202 port 10676 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.112:5060: INVITE sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK323a5bb0;rportFrom: "3046567257" ;tag=as16c0533fTo: Contact: Call-ID: 4ccd463d034f709b7b643cd840d71ac8@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10676 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7127e680;rportFrom: "3046567257" ;tag=as3633515bTo: Call-ID: 2cdac56c5c6c55e417e461ef409d1b89@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7127e680;rportFrom: "3046567257" ;tag=as3633515bTo: ;tag=f851ec27ee9855c9Call-ID: 2cdac56c5c6c55e417e461ef409d1b89@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> -- Called 712 asterisk-price*CLI> Audio is at 10.200.26.202 port 10098 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.115:5060: INVITE sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK652485cc;rportFrom: "3046567257" ;tag=as0f8ddf72To: Contact: Call-ID: 0c946b490a50dae26fc6f9dc2394f469@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10098 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK323a5bb0;rportFrom: "3046567257" ;tag=as16c0533fTo: Call-ID: 4ccd463d034f709b7b643cd840d71ac8@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> -- Called 715 asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK323a5bb0;rportFrom: "3046567257" ;tag=as16c0533fTo: ;tag=b55611151e252845Call-ID: 4ccd463d034f709b7b643cd840d71ac8@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK652485cc;rportFrom: "3046567257" ;tag=as0f8ddf72To: Call-ID: 0c946b490a50dae26fc6f9dc2394f469@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Audio is at 10.200.26.202 port 10206 asterisk-price*CLI> <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK652485cc;rportFrom: "3046567257" ;tag=as0f8ddf72To: ;tag=8989d69995ccc17aCall-ID: 0c946b490a50dae26fc6f9dc2394f469@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.117:5060: INVITE sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK33f8d843;rportFrom: "3046567257" ;tag=as4012d33dTo: Contact: Call-ID: 36a0bbd41626733f5afaa3640a75bf5c@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10206 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 717 asterisk-price*CLI> Audio is at 10.200.26.202 port 10964 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.118:5060: INVITE sip:718@10.200.26.118:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0bd13a76;rportFrom: "3046567257" ;tag=as11beab83To: Contact: Call-ID: 468f5b182758f63327957f6002a2da2c@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10964 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 718 asterisk-price*CLI> Audio is at 10.200.26.202 port 10336 asterisk-price*CLI> <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0bd13a76;rportFrom: "3046567257" ;tag=as11beab83To: Call-ID: 468f5b182758f63327957f6002a2da2c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0bd13a76;rportFrom: "3046567257" ;tag=as11beab83To: ;tag=92544fe29d67e864Call-ID: 468f5b182758f63327957f6002a2da2c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.122:5060: INVITE sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK516590b5;rportFrom: "3046567257" ;tag=as71cefeabTo: Contact: Call-ID: 6b5b184f4d7bb38614ab2763251e2d28@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10336 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 722 asterisk-price*CLI> Audio is at 10.200.26.202 port 10184 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.124:5060: INVITE sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0644e8b4;rportFrom: "3046567257" ;tag=as41075f38To: Contact: Call-ID: 63f93ebf3d2f2ed3219e5ee7463fd90e@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10184 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 724 asterisk-price*CLI> Audio is at 10.200.26.202 port 10858 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK516590b5;rportFrom: "3046567257" ;tag=as71cefeabTo: Call-ID: 6b5b184f4d7bb38614ab2763251e2d28@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK33f8d843;rportFrom: "3046567257" ;tag=as4012d33dTo: Call-ID: 36a0bbd41626733f5afaa3640a75bf5c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK516590b5;rportFrom: "3046567257" ;tag=as71cefeabTo: ;tag=abbe7309097443a9Call-ID: 6b5b184f4d7bb38614ab2763251e2d28@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK33f8d843;rportFrom: "3046567257" ;tag=as4012d33dTo: ;tag=5a4a3eab336b5925Call-ID: 36a0bbd41626733f5afaa3640a75bf5c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0644e8b4;rportFrom: "3046567257" ;tag=as41075f38To: Call-ID: 63f93ebf3d2f2ed3219e5ee7463fd90e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.124:5060 ---> asterisk-price*CLI> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0644e8b4;rportFrom: "3046567257" ;tag=as41075f38To: ;tag=91b85178a21b31b7Call-ID: 63f93ebf3d2f2ed3219e5ee7463fd90e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.125:5060: INVITE sip:725@10.200.26.125:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK36276216;rportFrom: "3046567257" ;tag=as7db3acb5To: Contact: Call-ID: 141fb5e702a101594634309005ee6202@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10858 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 725 asterisk-price*CLI> Audio is at 10.200.26.202 port 10370 asterisk-price*CLI> <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK36276216;rportFrom: "3046567257" ;tag=as7db3acb5To: Call-ID: 141fb5e702a101594634309005ee6202@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK36276216;rportFrom: "3046567257" ;tag=as7db3acb5To: ;tag=0aa93919d9e9a79bCall-ID: 141fb5e702a101594634309005ee6202@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.126:5060: INVITE sip:726@10.200.26.126:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK15b99abc;rportFrom: "3046567257" ;tag=as30546140To: Contact: Call-ID: 5f6607a7035511c56e5f6ef24b50691e@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10370 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 726 asterisk-price*CLI> Audio is at 10.200.26.202 port 10002 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.127:5062: INVITE sip:727@10.200.26.127:5062;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16a80f8d;rportFrom: "3046567257" ;tag=as6993c1adTo: Contact: Call-ID: 2b80f0cc26970ac8399186b30d291e28@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10002 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 727 asterisk-price*CLI> <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK15b99abc;rportFrom: "3046567257" ;tag=as30546140To: Call-ID: 5f6607a7035511c56e5f6ef24b50691e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK15b99abc;rportFrom: "3046567257" ;tag=as30546140To: ;tag=2304e14568c04cb4Call-ID: 5f6607a7035511c56e5f6ef24b50691e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Audio is at 10.200.26.202 port 10980 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5ea50161;rportFrom: "3046567257" ;tag=as2f677472To: Contact: Call-ID: 5f556141707977527d01ca3b4a6137ed@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10980 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 728 asterisk-price*CLI> Audio is at 10.200.26.202 port 10910 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.129:5060: INVITE sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK69ae819a;rportFrom: "3046567257" ;tag=as7db9b16eTo: Contact: Call-ID: 088b79831d246c642184592149793a47@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10910 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> <--- SIP read from 10.200.26.127:5062 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16a80f8d;rportFrom: "3046567257" ;tag=as6993c1adTo: Call-ID: 2b80f0cc26970ac8399186b30d291e28@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> -- Called 729 asterisk-price*CLI> Audio is at 10.200.26.202 port 10974 asterisk-price*CLI> <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK69ae819a;rportFrom: "3046567257" ;tag=as7db9b16eTo: Call-ID: 088b79831d246c642184592149793a47@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK69ae819a;rportFrom: "3046567257" ;tag=as7db9b16eTo: ;tag=2789349973cce07aCall-ID: 088b79831d246c642184592149793a47@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.127:5062 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16a80f8d;rportFrom: "3046567257" ;tag=as6993c1adTo: ;tag=66c762e379acf9acCall-ID: 2b80f0cc26970ac8399186b30d291e28@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.130:5060: INVITE sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16a9a4e7;rportFrom: "3046567257" ;tag=as7fdedfa9To: Contact: Call-ID: 0ec320170bdeb06b63f4e4234680d738@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10974 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 730 asterisk-price*CLI> Audio is at 10.200.26.202 port 10878 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.131:5060: INVITE sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK40356299;rportFrom: "3046567257" ;tag=as138f9936To: Contact: Call-ID: 0a19a8986195bb9260a74deb625bba71@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10878 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16a9a4e7;rportFrom: "3046567257" ;tag=as7fdedfa9To: Call-ID: 0ec320170bdeb06b63f4e4234680d738@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16a9a4e7;rportFrom: "3046567257" ;tag=as7fdedfa9To: ;tag=0c75ff6a1b23f38bCall-ID: 0ec320170bdeb06b63f4e4234680d738@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> -- Called 731 asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK40356299;rportFrom: "3046567257" ;tag=as138f9936To: Call-ID: 0a19a8986195bb9260a74deb625bba71@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> Audio is at 10.200.26.202 port 10196 asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK40356299;rportFrom: "3046567257" ;tag=as138f9936To: ;tag=b4e432e49dbee3afCall-ID: 0a19a8986195bb9260a74deb625bba71@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.110:5060: INVITE sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6f180ede;rportFrom: "3046567257" ;tag=as450e5588To: Contact: Call-ID: 7ca6d76f64e517b761ce7f6203d61435@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10196 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 710 asterisk-price*CLI> Audio is at 10.200.26.202 port 10566 asterisk-price*CLI> <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6f180ede;rportFrom: "3046567257" ;tag=as450e5588To: Call-ID: 7ca6d76f64e517b761ce7f6203d61435@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6f180ede;rportFrom: "3046567257" ;tag=as450e5588To: ;tag=ce799a29381a63a8Call-ID: 7ca6d76f64e517b761ce7f6203d61435@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.102:5060: INVITE sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK20f312b3;rportFrom: "3046567257" ;tag=as39293e6cTo: Contact: Call-ID: 1f13a32d3df5c22e7e7640170536970d@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10566 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 702 asterisk-price*CLI> Audio is at 10.200.26.202 port 10880 asterisk-price*CLI> <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK20f312b3;rportFrom: "3046567257" ;tag=as39293e6cTo: Call-ID: 1f13a32d3df5c22e7e7640170536970d@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK20f312b3;rportFrom: "3046567257" ;tag=as39293e6cTo: ;tag=c3b9d489d69c65eaCall-ID: 1f13a32d3df5c22e7e7640170536970d@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.107:5060: INVITE sip:707@10.200.26.107:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3f49811a;rportFrom: "3046567257" ;tag=as462ea666To: Contact: Call-ID: 375d3abf4dfcd1a92ad099220d92c29e@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10880 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 707 asterisk-price*CLI> -- SIP/701-0983adc8 is ringing asterisk-price*CLI> -- SIP/703-09842c20 is ringing asterisk-price*CLI> -- SIP/704-0981c8d8 is ringing asterisk-price*CLI> -- SIP/705-097c9618 is ringing asterisk-price*CLI> -- SIP/706-097bf390 is ringing asterisk-price*CLI> -- SIP/708-0984a588 is ringing asterisk-price*CLI> -- SIP/709-09840390 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3f49811a;rportFrom: "3046567257" ;tag=as462ea666To: Call-ID: 375d3abf4dfcd1a92ad099220d92c29e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> -- SIP/712-09825a50 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3f49811a;rportFrom: "3046567257" ;tag=as462ea666To: ;tag=ee1dc09748feb2a3Call-ID: 375d3abf4dfcd1a92ad099220d92c29e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> -- SIP/715-097cf358 is ringing asterisk-price*CLI> -- SIP/717-098691d0 is ringing asterisk-price*CLI> -- SIP/718-09816518 is ringing asterisk-price*CLI> -- SIP/722-09859998 is ringing asterisk-price*CLI> -- SIP/724-097b8718 is ringing asterisk-price*CLI> -- SIP/725-09854a00 is ringing asterisk-price*CLI> -- SIP/726-0984f450 is ringing asterisk-price*CLI> -- SIP/727-098509d0 is ringing asterisk-price*CLI> -- SIP/729-09937368 is ringing asterisk-price*CLI> -- SIP/730-0993c8a8 is ringing asterisk-price*CLI> -- SIP/731-09941ae8 is ringing asterisk-price*CLI> -- SIP/710-09830e80 is ringing asterisk-price*CLI> -- SIP/702-098c5970 is ringing asterisk-price*CLI> -- SIP/707-098c98d8 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> INFO sip:8884476@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK4a758ee39c1da563From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Contact: Proxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:8884476@10.200.26.202", nonce="2e618678", response="918830d551a7675744ef05b07f10ee0d"Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58899 INFOUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/dtmf-relayContent-Length: 22Signal=2Duration=800 <-------------> asterisk-price*CLI> --- (13 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 2 <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK4a758ee39c1da563;received=10.200.26.105From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58899 INFOUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> Retransmitting #1 (NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5ea50161;rportFrom: "3046567257" ;tag=as2f677472To: Contact: Call-ID: 5f556141707977527d01ca3b4a6137ed@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10980 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> Retransmitting #2 (NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5ea50161;rportFrom: "3046567257" ;tag=as2f677472To: Contact: Call-ID: 5f556141707977527d01ca3b4a6137ed@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10980 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> show channels<--- SIP read from 10.200.26.105:5060 ---> INFO sip:8884476@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK102a74fc018b822cFrom: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Contact: Proxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:8884476@10.200.26.202", nonce="2e618678", response="918830d551a7675744ef05b07f10ee0d"Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58900 INFOUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/dtmf-relayContent-Length: 22Signal=3Duration=960 <-------------> --- (13 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 3 <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK102a74fc018b822c;received=10.200.26.105From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58900 INFOUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> show channels Retransmitting #3 (NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5ea50161;rportFrom: "3046567257" ;tag=as2f677472To: Contact: Call-ID: 5f556141707977527d01ca3b4a6137ed@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10980 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> show channels Retransmitting #4 (NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5ea50161;rportFrom: "3046567257" ;tag=as2f677472To: Contact: Call-ID: 5f556141707977527d01ca3b4a6137ed@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:13:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10980 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> show channels<--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK36276216;rportFrom: "3046567257" ;tag=as7db3acb5To: ;tag=0aa93919d9e9a79bCall-ID: 141fb5e702a101594634309005ee6202@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=725 8000 8000 IN IP4 10.200.26.125s=SIP Callc=IN IP4 10.200.26.125t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.125:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.125:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.125, port 5060 Transmitting (NAT) to 10.200.26.125:5060: ACK sip:725@10.200.26.125:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK20a67d77;rportFrom: "3046567257" ;tag=as7db3acb5To: ;tag=0aa93919d9e9a79bContact: Call-ID: 141fb5e702a101594634309005ee6202@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- -- SIP/725-09854a00 answered Zap/7-1 Scheduling destruction of SIP dialog '375d3abf4dfcd1a92ad099220d92c29e@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.107:5060: CANCEL sip:707@10.200.26.107:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3f49811a;rportFrom: "3046567257" ;tag=as462ea666To: Call-ID: 375d3abf4dfcd1a92ad099220d92c29e@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '375d3abf4dfcd1a92ad099220d92c29e@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '1f13a32d3df5c22e7e7640170536970d@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.102:5060: CANCEL sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK20f312b3;rportFrom: "3046567257" ;tag=as39293e6cTo: Call-ID: 1f13a32d3df5c22e7e7640170536970d@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '1f13a32d3df5c22e7e7640170536970d@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '7ca6d76f64e517b761ce7f6203d61435@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.110:5060: CANCEL sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6f180ede;rportFrom: "3046567257" ;tag=as450e5588To: Call-ID: 7ca6d76f64e517b761ce7f6203d61435@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '7ca6d76f64e517b761ce7f6203d61435@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '0a19a8986195bb9260a74deb625bba71@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.131:5060: CANCEL sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK40356299;rportFrom: "3046567257" ;tag=as138f9936T asterisk-price*CLI> show channels o: Call-ID: 0a19a8986195bb9260a74deb625bba71@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '0a19a8986195bb9260a74deb625bba71@66.111.122.20' in 32000 ms (Method: INVITE) <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3f49811a;rportFrom: "3046567257" ;tag=as462ea666To: ;tag=ee1dc09748feb2a3Call-ID: 375d3abf4dfcd1a92ad099220d92c29e@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK20f312b3;rportFrom: "3046567257" ;tag=as39293e6cTo: ;tag=c3b9d489d69c65eaCall-ID: 1f13a32d3df5c22e7e7640170536970d@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3f49811a;rportFrom: "3046567257" ;tag=as462ea666To: ;tag=ee1dc09748feb2a3Call-ID: 375d3abf4dfcd1a92ad099220d92c29e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.107:5060: ACK sip:707@10.200.26.107:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3f49811a;rportFrom: "3046567257" ;tag=as462ea666To: ;tag=ee1dc09748feb2a3Contact: Call-ID: 375d3abf4dfcd1a92ad099220d92c29e@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6f180ede;rportFrom: "3046567257" ;tag=as450e5588To: ;tag=ce799a29381a63a8Call-ID: 7ca6d76f64e517b761ce7f6203d61435@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK40356299;rportFrom: "3046567257" ;tag=as138f9936To: ;tag=b4e432e49dbee3afCall-ID: 0a19a8986195bb9260a74deb625bba71@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '375d3abf4dfcd1a92ad099220d92c29e@66.111.122.20' Method: INVITE Scheduling destruction of SIP dialog '0ec320170bdeb06b63f4e4234680d738@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.130:5060: CANCEL sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16a9a4e7;rportFrom: "3046567257" ;tag=as7fdedfa9To: Call-ID: 0ec320170bdeb06b63f4e4234680d738@66.111.122.20CSeq: 102 CANCEL asterisk-price*CLI> show channels User-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK20f312b3;rportFrom: "3046567257" ;tag=as39293e6cTo: ;tag=c3b9d489d69c65eaCall-ID: 1f13a32d3df5c22e7e7640170536970d@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.102:5060: ACK sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK20f312b3;rportFrom: "3046567257" ;tag=as39293e6cTo: ;tag=c3b9d489d69c65eaContact: Call-ID: 1f13a32d3df5c22e7e7640170536970d@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6f180ede;rportFrom: "3046567257" ;tag=as450e5588To: ;tag=ce799a29381a63a8Call-ID: 7ca6d76f64e517b761ce7f6203d61435@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.110:5060: ACK sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6f180ede;rportFrom: "3046567257" ;tag=as450e5588To: ;tag=ce799a29381a63a8Contact: Call-ID: 7ca6d76f64e517b761ce7f6203d61435@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK40356299;rportFrom: "3046567257" ;tag=as138f9936To: ;tag=b4e432e49dbee3afCall-ID: 0a19a8986195bb9260a74deb625bba71@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.131:5060: ACK sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK40356299;rportFrom: "3046567257" ;tag=as138f9936To: ;tag=b4e432e49dbee3afContact: Call-ID: 0a19a8986195bb9260a74deb625bba71@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '1f13a32d3df5c22e7e7640170536970d@66.111.122.20' Method: INVITE Really destroying SIP dialog '7ca6d76f64e517b761ce7f6203d61435@66.111.122.20' Method: INVITE Really destroying SIP dialog '0a19a8986195bb9260a74deb625bba71@66.111.122.20' Method: INVITE Scheduling destruction of SIP dialog '0ec320170bdeb06b63f4e4234680d738@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '088b79831d246c642184592149793a47@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.129:5060: CANCEL sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK69ae819a;rportFrom: "3046567257" ;tag=as7db9b16eTo: Call-ID: 088b79831d246c642184592149793a47@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '088b79831d246c642184592149793a47@66.111.122.20' in 32000 ms (Method: INVITE) <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16a9a4e7;rportFrom: "3046567257" ;tag=as7fdedfa9To: ;tag=0c75ff6a1b23f38bCall-ID: 0ec320170bdeb06b63f4e4234680d738@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Scheduling destruction of SIP dialog '5f556141707977527d01ca3b4a6137ed@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '2b80f0cc26970ac8399186b30d291e28@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.127:5062: CANCEL sip:727@10.200.26.127:5062;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16a80f8d;rportFrom: "3046567257" ;tag=as6993c1adTo: Call-ID: 2b80f0cc26970ac8399186b30d291e28@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '2b80f0cc26970ac8399186b30d291e28@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '5f6607a7035511c56e5f6ef24b50691e@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.126:5060: CANCEL sip:726@10.200.26.126:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK15b99abc;rportFrom: "3046567257" ;tag=as30546140To: Call-ID: 5f6607a7035511c56e5f6ef24b50691e@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '5f6607a7035511c56e5f6ef24b50691e@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '63f93ebf3d2f2ed3219e5ee7463fd90e@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.124:5060: CANCEL sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0644e8b4;rportFrom: "3046567257" ;tag=as41075f38To: Call-ID: 63f93ebf3d2f2ed3219e5ee7463fd90e@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK69ae819a;rportFrom: "3046567257" ;tag=as7db9b16eTo: ;tag=2789349973cce07aCall-ID: 088b79831d246c642184592149793a47@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16a9a4e7;rportFrom: "3046567257" ;tag=as7fdedfa9To: ;tag=0c75ff6a1b23f38bCall-ID: 0ec320170bdeb06b63f4e4234680d738@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.130:5060: ACK sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16a9a4e7;rportFrom: "3046567257" ;tag=as7fdedfa9To: ;tag=0c75ff6a1b23f38bContact: Call-ID: 0ec320170bdeb06b63f4e4234680d738@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '0ec320170bdeb06b63f4e4234680d738@66.111.122.20' Method: INVITE Scheduling destruction of SIP dialog '63f93ebf3d2f2ed3219e5ee7463fd90e@66.111.122.20' in 32000 ms (Method: INVITE) <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK15b99abc;rportFrom: "3046567257" ;tag=as30546140To: ;tag=2304e14568c04cb4Call-ID: 5f6607a7035511c56e5f6ef24b50691e@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0644e8b4;rportFrom: "3046567257" ;tag=as41075f38To: ;tag=91b85178a21b31b7Call-ID: 63f93ebf3d2f2ed3219e5ee7463fd90e@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK69ae819a;rportFrom: "3046567257" ;tag=as7db9b16eTo: ;tag=2789349973cce07aCall-ID: 088b79831d246c642184592149793a47@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.129:5060: ACK sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK69ae819a;rportFrom: "3046567257" ;tag=as7db9b16eTo: ;tag=2789349973cce07aContact: Call-ID: 088b79831d246c642184592149793a47@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '088b79831d246c642184592149793a47@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK15b99abc;rportFrom: "3046567257" ;tag=as30546140To: ;tag=2304e14568c04cb4Call-ID: 5f6607a7035511c56e5f6ef24b50691e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.126:5060: ACK sip:726@10.200.26.126:5060 SIP/2.0V asterisk-price*CLI> show channels ia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK15b99abc;rportFrom: "3046567257" ;tag=as30546140To: ;tag=2304e14568c04cb4Contact: Call-ID: 5f6607a7035511c56e5f6ef24b50691e@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0644e8b4;rport asterisk-price*CLI> show channels From: "3046567257" ;tag=as41075f38To: ;tag=91b85178a21b31b7Call-ID: 63f93ebf3d2f2ed3219e5ee7463fd90e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.124:5060: ACK sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0644e8b4;rportFrom: "3046567257" ;tag=as41075f38To: ;tag=91b85178a21b31b7Contact: Call-ID: 63f93ebf3d2f2ed3219e5ee7463fd90e@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '6b5b184f4d7bb38614ab2763251e2d28@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.122:5060: CANCEL sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK516590b5;rportFrom: "3046567257" ;tag=as71cefeabTo: Call-ID: 6b5b184f4d7bb38614ab2763251e2d28@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '6b5b184f4d7bb38614ab2763251e2d28@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '468f5b182758f63327957f6002a2da2c@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.118:5060: CANCEL sip:718@10.200.26.118:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0bd13a76;rportFrom: "3046567257" ;tag=as11beab83To: Call-ID: 468f5b182758f63327957f6002a2da2c@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '468f5b182758f63327957f6002a2da2c@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '36a0bbd41626733f5afaa3640a75bf5c@66.111.122.20' in 32000 ms (Method: INVITE) <--- SIP read from 10.200.26.127:5062 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16a80f8d;rportFrom: "3046567257" ;tag=as6993c1adTo: ;tag=66c762e379acf9acCall-ID: 2b80f0cc26970ac8399186b30d291e28@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.4.18Supported: replaces, timerContent-Length: 0 <-------------> Reliably Transmitting (NAT) to 10.200.26.117:5060: CANCEL sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK33f8d843;rportFrom: "3046567257" ;tag=as4012d33dTo: Call-ID: 36a0bbd41626733f5afaa3640a75bf5c@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '36a0bbd41626733f5afaa3640a75bf5c@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '0c946b490a50dae26fc6f9dc2394f469@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.115:5060: CANCEL sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK652485cc;rportFrom: "3046567257" ;tag=as0f8ddf72To: Call-ID: 0c946b490a50dae26fc6f9dc2394f469@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '0c946b490a50dae26fc6f9dc2394f469@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '4ccd463d034f709b7b643cd840d71ac8@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.112:5060: CANCEL sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK323a5bb0;rportFrom: "3046567257" ;tag=as16c0533fTo: Call-ID: 4ccd463d034f709b7b643cd840d71ac8@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '4ccd463d034f709b7b643cd840d71ac8@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '2cdac56c5c6c55e417e461ef409d1b89@66.111.122.20' in 32000 ms (Method: INVITE) --- (9 headers 0 lines) --- Really destroying SIP dialog '5f6607a7035511c56e5f6ef24b50691e@66.111.122.20' Method: INVITE Really destroying SIP dialog '63f93ebf3d2f2ed3219e5ee7463fd90e@66.111.122.20' Method: INVITE Reliably Transmitting (NAT) to 10.200.26.109:5060: CANCEL sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7127e680;rportFrom: "3046567257" ;tag=as3633515bTo: Call-ID: 2cdac56c5c6c55e417e461ef409d1b89@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '2cdac56c5c6c55e417e461ef409d1b89@66.111.122.20' in 32000 ms (Method: INVITE) <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK516590b5;rportFrom: "3046567257" ;tag=as71cefeabTo: ;tag=abbe7309097443a9Call-ID: 6b5b184f4d7bb38614ab2763251e2d28@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.127:5062 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16a80f8d;rportFrom: "3046567257" ;tag=as6993c1adTo: ;tag=66c762e379acf9acCall-ID: 2b80f0cc26970ac8399186b30d291e28@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.127:5062: ACK sip:727@10.200.26.127:5062;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK16a80f8d;rportFrom: "3046567257" ;tag=as6993c1adTo: ;tag=66c762e379acf9acContact: Call-ID: 2b80f0cc26970ac8399186b30d291e28@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0bd13a76;rportFrom: "3046567257" ;tag=as11beab83To: ;tag=92544fe29d67e864Call-ID: 468f5b182758f63327957f6002a2da2c@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '2b80f0cc26970ac8399186b30d291e28@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK652485cc;rportFrom: "3046567257" ;tag=as0f8ddf72To: ;tag=8989d69995ccc17aCall-ID: 0c946b490a50dae26fc6f9dc2394f469@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK323a5bb0;rportFrom: "3046567257" ;tag=as16c0533fTo: ;tag=b55611151e252845Call-ID: 4ccd463d034f709b7b643cd840d71ac8@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK516590b5;rportFrom: "3046567257" ;tag=as71cefeabTo: ;tag=abbe7309097443a9Call-ID: 6b5b184f4d7bb38614ab2763251e2d28@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.122:5060: ACK sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK516590b5;rportFrom: "3046567257" ;tag=as71cefeabTo: ;tag=abbe7309097443a9Contact: Call-ID: 6b5b184f4d7bb38614ab2763251e2d28@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '6b5b184f4d7bb38614ab2763251e2d28@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0bd13a76;rportFrom: "3046567257" ;tag=as11beab83To: ;tag=92544fe29d67e864Call-ID: 468f5b182758f63327957f6002a2da2c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.118:5060: ACK sip:718@10.200.26.118:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0bd13a76;rportFrom: "3046567257" ;tag=as11beab83To: ;tag=92544fe29d67e864Contact: Call-ID: 468f5b182758f63327957f6002a2da2c@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7127e680;rportFrom: "3046567257" ;tag=as3633515bTo: ;tag=f851ec27ee9855c9Call-ID: 2cdac56c5c6c55e417e461ef409d1b89@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK652485cc;rportFrom: "3046567257" ;tag=as0f8ddf72To: ;tag=8989d69995ccc17aCall-ID: 0c946b490a50dae26fc6f9dc2394f469@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.115:5060: ACK sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK652485cc;rportFrom: "3046567257" ;tag=as0f8ddf72To: ;tag=8989d69995ccc17aContact: Call-ID: 0c946b490a50dae26fc6f9dc2394f469@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '468f5b182758f63327957f6002a2da2c@66.111.122.20' Method: INVITE Really destroying SIP dialog '0c946b490a50dae26fc6f9dc2394f469@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7127e680;rportFrom: "3046567257" ;tag=as3633515bTo: ;tag=f851ec27ee9855c9Call-ID: 2cdac56c5c6c55e417e461ef409d1b89@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.109:5060: ACK sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7127e680;rportFrom: "3046567257" ;tag=as3633515bTo: ;tag=f851ec27ee9855c9Contact: Call-ID: 2cdac56c5c6c55e417e461ef409d1b89@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '17335b417afd035a471b530f71e69cfe@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.108:5060: CANCEL sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7032ed3b;rportFrom: "3046567257" ;tag=as1fcefe51To: Call-ID: 17335b417afd035a471b530f71e69cfe@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '17335b417afd035a471b530f71e69cfe@66.111.122.20' in 32000 ms (Method: INVITE) <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK323a5bb0;rportFrom: "3046567257" ;tag=as16c0533fTo: ;tag=b55611151e252845Call-ID: 4ccd463d034f709b7b643cd840d71ac8@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.112:5060: ACK sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK323a5bb0;rportFrom: "3046567257" ;tag=as16c0533fTo: ;tag=b55611151e252845Contact: Call-ID: 4ccd463d034f709b7b643cd840d71ac8@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK33f8d843;rportFrom: "3046567257" ;tag=as4012d33dTo: ;tag=5a4a3eab336b5925Call-ID: 36a0bbd41626733f5afaa3640a75bf5c@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.4.18Supported: replaces, timerContent-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '4ccd463d034f709b7b643cd840d71ac8@66.111.122.20' Method: INVITE Really destroying SIP dialog '2cdac56c5c6c55e417e461ef409d1b89@66.111.122.20' Method: INVITE asterisk-price*CLI> show channels<--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK33f8d843;rportFrom: "3046567257" ;tag=as4012d33dTo: ;tag=5a4a3eab336b5925Call-ID: 36a0bbd41626733f5afaa3640a75bf5c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.117:5060: ACK sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK33f8d843;rportFrom: "3046567257" ;tag=as4012d33dTo: ;tag=5a4a3eab336b5925Contact: Call-ID: 36a0bbd41626733f5afaa3640a75bf5c@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '02fde9142b708f205511e3c57c27023b@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.106:5060: CANCEL sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4f4416d2;rportFrom: "3046567257" ;tag=as3c42d731To: Call-ID: 02fde9142b708f205511e3c57c27023b@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '02fde9142b708f205511e3c57c27023b@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '4b2eec007e7924a10b393fff40324a13@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.105:5060: CANCEL sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2f183300;rportFrom: "3046567257" ;tag=as781846c7To: Call-ID: 4b2eec007e7924a10b393fff40324a13@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '4b2eec007e7924a10b393fff40324a13@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '499efbc501c14c32650ed8a6494391e8@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.104:5060: CANCEL sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0b5a6a2e;rportFrom: "3046567257" ;tag=as6a83f71bTo: Call-ID: 499efbc501c14c32650ed8a6494391e8@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '499efbc501c14c32650ed8a6494391e8@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '4d6082a65d35c7191f72ee115ad0d1a0@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.103:5060: CANCEL sip:703@10.200.26.103:5060 SIP/2.0 asterisk-price*CLI> show channels Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK45fc296b;rportFrom: "3046567257" ;tag=as3b7850e1To: Call-ID: 4d6082a65d35c7191f72ee115ad0d1a0@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '4d6082a65d35c7191f72ee115ad0d1a0@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '72e6faa94ccfbe6f654e78bb6aa6ebd2@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.101:5060: CANCEL sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5203d1e0;rportFrom: "3046567257" ;tag=as733535c1To: Call-ID: 72e6faa94ccfbe6f654e78bb6aa6ebd2@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7032ed3b;rportFrom: "3046567257" ;tag=as1fcefe51To: ;tag=a185805521c5cea5Call-ID: 17335b417afd035a471b530f71e69cfe@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7032ed3b;rportFrom: "3046567257" ;tag=as1fcefe51To: ;tag=a185805521c5cea5Call-ID: 17335b417afd035a471b530f71e69cfe@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.108:5060: ACK sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7032ed3b;rportFrom: "3046567257" ;tag=as1fcefe51To: ;tag=a185805521c5cea5Contact: Call-ID: 17335b417afd035a471b530f71e69cfe@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '36a0bbd41626733f5afaa3640a75bf5c@66.111.122.20' Method: INVITE Really destroying SIP dialog '17335b417afd035a471b530f71e69cfe@66.111.122.20' Method: INVITE Scheduling destruction of SIP dialog '72e6faa94ccfbe6f654e78bb6aa6ebd2@66.111.122.20' in 32000 ms (Method: INVITE) <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4f4416d2;rportFrom: "3046567257" ;tag=as3c42d731To: ;tag=e83660f93bd1cdf5Call-ID: 02fde9142b708f205511e3c57c27023b@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2f183300;rportFrom: "3046567257" ;tag=as781846c7To: ;tag=9691a877e837c03aCall-ID: 4b2eec007e7924a10b393fff40324a13@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0b5a6a2e;rportFrom: "3046567257" ;tag=as6a83f71bTo: ;tag=0c8931e65edf3a59Call-ID: 499efbc501c14c32650ed8a6494391e8@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2f183300;rportFrom: "3046567257" ;tag=as781846c7To: ;tag=9691a877e837c03aCall-ID: 4b2eec007e7924a10b393fff40324a13@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.105:5060: ACK sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2f183300;rportFrom: "3046567257" ;tag=as781846c7To: ;tag=9691a877e837c03aContact: Call-ID: 4b2eec007e7924a10b393fff40324a13@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK45fc296b;rportFrom: "3046567257" ;tag=as3b7850e1To: ;tag=96065dd2c5f82b67Call-ID: 4d6082a65d35c7191f72ee115ad0d1a0@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4f4416d2;rportFrom: "3046567257" ;tag=as3c42d731To: ;tag=e83660f93bd1cdf5Call-ID: 02fde9142b708f205511e3c57c27023b@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.106:5060: ACK sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4f4416d2;rportFrom: "3046567257" ;tag=as3c42d731To: ;tag=e83660f93bd1cdf5Contact: Call-ID: 02fde9142b708f205511e3c57c27023b@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '02fde9142b708f205511e3c57c27023b@66.111.122.20' Method: INVITE Really destroying SIP dialog '4b2eec007e7924a10b393fff40324a13@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0b5a6a2e;rportFrom: "3046567257" ;tag=as6a83f71bTo: ;tag=0c8931e65edf3a59Call-ID: 499efbc501c14c32650ed8a6494391e8@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.104:5060: ACK sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0b5a6a2e;rportFrom: "3046567257" ;tag=as6a83f71bTo: ;tag=0c8931e65edf3a59Contact: Call-ID: 499efbc501c14c32650ed8a6494391e8@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK45fc296b;rportFrom: "3046567257" ;tag=as3b7850e1To: ;tag=96065dd2c5f82b67Call-ID: 4d6082a65d35c7191f72ee115ad0d1a0@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.103:5060: ACK sip:703@10.200.26.103:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK45fc296b;rportFrom: "3046567257" ;tag=as3b7850e1To: ;tag=96065dd2c5f82b67Contact: Call-ID: 4d6082a65d35c7191f72ee115ad0d1a0@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5203d1e0;rportFrom: "3046567257" ;tag=as733535c1To: ;tag=aa65fdf93915721bCall-ID: 72e6faa94ccfbe6f654e78bb6aa6ebd2@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '499efbc501c14c32650ed8a6494391e8@66.111.122.20' Method: INVITE Really destroying SIP dialog '4d6082a65d35c7191f72ee115ad0d1a0@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5203d1e0;rportFrom: "3046567257" ;tag=as733535c1To: ;tag=aa65fdf93915721bCall-ID: 72e6faa94ccfbe6f654e78bb6aa6ebd2@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.101:5060: ACK sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5203d1e0;rportFrom: "3046567257" ;tag=as733535c1To: ;tag=aa65fdf93915721bContact: Call-ID: 72e6faa94ccfbe6f654e78bb6aa6ebd2@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '72e6faa94ccfbe6f654e78bb6aa6ebd2@66.111.122.20' Method: INVITE asterisk-price*CLI> show channels<--- SIP read from 10.200.26.105:5060 ---> BYE sip:8884476@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKe2799b316fbacb69From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Proxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:8884476@10.200.26.202", nonce="2e618678", response="c68675bdb9863b4013866509d6831b74"Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58901 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.200.26.105 : 5060 (NAT) <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKe2799b316fbacb69;received=10.200.26.105From: "Main Office Conf Room" ;tag=018dad03cafbdd8fTo: ;tag=as5502b556Call-ID: 3f936ac11193096f@10.200.26.105CSeq: 58901 BYEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> show channels -- Hungup 'Zap/6-1' == Spawn extension (from-internal, 8884476, 1) exited non-zero on 'SIP/705-0982c0e0' -- Executing [h@from-internal:1] Hangup("SIP/705-0982c0e0", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/705-0982c0e0' asterisk-price*CLI> show channels Really destroying SIP dialog '3f936ac11193096f@10.200.26.105' Method: BYE asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) SIP/725-09854a00 (None) Up Bridged Call(Zap/7-1) Zap/7-1 900@from-zaptel:6 Up Dial(SIP/701&SIP/703&SIP/704&S Zap/4-1 (None) Up Bridged Call(SIP/709-09864628) SIP/709-09864628 6508554@from-interna Up Dial(ZAP/g0/6508554|120|rTt) SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) SIP/707-097a4ca0 202@parkedcalls:1 Up ParkedCall(202) Zap/2-1 900@from-zaptel:6 Up Bridged Call(SIP/707-097a4ca0) Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) 8 active channels 4 active calls asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> BYE sip:6508554@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK3d2b3179012cef63From: "Alan Young" ;tag=7daae6070f6f74d2To: ;tag=as04e22e10Proxy-Authorization: Digest username="709", realm="asterisk", algorithm=MD5, uri="sip:6508554@10.200.26.202", nonce="4192a1df", response="f8fc8c3333cb09a77354841317282e81"Call-ID: feb9b68b57d3f901@10.200.26.109CSeq: 44407 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.200.26.109 : 5060 (NAT) <--- Transmitting (NAT) to 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.109:5060;branch=z9hG4bK3d2b3179012cef63;received=10.200.26.109From: "Alan Young" ;tag=7daae6070f6f74d2To: ;tag=as04e22e10Call-ID: feb9b68b57d3f901@10.200.26.109CSeq: 44407 BYEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> -- Hungup 'Zap/4-1' == Spawn extension (from-internal, 6508554, 1) exited non-zero on 'SIP/709-09864628' -- Executing [h@from-internal:1] Hangup("SIP/709-09864628", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/709-09864628' asterisk-price*CLI> Really destroying SIP dialog 'feb9b68b57d3f901@10.200.26.109' Method: BYE asterisk-price*CLI> Really destroying SIP dialog '5f556141707977527d01ca3b4a6137ed@66.111.122.20' Method: INVITE asterisk-price*CLI> <--- SIP read from 10.200.26.125:5060 ---> BYE sip:3046567257@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.125:5060;branch=z9hG4bK8d74066c2154e5b4From: ;tag=0aa93919d9e9a79bTo: "3046567257" ;tag=as7db3acb5Call-ID: 141fb5e702a101594634309005ee6202@66.111.122.20CSeq: 61900 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.200.26.125 : 5060 (NAT) <--- Transmitting (NAT) to 10.200.26.125:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.125:5060;branch=z9hG4bK8d74066c2154e5b4;received=10.200.26.125From: ;tag=0aa93919d9e9a79bTo: "3046567257" ;tag=as7db3acb5Call-ID: 141fb5e702a101594634309005ee6202@66.111.122.20CSeq: 61900 BYEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> == Spawn extension (from-zaptel, 900, 6) exited non-zero on 'Zap/7-1' -- Executing [h@from-zaptel:1] Hangup("Zap/7-1", "") in new stack == Spawn extension (from-zaptel, h, 1) exited non-zero on 'Zap/7-1' -- Hungup 'Zap/7-1' asterisk-price*CLI> Really destroying SIP dialog '141fb5e702a101594634309005ee6202@66.111.122.20' Method: BYE asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) asterisk-price*CLI> SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) asterisk-price*CLI> SIP/707-097a4ca0 202@parkedcalls:1 Up ParkedCall(202) asterisk-price*CLI> Zap/2-1 900@from-zaptel:6 Up Bridged Call(SIP/707-097a4ca0) asterisk-price*CLI> Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) asterisk-price*CLI> 4 active channels asterisk-price*CLI> 2 active calls asterisk-price*CLI> sip show channelsasterisk-price*CLI> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message asterisk-price*CLI> 10.200.26.117 717 4b560a236b5 00102/00000 ulaw No Tx: ACK asterisk-price*CLI> 10.200.26.107 707 5453d6fdd72 00101/02042 ulaw No Rx: ACK asterisk-price*CLI> 2 active SIP channels asterisk-price*CLI> sip show channelsh<--- SIP read from 10.200.26.107:5060 ---> BYE sip:202@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.107:5060;branch=z9hG4bKcce480e3b7439daaFrom: "Tom Akers" ;tag=e52b9f8825d68620To: ;tag=as1219f5daProxy-Authorization: Digest username="707", realm="asterisk", algorithm=MD5, uri="sip:202@10.200.26.202", nonce="7f438b98", response="aad4d758cdf301fbb6b8a6405dc80e37"Call-ID: 5453d6fdd727d8c9@10.200.26.107CSeq: 2043 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.200.26.107 : 5060 (NAT) <--- Transmitting (NAT) to 10.200.26.107:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.107:5060;branch=z9hG4bKcce480e3b7439daa;received=10.200.26.107From: "Tom Akers" ;tag=e52b9f8825d68620To: ;tag=as1219f5daCall-ID: 5453d6fdd727d8c9@10.200.26.107CSeq: 2043 BYEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> -- Hungup 'Zap/2-1' == Spawn extension (parkedcalls, 202, 1) exited non-zero on 'SIP/707-097a4ca0' asterisk-price*CLI> show channels Really destroying SIP dialog '5453d6fdd727d8c9@10.200.26.107' Method: BYE asterisk-price*CLI> show channels -- Starting simple switch on 'Zap/2-1' asterisk-price*CLI> show channels -- Executing [s@from-zaptel:1] Goto("Zap/2-1", "900|1") in new stack -- Goto (from-zaptel,900,1) -- Executing [900@from-zaptel:1] NoOp("Zap/2-1", "SIP/701&SIP/703&SIP/704&SIP/705&SIP/706&SIP/708&SIP/709&SIP/712&SIP/715&SIP/717&SIP/718&SIP/722&SIP/724&SIP/725&SIP/726&SIP/727&SIP/728&SIP/729&SIP/730&SIP/731") in new stack -- Executing [900@from-zaptel:2] Set("Zap/2-1", "CURRENTDIALEXTENSIONS1=SIP/701&SIP/703&SIP/704&SIP/705&SIP/706&SIP/708&SIP/709&SIP/712&SIP/715&SIP/717&SIP/718&SIP/722&SIP/724&SIP/725&SIP/726&SIP/727&SIP/728&SIP/729&SIP/730&SIP/731&SIP/710") in new stack -- Executing [900@from-zaptel:3] Set("Zap/2-1", "CURRENTDIALEXTENSIONS2=SIP/701&SIP/703&SIP/704&SIP/705&SIP/706&SIP/708&SIP/709&SIP/712&SIP/715&SIP/717&SIP/718&SIP/722&SIP/724&SIP/725&SIP/726&SIP/727&SIP/728&SIP/729&SIP/730&SIP/731&SIP/710&SIP/702") in new stack -- Executing [900@from-zaptel:4] Set("Zap/2-1", "CURRENTDIALEXTENSIONS3=SIP/701&SIP/703&SIP/704&SIP/705&SIP/706&SIP/708&SIP/709&SIP/712&SIP/715&SIP/717&SIP/718&SIP/722&SIP/724&SIP/725&SIP/726&SIP/727&SIP/728&SIP/729&SIP/730&SIP/731&SIP/710&SIP/702&SIP/707") in new stack -- Executing [900@from-zaptel:5] NoOp("Zap/2-1", "SIP/701&SIP/703&SIP/704&SIP/705&SIP/706&SIP/708&SIP/709&SIP/712&SIP/715&SIP/717&SIP/718&SIP/722&SIP/724&SIP/725&SIP/726&SIP/727&SIP/728&SIP/729&SIP/730&SIP/731&SIP/710&SIP/702&SIP/707") in new stack -- Executing [900@from-zaptel:6] Dial("Zap/2-1", "SIP/701&SIP/703&SIP/704&SIP/705&SIP/706&SIP/708&SIP/709&SIP/712&SIP/715&SIP/717&SIP/718&SIP/722&SIP/724&SIP/725&SIP/726&SIP/727&SIP/728&SIP/729&SIP/730&SIP/731&SIP/710&SIP/702&SIP/707|20|Tt") in new stack Audio is at 10.200.26.202 port 10688 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.101:5060: INVITE sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3972f4e2;rportFrom: "8014525335" ;tag=as1568398dTo: Contact: Call-ID: 3e25c493181c50652235f8fe03cbb490@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10688 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 701 Audio is at 10.200.26.202 port 10026 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.103:5060: INVITE sip:703@10.200.26.103:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3b90e921;rportFrom: "8014525335" ;tag=as63fdb7b7To: Contact: Call-ID: 581f9c973a03e74436aabeb31b8fd66a@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10026 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 703 Audio is at 10.200.26.202 port 10560 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.104:5060: INVITE sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1586759f;rportFrom: "8014525335" ;tag=as5863900fTo: Contact: Call-ID: 7bd1c72c39327a2b58461d58163bdf87@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10560 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3972f4e2;rportFrom: "8014525335" ;tag=as1568398dTo: Call-ID: 3e25c493181c50652235f8fe03cbb490@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3972f4e2;rportFrom: "8014525335" ;tag=as1568398dTo: ;tag=0ad5d52de844d478Call-ID: 3e25c493181c50652235f8fe03cbb490@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> asterisk-price*CLI> show channels --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3b90e921;rportFrom: "8014525335" ;tag=as63fdb7b7To: Call-ID: 581f9c973a03e74436aabeb31b8fd66a@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 180 RingingV asterisk-price*CLI> show channels ia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3b90e921;rportFrom: "8014525335" ;tag=as63fdb7b7To: ;tag=885596bb8f8c9b54Call-ID: 581f9c973a03e74436aabeb31b8fd66a@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1586759f;rportFrom: "8014525335" ;tag=as5863900fTo: Call-ID: 7bd1c72c39327a2b58461d58163bdf87@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- Called 704 Audio is at 10.200.26.202 port 10056 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.105:5060: INVITE sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK600665b8;rportFrom: "8014525335" ;tag=as614446beTo: Contact: Call-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10056 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> show channels -- Called 705 Audio is at 10.200.26.202 port 10800 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.106:5060: INVITE sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18c3bed9;rportFrom: "8014525335" ;tag=as00245baaTo: Contact: Call-ID: 5cd4159962e780b05bf08ca54b444b3c@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10800 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 706 <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1586759f;rportFrom: "8014525335" ;tag=as5863900fTo: ;tag=e1d91c116dd75e6bCall-ID: 7bd1c72c39327a2b58461d58163bdf87@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> asterisk-price*CLI> show channels --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK600665b8;rportFrom: "8014525335" ;tag=as614446beTo: Call-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 180 RingingV asterisk-price*CLI> show channels ia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK600665b8;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbCall-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> show channels Audio is at 10.200.26.202 port 10100 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.108:5060: INVITE sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d4e4674;rportFrom: "8014525335" ;tag=as70fd90b5To: Contact: Call-ID: 18b5aae679f8e0f84bcb5d5f34c42ae1@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10100 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 708 Audio is at 10.200.26.202 port 10580 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.109:5060: INVITE sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6b428b72;rportFrom: "8014525335" ;tag=as2ba931d9To: Contact: Call-ID: 2709ad5d203efe7931f2ef424c9ef70b@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10580 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18c3bed9;rportFrom: "8014525335" ;tag=as00245baaTo: Call-ID: 5cd4159962e780b05bf08ca54b444b3c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> show channels --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18c3bed9;rportFrom: "8014525335" ;tag=as00245baaTo: ;tag=5bc5ee18620f519bCall-ID: 5cd4159962e780b05bf08ca54b444b3c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> show channels -- Called 709 Audio is at 10.200.26.202 port 10564 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.112:5060: INVITE sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5b8956a6;rportFrom: "8014525335" ;tag=as1a70a7d8To: Contact: Call-ID: 77058eb86663c0bf247c12b04e72f0ee@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10564 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d4e4674;rportFrom: "8014525335" ;tag=as70fd90b5To: Call-ID: 18b5aae679f8e0f84bcb5d5f34c42ae1@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d4e4674;rportFrom: "8014525335" ;tag=as70fd90b5To: ;tag=2ca4297b03bbd653Call-ID: 18b5aae679f8e0f84bcb5d5f34c42ae1@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> show channels<--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6b428b72;rportFrom: "8014525335" ;tag=as2ba931d9To: Call-ID: 2709ad5d203efe7931f2ef424c9ef70b@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6b428b72;rportFrom: "8014525335" ;tag=as2ba931d9To: ;tag=8aecef63ef17630dCall-ID: 2709ad5d203efe7931f2ef424c9ef70b@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- -- Called 712 Audio is at 10.200.26.202 port 10764 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.115:5060: INVITE sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d74be68;rportFrom: "8014525335" ;tag=as16e85ad1To: Contact: Call-ID: 49273abf515268df002b21d261901603@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10764 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 715 <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5b8956a6;rportFrom: "8014525335" ;tag=as1a70a7d8To: Call-ID: 77058eb86663c0bf247c12b04e72f0ee@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> show channels --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5b8956a6;rportFrom: "8014525335" ;tag=as1a70a7d8To: ;tag=abd90701efbec24aCall-ID: 77058eb86663c0bf247c12b04e72f0ee@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d74be68;rportFrom: "8014525335" ;tag=as16e85ad1To: Call-ID: 49273abf515268df002b21d261901603@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> Audio is at 10.200.26.202 port 10234 asterisk-price*CLI> show channels Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.117:5060: INVITE sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7fb94961;rportFrom: "8014525335" ;tag=as18ae777aTo: Contact: Call-ID: 623c8d53297c330112f1d1ad003324e9@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10234 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 717 Audio is at 10.200.26.202 port 10076 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.118:5060: INVITE sip:718@10.200.26.118:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK07ad3000;rportFrom: "8014525335" ;tag=as66eb57a0To: Contact: Call-ID: 1e16a7ba5be35d6e505ad1d660621ca7@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10076 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 718 Audio is at 10.200.26.202 port 10570 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> show channels Reliably Transmitting (NAT) to 10.200.26.122:5060: INVITE sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK51266eb2;rportFrom: "8014525335" ;tag=as635f6292To: Contact: Call-ID: 2577cddf74ecb70f1ca7bd985a9ea3a8@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10570 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 722 Audio is at 10.200.26.202 port 10092 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.124:5060: INVITE sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK443c6cba;rportFrom: "8014525335" ;tag=as67af68c8To: Contact: Call-ID: 260d46712d1f57243552d01f0ca4830b@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10092 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 724 Audio is at 10.200.26.202 port 10492 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.125:5060: INVITE sip:725@10.200.26.125:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7bab5ceb;rportFrom: "8014525335" ;tag=as0f8d9c16To: Contact: Call-ID: 21c7d96f7a31e7332e8c819d2e2b6742@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10492 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 725 asterisk-price*CLI> show channels Audio is at 10.200.26.202 port 10222 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.126:5060: INVITE sip:726@10.200.26.126:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2aec2310;rportFrom: "8014525335" ;tag=as424157afTo: Contact: Call-ID: 03fd70a67d97542542fc95033d2600c2@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10222 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 726 --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d74be68;rportFrom: "8014525335" ;tag=as16e85ad1To: ;tag=88f944e1ccb18f5aCall-ID: 49273abf515268df002b21d261901603@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK07ad3000;rportFrom: "8014525335" ;tag=as66eb57a0To: Call-ID: 1e16a7ba5be35d6e505ad1d660621ca7@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK07ad3000;rportFrom: "8014525335" ;tag=as66eb57a0To: ;tag=63237d37e48736d1Call-ID: 1e16a7ba5be35d6e505ad1d660621ca7@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK51266eb2;rportFrom: "8014525335" ;tag=as635f6292To: Call-ID: 2577cddf74ecb70f1ca7bd985a9ea3a8@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK51266eb2;rportFrom: "8014525335" ;tag=as635f6292To: ;tag=83c5d44b69f22d5cCall-ID: 2577cddf74ecb70f1ca7bd985a9ea3a8@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK443c6cba;rportFrom: "8014525335" ;tag=as67af68c8To: Call-ID: 260d46712d1f57243552d01f0ca4830b@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> show channels Audio is at 10.200.26.202 port 10864 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.127:5062: INVITE sip:727@10.200.26.127:5062;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6c339a68;rportFrom: "8014525335" ;tag=as2eb6bd6fTo: Contact: Call-ID: 315fa8bf5c07b72838a2b4d261dca988@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10864 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 727 Audio is at 10.200.26.202 port 10880 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2d2fac63;rportFrom: "8014525335" ;tag=as7c559428To: Contact: Call-ID: 1b85e9a7609ea1e74a336cd7023b28b5@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10880 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 728 Audio is at 10.200.26.202 port 10660 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.129:5060: INVITE sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK402bf19d;rportFrom: "8014525335" ;tag=as7aa604e9To: Contact: Call-ID: 0c98965a369fd60f60d237956c94af1f@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10660 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16 asterisk-price*CLI> show channels a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> show channels --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK443c6cba;rportFrom: "8014525335" ;tag=as67af68c8To: ;tag=80d92bf003f125bbCall-ID: 260d46712d1f57243552d01f0ca4830b@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7bab5ceb;rportFrom: "8014525335" ;tag=as0f8d9c16To: Call-ID: 21c7d96f7a31e7332e8c819d2e2b6742@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7fb94961;rportFrom: "8014525335" ;tag=as18ae777aTo: Call-ID: 623c8d53297c330112f1d1ad003324e9@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7bab5ceb;rportFrom: "8014525335" ;tag=as0f8d9c16To: ;tag=7237aba9d156f977Call-ID: 21c7d96f7a31e7332e8c819d2e2b6742@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2aec2310;rportFrom: "8014525335" ;tag=as424157afTo: Call-ID: 03fd70a67d97542542fc95033d2600c2@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2aec2310;rportFrom: "8014525335" ;tag=as424157afTo: ;tag=e4f9fff162ea91bbCall-ID: 03fd70a67d97542542fc95033d2600c2@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7fb94961;rportFrom: "8014525335" ;tag=as18ae777aTo: ;tag=f713694cfa362d5aCall-ID: 623c8d53297c330112f1d1ad003324e9@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> -- Called 729 asterisk-price*CLI> show channels Audio is at 10.200.26.202 port 10252 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.130:5060: INVITE sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK753eac34;rportFrom: "8014525335" ;tag=as11a97179To: Contact: Call-ID: 4d4eefa21b4907a67c34f1565d53fff0@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10252 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK402bf19d;rportFrom: "8014525335" ;tag=as7aa604e9To: Call-ID: 0c98965a369fd60f60d237956c94af1f@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK402bf19d;rportFrom: "8014525335" ;tag=as7aa604e9To: ;tag=63c9ad8085a108c9Call-ID: 0c98965a369fd60f60d237956c94af1f@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.127:5062 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6c339a68;rportFrom: "8014525335" ;tag=as2eb6bd6fTo: Call-ID: 315fa8bf5c07b72838a2b4d261dca988@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Content-Length: 0 <-------------> asterisk-price*CLI> show channels -- Called 730 Audio is at 10.200.26.202 port 10022 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.131:5060: INVITE sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK063bbde9;rportFrom: "8014525335" ;tag=as32280dc8To: Contact: Call-ID: 6990abcc12416e630b4d02d164bab3e6@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10022 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- --- (8 headers 0 lines) --- asterisk-price*CLI> show channels<--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK753eac34;rportFrom: "8014525335" ;tag=as11a97179To: Call-ID: 4d4eefa21b4907a67c34f1565d53fff0@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK753eac34;rportFrom: "8014525335" ;tag=as11a97179To: ;tag=09c952b0c239d832Call-ID: 4d4eefa21b4907a67c34f1565d53fff0@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.127:5062 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6c339a68;rportFrom: "8014525335" ;tag=as2eb6bd6fTo: ;tag=4275227583281ceaCall-ID: 315fa8bf5c07b72838a2b4d261dca988@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- -- Called 731 Audio is at 10.200.26.202 port 10284 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.110:5060: INVITE sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3ae0bf56;rportFrom: "8014525335" ;tag=as167df14eTo: Contact: Call-ID: 4203595b58bd72720c1f49221a1fc37a@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10284 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 710 asterisk-price*CLI> show channels<--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK063bbde9;rportFrom: "8014525335" ;tag=as32280dc8To: Call-ID: 6990abcc12416e630b4d02d164bab3e6@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK063bbde9;rportFrom: "8014525335" ;tag=as32280dc8To: ;tag=387b89739ce29e6bCall-ID: 6990abcc12416e630b4d02d164bab3e6@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- Audio is at 10.200.26.202 port 10872 asterisk-price*CLI> show channels Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.102:5060: INVITE sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK412768d4;rportFrom: "8014525335" ;tag=as675a62adTo: Contact: Call-ID: 1d3d0edc6b0fb15a6e3523f87dff19a0@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10872 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3ae0bf56;rportFrom: "8014525335" ;tag=as167df14eTo: Call-ID: 4203595b58bd72720c1f49221a1fc37a@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3ae0bf56;rportFrom: "8014525335" ;tag=as167df14eTo: ;tag=48d92401cdbec04aCall-ID: 4203595b58bd72720c1f49221a1fc37a@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> show channels -- Called 702 asterisk-price*CLI> show channels Audio is at 10.200.26.202 port 10434 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.107:5060: INVITE sip:707@10.200.26.107:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2c6d2ffd;rportFrom: "8014525335" ;tag=as43e7fb47To: Contact: Call-ID: 50d380237e76b23f5aa665f8650180f4@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10434 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called 707 -- SIP/701-097a4ca0 is ringing -- SIP/703-09835ed0 is ringing -- SIP/704-09864628 is ringing -- SIP/705-0982c0e0 is ringing asterisk-price*CLI> show channels -- SIP/706-0983adc8 is ringing -- SIP/708-0991d8c0 is ringing -- SIP/709-09842c20 is ringing -- SIP/712-098625b8 is ringing -- SIP/715-098211c8 is ringing -- SIP/717-0981c8d8 is ringing -- SIP/718-097c9618 is ringing -- SIP/722-097bf390 is ringing -- SIP/724-0984a588 is ringing -- SIP/725-09854a00 is ringing -- SIP/726-09840390 is ringing -- SIP/727-09825a50 is ringing -- SIP/729-09939d20 is ringing -- SIP/730-097cf358 is ringing -- SIP/731-098691d0 is ringing <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK412768d4;rportFrom: "8014525335" ;tag=as675a62adTo: Call-ID: 1d3d0edc6b0fb15a6e3523f87dff19a0@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> show channels --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK412768d4;rportFrom: "8014525335" ;tag=as675a62adTo: ;tag=2fb9c820406f62a8Call-ID: 1d3d0edc6b0fb15a6e3523f87dff19a0@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2c6d2ffd;rportFrom: "8014525335" ;tag=as43e7fb47To: Call-ID: 50d380237e76b23f5aa665f8650180f4@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2c6d2ffd;rportFrom: "8014525335" ;tag=as43e7fb47To: ;tag=2db9ac6ab04a3d92Call-ID: 50d380237e76b23f5aa665f8650180f4@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/710-09816518 is ringing -- SIP/702-097b8718 is ringing -- SIP/707-0991ae00 is ringing asterisk-price*CLI> show channels Retransmitting #1 (NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2d2fac63;rportFrom: "8014525335" ;tag=as7c559428To: Contact: Call-ID: 1b85e9a7609ea1e74a336cd7023b28b5@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10880 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> show channels Retransmitting #2 (NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2d2fac63;rportFrom: "8014525335" ;tag=as7c559428To: Contact: Call-ID: 1b85e9a7609ea1e74a336cd7023b28b5@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:15:43 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10880 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> show channels<--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK600665b8;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbCall-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=705 8000 8000 IN IP4 10.200.26.105s=SIP Callc=IN IP4 10.200.26.105t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.105:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.105:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.105, port 5060 Transmitting (NAT) to 10.200.26.105:5060: ACK sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5090dfd7;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbContact: Call-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- -- SIP/705-0982c0e0 answered Zap/2-1 Scheduling destruction of SIP dialog '50d380237e76b23f5aa665f8650180f4@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.107:5060: CANCEL sip:707@10.200.26.107:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2c6d2ffd;rportFrom: "8014525335" ;tag=as43e7fb47To: Call-ID: 50d380237e76b23f5aa665f8650180f4@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '50d380237e76b23f5aa665f8650180f4@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '1d3d0edc6b0fb15a6e3523f87dff19a0@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.102:5060: CANCEL sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK412768d4;rportFrom: "8014525335" ;tag=as675a62adTo: Call-ID: 1d3d0edc6b0fb15a6e3523f87dff19a0@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '1d3d0edc6b0fb15a6e3523f87dff19a0@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '4203595b58bd72720c1f49221a1fc37a@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.110:5060: CANCEL sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3ae0bf56;rportFrom: "8014525335" ;tag=as167df14eTo: Call-ID: 4203595b58bd72720c1f49221a1fc37a@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '4203595b58bd72720c1f49221a1fc37a@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '6990abcc12416e630b4d02d164bab3e6@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.131:5060: CANCEL sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK063bbde9;rportFrom: "8014525335" ;tag=as32280dc8T asterisk-price*CLI> show channels o: Call-ID: 6990abcc12416e630b4d02d164bab3e6@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '6990abcc12416e630b4d02d164bab3e6@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '4d4eefa21b4907a67c34f1565d53fff0@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.130:5060: CANCEL sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK753eac34;rportFrom: "8014525335" ;tag=as11a97179To: Call-ID: 4d4eefa21b4907a67c34f1565d53fff0@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '4d4eefa21b4907a67c34f1565d53fff0@66.111.122.20' in 32000 ms (Method: INVITE) <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2c6d2ffd;rportFrom: "8014525335" ;tag=as43e7fb47To: ;tag=2db9ac6ab04a3d92Call-ID: 50d380237e76b23f5aa665f8650180f4@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> Scheduling destruction of SIP dialog '0c98965a369fd60f60d237956c94af1f@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.129:5060: CANCEL sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK402bf19d;rportFrom: "8014525335" ;tag=as7aa604e9To: Call-ID: 0c98965a369fd60f60d237956c94af1f@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '0c98965a369fd60f60d237956c94af1f@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '1b85e9a7609ea1e74a336cd7023b28b5@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '315fa8bf5c07b72838a2b4d261dca988@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.127:5062: CANCEL sip:727@10.200.26.127:5062;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6c339a68;rportFrom: "8014525335" ;tag=as2eb6bd6fTo: Call-ID: 315fa8bf5c07b72838a2b4d261dca988@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '315fa8bf5c07b72838a2b4d261dca988@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '03fd70a67d97542542fc95033d2600c2@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.126:5060: CANCEL sip:726@10.200.26.126:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2aec2310;rportFrom: "8014525335" ;tag=as424157afTo: Call-ID: 03fd70a67d97542542fc95033d2600c2@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '03fd70a67d97542542fc95033d2600c2@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '21c7d96f7a31e7332e8c819d2e2b6742@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.125:5060: CANCEL sip:725@10.200.26.125:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7bab5ceb;rportFrom: "8014525335" ;tag=as0f8d9c16To: Call-ID: 21c7d96f7a31e7332e8c819d2e2b6742@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '21c7d96f7a31e7332e8c819d2e2b6742@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '260d46712d1f57243552d01f0ca4830b@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.124:5060: CANCEL sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK443c6cba;rportFrom: "8014525335" ;tag=as67af68c8To: Call-ID: 260d46712d1f57243552d01f0ca4830b@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '260d46712d1f57243552d01f0ca4830b@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '2577cddf74ecb70f1ca7bd985a9ea3a8@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.122:5060: CANCEL sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK51266eb2;rportFrom: "8014525335" ;tag=as635f6292To: Call-ID: 2577cddf74ecb70f1ca7bd985a9ea3a8@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '2577cddf74ecb70f1ca7bd985a9ea3a8@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '1e16a7ba5be35d6e505ad1d660621ca7@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.118:5060: CANCEL sip:718@10.200.26.118:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK07ad3000;rportFrom: "8014525335" ;tag=as66eb57a0To: Call-ID: 1e16a7ba5be35d6e505ad1d660621ca7@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '1e16a7ba5be35d6e505ad1d660621ca7@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '623c8d53297c330112f1d1ad003324e9@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.117:5060: CANCEL sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7fb94961;rportFrom: "8014525335" ;tag=as18ae777aTo: Call-ID: 623c8d53297c330112f1d1ad003324e9@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 asterisk-price*CLI> show channels --- Scheduling destruction of SIP dialog '623c8d53297c330112f1d1ad003324e9@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '49273abf515268df002b21d261901603@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.115:5060: CANCEL sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d74be68;rportFrom: "8014525335" ;tag=as16e85ad1To: Call-ID: 49273abf515268df002b21d261901603@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '49273abf515268df002b21d261901603@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '77058eb86663c0bf247c12b04e72f0ee@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.112:5060: CANCEL sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5b8956a6;rportFrom: "8014525335" ;tag=as1a70a7d8To: Call-ID: 77058eb86663c0bf247c12b04e72f0ee@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '77058eb86663c0bf247c12b04e72f0ee@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '2709ad5d203efe7931f2ef424c9ef70b@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.109:5060: CANCEL sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6b428b72;rportFrom: "8014525335" ;tag=as2ba931d9To: Call-ID: 2709ad5d203efe7931f2ef424c9ef70b@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '2709ad5d203efe7931f2ef424c9ef70b@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '18b5aae679f8e0f84bcb5d5f34c42ae1@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.108:5060: CANCEL sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d4e4674;rportFrom: "8014525335" ;tag=as70fd90b5To: Call-ID: 18b5aae679f8e0f84bcb5d5f34c42ae1@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 - asterisk-price*CLI> show channels -- Scheduling destruction of SIP dialog '18b5aae679f8e0f84bcb5d5f34c42ae1@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '5cd4159962e780b05bf08ca54b444b3c@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.106:5060: CANCEL sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18c3bed9;rportFrom: "8014525335" ;tag=as00245baaTo: Call-ID: 5cd4159962e780b05bf08ca54b444b3c@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '5cd4159962e780b05bf08ca54b444b3c@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '7bd1c72c39327a2b58461d58163bdf87@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.104:5060: CANCEL sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1586759f;rportFrom: "8014525335" ;tag=as5863900fTo: Call-ID: 7bd1c72c39327a2b58461d58163bdf87@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '7bd1c72c39327a2b58461d58163bdf87@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '581f9c973a03e74436aabeb31b8fd66a@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.103:5060: CANCEL sip:703@10.200.26.103:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3b90e921;rportFrom: "8014525335" ;tag=as63fdb7b7To: Call-ID: 581f9c973a03e74436aabeb31b8fd66a@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '581f9c973a03e74436aabeb31b8fd66a@66.111.122.20' in 32000 ms (Method: INVITE) Scheduling destruction of SIP dialog '3e25c493181c50652235f8fe03cbb490@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.101:5060: CANCEL sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3972f4e2;rportFrom: "8014525335" ;tag=as1568398dTo: Call-ID: 3e25c493181c50652235f8fe03cbb490@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> show channels Scheduling destruction of SIP dialog '3e25c493181c50652235f8fe03cbb490@66.111.122.20' in 32000 ms (Method: INVITE) --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3ae0bf56;rportFrom: "8014525335" ;tag=as167df14eTo: ;tag=48d92401cdbec04aCall-ID: 4203595b58bd72720c1f49221a1fc37a@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14 asterisk-price*CLI> show channels Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK412768d4;rportFrom: "8014525335" ;tag=as675a62adTo: ;tag=2fb9c820406f62a8Call-ID: 1d3d0edc6b0fb15a6e3523f87dff19a0@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK063bbde9;rportFrom: "8014525335" ;tag=as32280dc8To: ;tag=387b89739ce29e6bCall-ID: 6990abcc12416e630b4d02d164bab3e6@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK753eac34;rportFrom: "8014525335" ;tag=as11a97179To: ;tag=09c952b0c239d832Call-ID: 4d4eefa21b4907a67c34f1565d53fff0@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2c6d2ffd;rportFrom: "8014525335" ;tag=as43e7fb47To: ;tag=2db9ac6ab04a3d92Call-ID: 50d380237e76b23f5aa665f8650180f4@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.107:5060: ACK sip:707@10.200.26.107:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2c6d2ffd;rportFrom: "8014525335" ;tag=as43e7fb47To: ;tag=2db9ac6ab04a3d92Contact: Call-ID: 50d380237e76b23f5aa665f8650180f4@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK402bf19d;rportFrom: "8014525335" ;tag=as7aa604e9To: ;tag=63c9ad8085a108c9Call-ID: 0c98965a369fd60f60d237956c94af1f@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '50d380237e76b23f5aa665f8650180f4@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3ae0bf56;rportFrom: "8014525335" ;tag=as167df14eTo: ;tag=48d92401cdbec04aCall-ID: 4203595b58bd72720c1f49221a1fc37a@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.110:5060: ACK sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3ae0bf56;rportFrom: "8014525335" ;tag=as167df14eTo: ;tag=48d92401cdbec04aContact: Call-ID: 4203595b58bd72720c1f49221a1fc37a@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK063bbde9;rportFrom: "8014525335" ;tag=as32280dc8To: ;tag=387b89739ce29e6bCall-ID: 6990abcc12416e630b4d02d164bab3e6@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.131:5060: ACK sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK063bbde9;rportFrom: "8014525335" ;tag=as32280dc8To: ;tag=387b89739ce29e6bContact: Call-ID: 6990abcc12416e630b4d02d164bab3e6@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK412768d4;rportFrom: "8014525335" ;tag=as675a62adTo: ;tag=2fb9c820406f62a8Call-ID: 1d3d0edc6b0fb15a6e3523f87dff19a0@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.102:5060: ACK sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK412768d4;rportFrom: "8014525335" ;tag=as675a62adTo: ;tag=2fb9c820406f62a8Contact: Call-ID: 1d3d0edc6b0fb15a6e3523f87dff19a0@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '1d3d0edc6b0fb15a6e3523f87dff19a0@66.111.122.20' Method: INVITE Really destroying SIP dialog '4203595b58bd72720c1f49221a1fc37a@66.111.122.20' Method: INVITE Really destroying SIP dialog '6990abcc12416e630b4d02d164bab3e6@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK753eac34;rportFrom: "8014525335" ;tag=as11a97179To: ;tag=09c952b0c239d832Call-ID: 4d4eefa21b4907a67c34f1565d53fff0@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.130:5060: ACK sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK753eac34;rportFrom: "8014525335" ;tag=as11a97179To: ;tag=09c952b0c239d832Contact: Call-ID: 4d4eefa21b4907a67c34f1565d53fff0@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2aec2310;rportFrom: "8014525335" ;tag=as424157afTo: ;tag=e4f9fff162ea91bbCall-ID: 03fd70a67d97542542fc95033d2600c2@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK402bf19d;rportF asterisk-price*CLI> show channels rom: "8014525335" ;tag=as7aa604e9To: ;tag=63c9ad8085a108c9Call-ID: 0c98965a369fd60f60d237956c94af1f@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.129:5060: ACK sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK402bf19d;rportFrom: "8014525335" ;tag=as7aa604e9To: ;tag=63c9ad8085a108c9Contact: Call-ID: 0c98965a369fd60f60d237956c94af1f@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '4d4eefa21b4907a67c34f1565d53fff0@66.111.122.20' Method: INVITE Really destroying SIP dialog '0c98965a369fd60f60d237956c94af1f@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7bab5ceb;rportFrom: "8014525335" ;tag=as0f8d9c16To: ;tag=7237aba9d156f977Call-ID: 21c7d96f7a31e7332e8c819d2e2b6742@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> show channels<--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK443c6cba;rportFrom: "8014525335" ;tag=as67af68c8To: ;tag=80d92bf003f125bbCall-ID: 260d46712d1f57243552d01f0ca4830b@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK51266eb2;rportFrom: "8014525335" ;tag=as635f6292To: ;tag=83c5d44b69f22d5cCall-ID: 2577cddf74ecb70f1ca7bd985a9ea3a8@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2aec2310;rportFrom: "8014525335" ;tag=as424157afTo: ;tag=e4f9fff162ea91bbCall-ID: 03fd70a67d97542542fc95033d2600c2@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.126:5060: ACK sip:726@10.200.26.126:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2aec2310;rportFrom: "8014525335" ;tag=as424157afTo: ;tag=e4f9fff162ea91bbContact: Call-ID: 03fd70a67d97542542fc95033d2600c2@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK07ad3000;rportFrom: "8014525335" ;tag=as66eb57a0To: ;tag=63237d37e48736d1Call-ID: 1e16a7ba5be35d6e505ad1d660621ca7@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7bab5ceb;rportFrom: "8014525335" ;tag=as0f8d9c16To: ;tag=7237aba9d156f977Call-ID: 21c7d96f7a31e7332e8c819d2e2b6742@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.125:5060: ACK sip:725@10.200.26.125:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7bab5ceb;rportFrom: "8014525335" ;tag=as0f8d9c16To: ;tag=7237aba9d156f977Contact: Call-ID: 21c7d96f7a31e7332e8c819d2e2b6742@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '03fd70a67d97542542fc95033d2600c2@66.111.122.20' Method: INVITE Really destroying SIP dialog '21c7d96f7a31e7332e8c819d2e2b6742@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK443c6cba;rportFrom: "8014525335" ;tag=as67af68c8To: ;tag=80d92bf003f125bbCall-ID: 260d46712d1f57243552d01f0ca4830b@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.124:5060: ACK sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK443c6cba;rportFrom: "8014525335" ;tag=as67af68c8To: ;tag=80d92bf003f125bbContact: Call-ID: 260d46712d1f57243552d01f0ca4830b@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d74be68;rportFrom: "8014525335" ;tag=as16e85ad1To: ;tag=88f944e1ccb18f5aCall-ID: 49273abf515268df002b21d261901603@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK51266eb2;rportFrom: "8014525335" ;tag=as635f6292To: ;tag=83c5d44b69f22d5cCall-ID: 2577cddf74ecb70f1ca7bd985a9ea3a8@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.122:5060: ACK sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK51266eb2;rportFrom: "8014525335" ;tag=as635f6292To: ;tag=83c5d44b69f22d5cContact: Call-ID: 2577cddf74ecb70f1ca7bd985a9ea3a8@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 - asterisk-price*CLI> show channels -- Really destroying SIP dialog '260d46712d1f57243552d01f0ca4830b@66.111.122.20' Method: INVITE Really destroying SIP dialog '2577cddf74ecb70f1ca7bd985a9ea3a8@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5b8956a6;rportFrom: "8014525335" ;tag=as1a70a7d8To: ;tag=abd90701efbec24aCall-ID: 77058eb86663c0bf247c12b04e72f0ee@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6b428b72;rportFrom: "8014525335" ;tag=as2ba931d9To: ;tag=8aecef63ef17630dCall-ID: 2709ad5d203efe7931f2ef424c9ef70b@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d4e4674;rportFrom: "8014525335" ;tag=as70fd90b5To: ;tag=2ca4297b03bbd653Call-ID: 18b5aae679f8e0f84bcb5d5f34c42ae1@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK07ad3000;rportFrom: "8014525335" ;tag=as66eb57a0To: ;tag=63237d37e48736d1Call-ID: 1e16a7ba5be35d6e505ad1d660621ca7@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.118:5060: ACK sip:718@10.200.26.118:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK07ad3000;rportFrom: "8014525335" ;tag=as66eb57a0To: ;tag=63237d37e48736d1Contact: Call-ID: 1e16a7ba5be35d6e505ad1d660621ca7@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.127:5062 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6c339a68;rportFrom: "8014525335" ;tag=as2eb6bd6fTo: ;tag=4275227583281ceaCall-ID: 315fa8bf5c07b72838a2b4d261dca988@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.4.18Supported: replaces, timerContent-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d74be68;rportFrom: "8014525335" ;tag=as16e85ad1To: ;tag=88f944e1ccb18f5aCall-ID: 49273abf515268df002b21d261901603@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.115:5060: ACK sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d74be68;rportFrom: "8014525335" ;tag=as16e85ad1To: ;tag=88f944e1ccb18f5aContact: Call-ID: 49273abf515268df002b21d261901603@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '1e16a7ba5be35d6e505ad1d660621ca7@66.111.122.20' Method: INVITE Really destroying SIP dialog '49273abf515268df002b21d261901603@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5b8956a6;rportFrom: "8014525335" ;tag=as1a70a7d8To: ;tag=abd90701efbec24aCall-ID: 77058eb86663c0bf247c12b04e72f0ee@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.112:5060: ACK sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5b8956a6;rportFrom: "8014525335" ;tag=as1a70a7d8To: ;tag=abd90701efbec24aContact: Call-ID: 77058eb86663c0bf247c12b04e72f0ee@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6b428b72;rportFrom: "8014525335" ;tag=as2ba931d9To: ;tag=8aecef63ef17630dCall-ID: 2709ad5d203efe7931f2ef424c9ef70b@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.109:5060: ACK sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6b428b72;rportFrom: "8014525335" ;tag=as2ba931d9To: ;tag=8aecef63ef17630dContact: Call-ID: 2709ad5d203efe7931f2ef424c9ef70b@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18c3bed9;rportFrom: "8014525335" ;tag=as00245baaTo: ;tag=5bc5ee18620f519bCall-ID: 5cd4159962e780b05bf08ca54b444b3c@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '77058eb86663c0bf247c12b04e72f0ee@66.111.122.20' Method: INVITE Really destroying SIP dialog '2709ad5d203efe7931f2ef424c9ef70b@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d4e4674;rportFrom: "8014525335" ;tag=as70fd90b5To: ;tag=2ca4297b03bbd653Call-ID: 18b5aae679f8e0f84bcb5d5f34c42ae1@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.108:5060: ACK sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d4e4674;rportFrom: "8014525335" ;tag=as70fd90b5To: ;tag=2ca4297b03bbd653Contact: Call-ID: 18b5aae679f8e0f84bcb5d5f34c42ae1@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.127:5062 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6c339a68;rportFrom: "8014525335" ;tag=as2eb6bd6fTo: ;tag=4275227583281ceaCall-ID: 315fa8bf5c07b72838a2b4d261dca988@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.127:5062: ACK sip:727@10.200.26.127:5062;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6c339a68;rportFrom: "8014525335" ;tag=as2eb6bd6fTo: ;tag=4275227583281ceaContact: Call-ID: 315fa8bf5c07b72838a2b4d261dca988@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1586759f;rportFrom: "8014525335" ;tag=as5863900fTo: ;tag=e1d91c116dd75e6bCall-ID: 7bd1c72c39327a2b58461d58163bdf87@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '315fa8bf5c07b72838a2b4d261dca988@66.111.122.20' Method: INVITE Really destroying SIP dialog '18b5aae679f8e0f84bcb5d5f34c42ae1@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3b90e921;rportFrom: "8014525335" ;tag=as63fdb7b7To: ;tag=885596bb8f8c9b54Call-ID: 581f9c973a03e74436aabeb31b8fd66a@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7fb94961;rportFrom: "8014525335" ;tag=as18ae777aTo: ;tag=f713694cfa362d5aCall-ID: 623c8d53297c330112f1d1ad003324e9@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.4.18Supported: replaces, timerContent-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3972f4e2;rportFrom: "8014525335" ;tag=as1568398dTo: ;tag=0ad5d52de844d478Call-ID: 3e25c493181c50652235f8fe03cbb490@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18c3bed9;rportFrom: "8014525335" ;tag=as00245baaTo: ;tag=5bc5ee18620f519bCall-ID: 5cd4159962e780b05bf08ca54b444b3c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.106:5060: ACK sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18c3bed9;rportFrom: "8014525335" ;tag=as00245baaTo: ;tag=5bc5ee18620f519bContact: Call-ID: 5cd4159962e780b05bf08ca54b444b3c@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1586759f;rportFrom: "8014525335" ;tag=as5863900fTo: ;tag=e1d91c116dd75e6bCall-ID: 7bd1c72c39327a2b58461d58163bdf87@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.104:5060: ACK sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1586759f;rportFrom: "8014525335" ;tag=as5863900fTo: ;tag=e1d91c116dd75e6bContact: Call-ID: 7bd1c72c39327a2b58461d58163bdf87@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3b90e921;rportFrom: "8014525335" ;tag=as63fdb7b7To: ;tag=885596bb8f8c9b54Call-ID: 581f9c973a03e74436aabeb31b8fd66a@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.103:5060: ACK sip:703@10.200.26.103:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3b90e921;rportFrom: "8014525335" ;tag=as63fdb7b7To: ;tag=885596bb8f8c9b54Contact: Call-ID: 581f9c973a03e74436aabeb31b8fd66a@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '5cd4159962e780b05bf08ca54b444b3c@66.111.122.20' Method: INVITE Really destroying SIP dialog '7bd1c72c39327a2b58461d58163bdf87@66.111.122.20' Method: INVITE Really destroying SIP dialog '581f9c973a03e74436aabeb31b8fd66a@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3972f4e2;rportFrom: "8014525335" ;tag=as1568398dTo: ;tag=0ad5d52de844d478Call-ID: 3e25c493181c50652235f8fe03cbb490@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.101:5060: ACK sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3972f4e2;rportFrom: "8014525335" ;tag=as1568398dTo: ;tag=0ad5d52de844d478Contact: Call-ID: 3e25c493181c50652235f8fe03cbb490@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7fb94961;rportFrom: "8014525335" ;tag=as18ae777aTo: ;tag=f713694cfa362d5aCall-ID: 623c8d53297c330112f1d1ad003324e9@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.117:5060: ACK sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7fb94961;rportFrom: "8014525335" ;tag=as18ae777aTo: ;tag=f713694cfa362d5aContact: Call-ID: 623c8d53297c330112f1d1ad003324e9@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '623c8d53297c330112f1d1ad003324e9@66.111.122.20' Method: INVITE Really destroying SIP dialog '3e25c493181c50652235f8fe03cbb490@66.111.122.20' Method: INVITE asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) asterisk-price*CLI> SIP/705-0982c0e0 (None) Up Bridged Call(Zap/2-1) asterisk-price*CLI> Zap/2-1 900@from-zaptel:6 Up Dial(SIP/701&SIP/703&SIP/704&S asterisk-price*CLI> SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) asterisk-price*CLI> Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) asterisk-price*CLI> 4 active channels asterisk-price*CLI> 2 active calls asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> INVITE sip:8014525335@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK05efd31f5c8d65d1From: ;tag=5dc52ad73787efbbTo: "8014525335" ;tag=as614446beContact: Supported: replaces, timerCall-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 59650 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 280v=0o=705 8000 8001 IN IP4 10.200.26.105s=SIP Callc=IN IP4 10.200.26.105t=0 0m=audio 5004 RTP/AVP 0 8 4 18 2 3a=sendonlya=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:4 G723/8000a=rtpmap:18 G729/8000a=rtpmap:2 G726-32/8000a=rtpmap:3 GSM/8000a=ptime:20 <-------------> --- (13 headers 14 lines) --- Sending to 10.200.26.105 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 3 Peer audio RTP is at port 10.200.26.105:5004 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G723 for ID 4 Found description format G729 for ID 18 Found description format G726-32 for ID 2 Found description format GSM for ID 3 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.105:5004 Audio is at 10.200.26.202 port 10056 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK05efd31f5c8d65d1;received=10.200.26.105From: ;tag=5dc52ad73787efbbTo: "8014525335" ;tag=as614446beCall-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 59650 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Type: application/sdpContent-Length: 209v=0o=root 10024 10025 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10056 RTP/AVP 0 3a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=silenceSupp:off - - - -a=ptime:20a=recvonly <------------> -- Started music on hold, class 'default', on Zap/2-1 asterisk-price*CLI> [Jul 20 12:15:59] WARNING[11291]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> ACK sip:8014525335@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKb57a7a1abdf1880cFrom: ;tag=5dc52ad73787efbbTo: "8014525335" ;tag=as614446beContact: Call-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 59650 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> REFER sip:8014525335@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK2f079a0d4e8839f6From: ;tag=5dc52ad73787efbbTo: "8014525335" ;tag=as614446beContact: Supported: replacesRefer-To: Referred-By: Call-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 59651 REFERUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (14 headers 0 lines) --- Call 2665885949c42d88012168484f1e39f1@66.111.122.20 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 200@from-internal by 705@10.200.26.202 asterisk-price*CLI> == Spawn extension (from-zaptel, 900, 6) exited non-zero on 'Zap/2-1' -- Executing [h@from-zaptel:1] Hangup("Zap/2-1", "") in new stack == Spawn extension (from-zaptel, h, 1) exited non-zero on 'Zap/2-1' asterisk-price*CLI> -- Started music on hold, class 'default', on Zap/2-1 asterisk-price*CLI> [Jul 20 12:16:00] WARNING[10059]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 asterisk-price*CLI> == Parked Zap/2-1 on 201@parkedcalls. Will timeout back to extension [from-zaptel] 900, 6 in 75 seconds asterisk-price*CLI> -- Added extension '201' priority 1 to parkedcalls asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.127:5060: NOTIFY sip:727@10.200.26.127:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK508a3a91;rportFrom: ;tag=as0fac1be1To: "Conference Room" ;tag=1e94d3e74ab057e7Contact: Call-ID: cc74970ec691e7e1@10.200.26.127CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: active asterisk-price*CLI> Content-Length: 204 confirmed --- asterisk-price*CLI> Extension Changed 201 new state InUse for Notify User 727 Reliably Transmitting (NAT) to 10.200.26.131:5060: NOTIFY sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2598d4f2;rportFrom: ;tag=as44d1397bTo: "Lobby" ;tag=99f4161d5df43d88Contact: Call-ID: 62279987519ceb08@10.200.26.131CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 731 Reliably Transmitting (NAT) to 10.200.26.130:5060: NOTIFY sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3ee6bc33;rportFrom: ;tag=as73346657To: "Foyer" ;tag=bee43bdcf28e33c2Contact: Call-ID: 0c17b027f44a0b16@10.200.26.130CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 730 Reliably Transmitting (NAT) to 10.200.26.129:5060: NOTIFY sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK37e7da51;rportFrom: ;tag=as0b80a390To: "Will Jaimez" ;tag=18b4544c3b75f992Contact: Call-ID: 46071ba670da2a07@10.200.26.129CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 729 Reliably Transmitting (NAT) to 10.200.26.117:5060: NOTIFY sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5e078fc6;rportFrom: ;tag=as5ab9c51aTo: "Ken Williams" ;tag=85b710412fc42025Contact: Call-ID: d416754dbbee5bda@10.200.26.117CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- asterisk-price*CLI> Extension Changed 201 new state InUse for Notify User 717 Reliably Transmitting (NAT) to 10.200.26.126:5060: NOTIFY sip:726@10.200.26.126:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK49724fc4;rportFrom: ;tag=as3f56cc30To: "Bob Teny" ;tag=94c4908c7746b742Contact: Call-ID: 8e17d427fa4a6426@10.200.26.126CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 726 Reliably Transmitting (NAT) to 10.200.26.125:5060: NOTIFY sip:725@10.200.26.125:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d0c52a7;rportFrom: ;tag=as38f74437To: "Front Desk" ;tag=1bd437ccfd6ebe92Contact: Call-ID: 62473a6832fbae08@10.200.26.125CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 725 Reliably Transmitting (NAT) to 10.200.26.124:5060: NOTIFY sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6ce9774e;rportFrom: ;tag=as35039685To: "Bobby Houston" ;tag=79c4178cb056f172Contact: Call-ID: 8d27b397d8fca0f7@10.200.26.124CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 724 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.121:5060: NOTIFY sip:721@10.200.26.121:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK17742dd1;rportFrom: ;tag=as52c9f68fTo: "Danny D'Ambrosio" ;tag=91e4dffc190f5ae1Contact: Call-ID: 8c47d4781d5ccab6@10.200.26.121CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 721 Reliably Transmitting (NAT) to 10.200.26.122:5060: NOTIFY sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK68fd7710;rportFrom: ;tag=as3bd2c35dTo: "Jake Erramouspe" ;tag=d0e49efc170f18e1Contact: Call-ID: a94732785b5c49b6@10.200.26.122CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 722 Reliably Transmitting (NAT) to 10.200.26.120:5060: NOTIFY sip:720@10.200.26.120:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK782717a5;rportFrom: ;tag=as26caf817To: "Darek Martinez" ;tag=1ae4770db0f4d3b8Contact: Call-ID: 0067f749305becc6@10.200.26.120CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 720 Reliably Transmitting (NAT) to 10.200.26.119:5060: NOTIFY sip:719@10.200.26.119:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5f85ed30;rportFrom: ;tag=as4d16fad1To: "Randy Leffler" ;tag=9e94dba4740eb5d1Contact: Call-ID: 2f273697fcfc87f7@10.200.26.119CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2598d4f2;rportFrom: ;tag=as44d1397bTo: "Lobby" ;tag=99f4161d5df43d88Call-ID: 62279987519ceb08@10.200.26.131CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> Extension Changed 201 new state InUse for Notify User 719 Reliably Transmitting (NAT) to 10.200.26.118:5060: NOTIFY sip:718@10.200.26.118:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5dc7307b;rportFrom: ;tag=as501b143eTo: "Chris / Ed" ;tag=f9a4973cd07533c2Contact: Call-ID: 8647dd68f4fb0f08@10.200.26.118CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 718 Reliably Transmitting (NAT) to 10.200.26.116:5060: NOTIFY sip:716@10.200.26.116:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2c0e5869;rportFrom: ;tag=as71101529To: "Shop Kitchen" ;tag=3ea41b3c727533c2Contact: Call-ID: 484791787a5c06b6@10.200.26.116CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 716 Reliably Transmitting (NAT) to 10.200.26.111:5060: NOTIFY sip:711@10.200.26.111:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK163c438d;rportFrom: ;tag=as4dacd753To: "Center Shop" ;tag=cf7dedb2b53fa60fContact: Call-ID: 43ce35395af9fd6f@10.200.26.111CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 711 Reliably Transmitting (NAT) to 10.200.26.115:5060: NOTIFY sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5333e1f1;rportFrom: ;tag=as039135acTo: "Jerry Wright" ;tag=34b4537c3d36fe42Contact: Call-ID: af47d878926c2027@10.200.26.115CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- asterisk-price*CLI> Extension Changed 201 new state InUse for Notify User 715 Reliably Transmitting (NAT) to 10.200.26.113:5060: NOTIFY sip:713@10.200.26.113:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK02af0efd;rportFrom: ;tag=as05b1e1acTo: "Front Shop" ;tag=37a152d9516ad0eeContact: Call-ID: 49e06e58c4d76003@10.200.26.113CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 713 Reliably Transmitting (NAT) to 10.200.26.112:5060: NOTIFY sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0b668b0d;rportFrom: ;tag=as0b0c2072To: "Lorrie Blake" ;tag=d3947f943676b5a2Contact: Call-ID: 0d175327994ac226@10.200.26.112CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 712 Reliably Transmitting (NAT) to 10.200.26.110:5060: NOTIFY sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK52a2c836;rportFrom: ;tag=as3b8bd8b5To: "Shawn Norton" ;tag=d481dbf1f79a741fContact: Call-ID: 0ca0cf97a319ed9e@10.200.26.110CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 710 Reliably Transmitting (NAT) to 10.200.26.109:5060: NOTIFY sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7232718b;rportFrom: ;tag=as24dbb862To: "Alan Young" ;tag=597133f1324431aeContact: Call-ID: ace04268693825af@10.200.26.109CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 709 asterisk-price*CLI> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3ee6bc33;rportFrom: ;tag=as73346657To: "Foyer" ;tag=bee43bdcf28e33c2Call-ID: 0c17b027f44a0b16@10.200.26.130CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK37e7da51;rportFrom: ;tag=as0b80a390To: "Will Jaimez" ;tag=18b4544c3b75f992Call-ID: 46071ba670da2a07@10.200.26.129CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Reliably Transmitting (NAT) to 10.200.26.108:5060: NOTIFY sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7c58cf38;rportFrom: ;tag=as23529257To: "Jake Ori" ;tag=3b5457d35e7edd92Contact: Call-ID: ae07f2b6183b61b5@10.200.26.108CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- asterisk-price*CLI> Extension Changed 201 new state InUse for Notify User 708 Reliably Transmitting (NAT) to 10.200.26.107:5060: NOTIFY sip:707@10.200.26.107:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2b867ae4;rportFrom: ;tag=as55ec459bTo: "Tom Akers" ;tag=b864d414bbf47a88Contact: Call-ID: 05271c87d29c4c08@10.200.26.107CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 707 Reliably Transmitting (NAT) to 10.200.26.106:5060: NOTIFY sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK63e56f3c;rportFrom: ;tag=as12414dc8To: "Jenine Bentley" ;tag=19cef75d507ad439Contact: Call-ID: ba019788181b78b6@10.200.26.106CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 706 Reliably Transmitting (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0c18b3cc;rportFrom: ;tag=as30d10f77To: "Main Office Conf Room" ;tag=5b5475d3ba7e3992Contact: Call-ID: a2171917ddd9a517@10.200.26.105CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 705 Reliably Transmitting (NAT) to 10.200.26.104:5060: NOTIFY sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK68b28d01;rportFrom: ;tag=as6afb6310To: "Suzy Iorg" ;tag=06f2f98b43a41526Contact: Call-ID: 97b4dff5782a54a2@10.200.26.104CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- asterisk-price*CLI> Extension Changed 201 new state InUse for Notify User 704 Reliably Transmitting (NAT) to 10.200.26.103:5060: NOTIFY sip:703@10.200.26.103:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK59b41906;rportFrom: ;tag=as0f08e20bTo: "Shelli Marvidakis" ;tag=9984b794f0a6b218Contact: Call-ID: 0a5774e87eaceb46@10.200.26.103CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 703 Reliably Transmitting (NAT) to 10.200.26.102:5060: NOTIFY sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK68655b0a;rportFrom: ;tag=as414071ddTo: "Jason Worley" ;tag=5b84999412a69418Contact: Call-ID: 8a5755e830bc6ea6@10.200.26.102CSeq: 115 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 702 Reliably Transmitting (NAT) to 10.200.26.101:5060: NOTIFY sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3788c757;rportFrom: ;tag=as15c13c13To: "John Houston" ;tag=1d74fa5434d55668Contact: Call-ID: 88479078785cc3b6@10.200.26.101CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 204 confirmed --- Extension Changed 201 new state InUse for Notify User 701 <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK49724fc4;rportFrom: ;tag=as3f56cc30To: "Bob Teny" ;tag=94c4908c7746b742Call-ID: 8e17d427fa4a6426@10.200.26.126CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d0c52a7;rportFrom: ;tag=as38f74437To: "Front Desk" ;tag=1bd437ccfd6ebe92Call-ID: 62473a6832fbae08@10.200.26.125CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6ce9774e;rportFrom: ;tag=as35039685To: "Bobby Houston" ;tag=79c4178cb056f172Call-ID: 8d27b397d8fca0f7@10.200.26.124CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.121:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK17742dd1;rportFrom: ;tag=as52c9f68fTo: "Danny D'Ambrosio" ;tag=91e4dffc190f5ae1Call-ID: 8c47d4781d5ccab6@10.200.26.121CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK68fd7710;rportFrom: ;tag=as3bd2c35dTo: "Jake Erramouspe" ;tag=d0e49efc170f18e1Call-ID: a94732785b5c49b6@10.200.26.122CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.120:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK782717a5;rportFrom: ;tag=as26caf817To: "Darek Martinez" ;tag=1ae4770db0f4d3b8Call-ID: 0067f749305becc6@10.200.26.120CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.119:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5f85ed30;rportFrom: ;tag=as4d16fad1To: "Randy Leffler" ;tag=9e94dba4740eb5d1Call-ID: 2f273697fcfc87f7@10.200.26.119CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5dc7307b;rportFrom: ;tag=as501b143eTo: "Chris / Ed" ;tag=f9a4973cd07533c2Call-ID: 8647dd68f4fb0f08@10.200.26.118CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.116:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2c0e5869;rportFrom: ;tag=as71101529To: "Shop Kitchen" ;tag=3ea41b3c727533c2Call-ID: 484791787a5c06b6@10.200.26.116CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5333e1f1;rportFrom: ;tag=as039135acTo: "Jerry Wright" ;tag=34b4537c3d36fe42Call-ID: af47d878926c2027@10.200.26.115CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.111:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK163c438d;rportFrom: ;tag=as4dacd753To: "Center Shop" ;tag=cf7dedb2b53fa60fCall-ID: 43ce35395af9fd6f@10.200.26.111CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.2.23Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.113:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK02af0efd;rportFrom: ;tag=as05b1e1acTo: "Front Shop" ;tag=37a152d9516ad0eeCall-ID: 49e06e58c4d76003@10.200.26.113CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0b668b0d;rportFrom: ;tag=as0b0c2072To: "Lorrie Blake" ;tag=d3947f943676b5a2Call-ID: 0d175327994ac226@10.200.26.112CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK52a2c836;rportFrom: ;tag=as3b8bd8b5To: "Shawn Norton" ;tag=d481dbf1f79a741fCall-ID: 0ca0cf97a319ed9e@10.200.26.110CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7232718b;rportFrom: ;tag=as24dbb862To: "Alan Young" ;tag=597133f1324431aeCall-ID: ace04268693825af@10.200.26.109CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.127:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK508a3a91;rportFrom: ;tag=as0fac1be1To: "Conference Room" ;tag=1e94d3e74ab057e7Call-ID: cc74970ec691e7e1@10.200.26.127CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7c58cf38;rportFrom: ;tag=as23529257To: "Jake Ori" ;tag=3b5457d35e7edd92Call-ID: ae07f2b6183b61b5@10.200.26.108CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2b867ae4;rportFrom: ;tag=as55ec459bTo: "Tom Akers" ;tag=b864d414bbf47a88Call-ID: 05271c87d29c4c08@10.200.26.107CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK63e56f3c;rportFrom: ;tag=as12414dc8To: "Jenine Bentley" ;tag=19cef75d507ad439Call-ID: ba019788181b78b6@10.200.26.106CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0c18b3cc;rportFrom: ;tag=as30d10f77To: "Main Office Conf Room" ;tag=5b5475d3ba7e3992Call-ID: a2171917ddd9a517@10.200.26.105CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK68b28d01;rportFrom: ;tag=as6afb6310To: "Suzy Iorg" ;tag=06f2f98b43a41526Call-ID: 97b4dff5782a54a2@10.200.26.104CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK59b41906;rportFrom: ;tag=as0f08e20bTo: "Shelli Marvidakis" ;tag=9984b794f0a6b218Call-ID: 0a5774e87eaceb46@10.200.26.103CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK68655b0a;rportFrom: ;tag=as414071ddTo: "Jason Worley" ;tag=5b84999412a69418Call-ID: 8a5755e830bc6ea6@10.200.26.102CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14 asterisk-price*CLI> Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3788c757;rportFrom: ;tag=as15c13c13To: "John Houston" ;tag=1d74fa5434d55668Call-ID: 88479078785cc3b6@10.200.26.101CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5e078fc6;rportFrom: ;tag=as5ab9c51aTo: "Ken Williams" ;tag=85b710412fc42025Call-ID: d416754dbbee5bda@10.200.26.117CSeq: 115 NOTIFYUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 202 AcceptedVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK2f079a0d4e8839f6;received=10.200.26.105From: ;tag=5dc52ad73787efbbTo: "8014525335" ;tag=as614446beCall-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 59651 REFERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> set_destination: Parsing for address/port to send to asterisk-price*CLI> set_destination: set destination to 10.200.26.105, port 5060 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK024acab2;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbContact: Call-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=59651Subscription-state: terminated;reason=noresourceContent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 16SIP/2.0 200 OK --- asterisk-price*CLI> Scheduling destruction of SIP dialog '2665885949c42d88012168484f1e39f1@66.111.122.20' in 32000 ms (Method: REFER) asterisk-price*CLI> set_destination: Parsing for address/port to send to asterisk-price*CLI> set_destination: set destination to 10.200.26.105, port 5060 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.105:5060: BYE sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK008d539e;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbCall-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 104 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK024acab2;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbCall-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK008d539e;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbCall-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 104 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> SIP Response message for INCOMING dialog BYE arrived asterisk-price*CLI> Retransmitting #1 (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK024acab2;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbContact: Call-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=59651Subscription-state: terminated;reason=noresourceC asterisk-price*CLI> ontent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 16SIP/2.0 200 OK --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK024acab2;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbCall-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> Retransmitting #2 (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK024acab2;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbContact: Call-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=59651Subscription-state: terminated;reason=noresourceC asterisk-price*CLI> ontent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 16SIP/2.0 200 OK --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK024acab2;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbCall-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> INVITE sip:710@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK366eaac35e7fad2cFrom: "Main Office Conf Room" ;tag=c520c2975c39b4c7To: Contact: Supported: replaces, timerCall-ID: b2dedf24abac3c10@10.200.26.105CSeq: 22948 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 280v=0o=705 8000 8000 IN IP4 10.200.26.105s=SIP Callc=IN IP4 10.200.26.105t=0 0m=audio 5004 RTP/AVP 0 8 4 18 2 3a=sendrecva=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:4 G723/8000a=rtpmap:18 G729/8000a=rtpmap:2 G726-32/8000a=rtpmap:3 GSM/8000a=ptime:20 <-------------> --- (13 headers 14 lines) --- Sending to 10.200.26.105 : 5060 (NAT) Using INVITE request as basis request - b2dedf24abac3c10@10.200.26.105 <--- Reliably Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK366eaac35e7fad2c;received=10.200.26.105From: "Main Office Conf Room" ;tag=c520c2975c39b4c7To: ;tag=as74a874b9Call-ID: b2dedf24abac3c10@10.200.26.105CSeq: 22948 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesProxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="67b41c69"Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'b2dedf24abac3c10@10.200.26.105' in 32000 ms (Method: INVITE) Found user '705' asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> ACK sip:710@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK366eaac35e7fad2cFrom: "Main Office Conf Room" ;tag=c520c2975c39b4c7To: ;tag=as74a874b9Contact: Call-ID: b2dedf24abac3c10@10.200.26.105CSeq: 22948 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE asterisk-price*CLI> Content-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> INVITE sip:710@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKee1a0671e3a280e4From: "Main Office Conf Room" ;tag=c520c2975c39b4c7To: Contact: Supported: replaces, timerProxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:710@10.200.26.202", nonce="67b41c69", response="75d75b8a6370d5421cbe9540ba555e49"Call-ID: b2dedf24abac3c10@10.200.26.105CSeq: 22949 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 280v=0o=705 8000 8001 IN IP4 10.200.26.105s=SIP Callc=IN IP4 10.200.26.105t=0 0m=audio 5004 RTP/AVP 0 8 4 18 2 3a=sendrecva=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:4 G723/8000a=rtpmap:18 G729/8000a=rtpmap:2 G726-32/8000a=rtpmap:3 GSM/8000a asterisk-price*CLI> =ptime:20 <-------------> --- (14 headers 14 lines) --- Sending to 10.200.26.105 : 5060 (NAT) Using INVITE request as basis request - b2dedf24abac3c10@10.200.26.105 Found user '705' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 3 Peer audio RTP is at port 10.200.26.105:5004 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G723 for ID 4 asterisk-price*CLI> Found description format G729 for ID 18 Found description format G726-32 for ID 2 Found description format GSM for ID 3 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.105:5004 Looking for 710 in from-internal (domain 10.200.26.202) list_route: hop: <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKee1a0671e3a280e4;received=10.200.26.105From: "Main Office Conf Room" ;tag=c520c2975c39b4c7To: Call-ID: b2dedf24abac3c10@10.200.26.105CSeq: 22949 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> -- Executing [710@from-internal:1] SIPAddHeader("SIP/705-097a4ca0", "Call-Info: answer-after=0") in new stack asterisk-price*CLI> -- Executing [710@from-internal:2] Dial("SIP/705-097a4ca0", "SIP/710|20|Tt") in new stack asterisk-price*CLI> Audio is at 10.200.26.202 port 10448 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.110:5060: INVITE sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK20e0f2bc;rportFrom: "Conference Room" ;tag=as64e11dfeTo: Contact: Call-ID: 17ca1d5254333192772bf8c242f5ca92@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:03 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10448 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 710 asterisk-price*CLI> <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK20e0f2bc;rportFrom: "Conference Room" ;tag=as64e11dfeTo: Call-ID: 17ca1d5254333192772bf8c242f5ca92@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK20e0f2bc;rportFrom: "Conference Room" ;tag=as64e11dfeTo: ;tag=5d190352a8cfeaceCall-ID: 17ca1d5254333192772bf8c242f5ca92@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> -- SIP/710-09835ed0 is ringing asterisk-price*CLI> <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKee1a0671e3a280e4;received=10.200.26.105From: "Main Office Conf Room" ;tag=c520c2975c39b4c7To: ;tag=as75ce1708Call-ID: b2dedf24abac3c10@10.200.26.105CSeq: 22949 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK20e0f2bc;rportFrom: "Conference Room" ;tag=as64e11dfeTo: ;tag=5d190352a8cfeaceCall-ID: 17ca1d5254333192772bf8c242f5ca92@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=710 8000 8000 IN IP4 10.200.26.110s=SIP Callc=IN IP4 10.200.26.110t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.110:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.110:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.110, port 5060 Transmitting (NAT) to 10.200.26.110:5060: ACK sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK19a7021a;rportFrom: "Conference Room" ;tag=as64e11dfeTo: ;tag=5d190352a8cfeaceContact: Call-ID: 17ca1d5254333192772bf8c242f5ca92@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/710-09835ed0 answered SIP/705-097a4ca0 Audio is at 10.200.26.202 port 10678 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP <--- Reliably Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKee1a0671e3a280e4;received=10.200.26.105From: "Main Office Conf Room" ;tag=c520c2975c39b4c7To: ;tag=as75ce1708Call-ID: b2dedf24abac3c10@10.200.26.105CSeq: 22949 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Type: application/sdpContent-Length: 209v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10678 RTP/AVP 0 3a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=silenceSupp:off - - - -a=ptime:20a=sendrecv <------------> asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> ACK sip:710@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK30be97b03d109987From: "Main Office Conf Room" ;tag=c520c2975c39b4c7To: ;tag=as75ce1708Contact: Proxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:710@10.200.26.202", nonce="67b41c69", response="b9fba7b0678865ccebe46414804ab07f"Call-ID: b2dedf24abac3c10@10.200.26.105CSeq: 22949 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (12 headers 0 lines) --- asterisk-price*CLI> Retransmitting #3 (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK024acab2;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbContact: Call-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=59651Subscription-state: terminated;reason=noresourceC asterisk-price*CLI> ontent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 16SIP/2.0 200 OK --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK024acab2;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbCall-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) asterisk-price*CLI> SIP/710-09835ed0 (None) Up Bridged Call(SIP/705-097a4ca0) asterisk-price*CLI> SIP/705-097a4ca0 710@from-internal:2 Up Dial(SIP/710|20|Tt) asterisk-price*CLI> Zap/2-1 900@from-zaptel:6 Up Parked Call() asterisk-price*CLI> SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) asterisk-price*CLI> Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) asterisk-price*CLI> 5 active channels asterisk-price*CLI> 2 active calls asterisk-price*CLI> Retransmitting #4 (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK024acab2;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbContact: Call-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=59651Subscription-state: terminated;reason=noresourceC asterisk-price*CLI> ontent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 16SIP/2.0 200 OK --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK024acab2;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbCall-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> BYE sip:710@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK2156652d7dd9dd71From: "Main Office Conf Room" ;tag=c520c2975c39b4c7To: ;tag=as75ce1708Proxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:710@10.200.26.202", nonce="67b41c69", response="fd41b8b0b27c9a234c7f77e27166a80b"Call-ID: b2dedf24abac3c10@10.200.26.105CSeq: 22950 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.200.26.105 : 5060 (NAT) <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK2156652d7dd9dd71;received=10.200.26.105From: "Main Office Conf Room" ;tag=c520c2975c39b4c7To: ;tag=as75ce1708Call-ID: b2dedf24abac3c10@10.200.26.105CSeq: 22950 BYEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> Scheduling destruction of SIP dialog '17ca1d5254333192772bf8c242f5ca92@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.110, port 5060 Reliably Transmitting (NAT) to 10.200.26.110:5060: BYE sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2e27e3ba;rportFrom: "Conference Room" ;tag=as64e11dfeTo: ;tag=5d190352a8cfeaceCall-ID: 17ca1d5254333192772bf8c242f5ca92@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- == Spawn extension (from-internal, 710, 2) exited non-zero on 'SIP/705-097a4ca0' -- Executing [h@from-internal:1] Hangup("SIP/705-097a4ca0", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/705-097a4ca0' asterisk-price*CLI> Really destroying SIP dialog 'b2dedf24abac3c10@10.200.26.105' Method: BYE asterisk-price*CLI> <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2e27e3ba;rportFrom: "Conference Room" ;tag=as64e11dfeTo: ;tag=5d190352a8cfeaceCall-ID: 17ca1d5254333192772bf8c242f5ca92@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> Really destroying SIP dialog '17ca1d5254333192772bf8c242f5ca92@66.111.122.20' Method: INVITE asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> INVITE sip:760@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKf452bfe324f01c81From: "Main Office Conf Room" ;tag=4aed5c1b08e0dbdfTo: Contact: Supported: replaces, timerCall-ID: c05b3f48efc16884@10.200.26.105CSeq: 20223 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 280v=0o=705 8000 8000 IN IP4 10.200.26.105s=SIP Callc=IN IP4 10.200.26.105t=0 0m=audio 5004 RTP/AVP 0 8 4 18 2 3a=sendrecva=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:4 G723/8000a=rtpmap:18 G729/8000a=rtpmap:2 G726-32/8000a=rtpmap:3 GSM/8000a=ptime:20 <-------------> --- (13 headers 14 lines) --- Sending to 10.200.26.105 : 5060 (NAT) Using INVITE request as basis request - c05b3f48efc16884@10.200.26.105 <--- Reliably Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKf452bfe324f01c81;received=10.200.26.105From: "Main Office Conf Room" ;tag=4aed5c1b08e0dbdfTo: ;tag=as53fa9306Call-ID: c05b3f48efc16884@10.200.26.105CSeq: 20223 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesProxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0c3f45ae"Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'c05b3f48efc16884@10.200.26.105' in 32000 ms (Method: INVITE) Found user '705' asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> ACK sip:760@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKf452bfe324f01c81From: "Main Office Conf Room" ;tag=4aed5c1b08e0dbdfTo: ;tag=as53fa9306Contact: Call-ID: c05b3f48efc16884@10.200.26.105CSeq: 20223 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE asterisk-price*CLI> Content-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> INVITE sip:760@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKfd94ec7fcf16ddffFrom: "Main Office Conf Room" ;tag=4aed5c1b08e0dbdfTo: Contact: Supported: replaces, timerProxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:760@10.200.26.202", nonce="0c3f45ae", response="c1a61c7c97e43f100c4778b5f30986c8"Call-ID: c05b3f48efc16884@10.200.26.105CSeq: 20224 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 280v=0o=705 8000 8001 IN IP4 10.200.26.105s=SIP Callc=IN IP4 10.200.26.105t=0 0m=audio 5004 RTP/AVP 0 8 4 18 2 3a=sendrecva=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:4 G723/8000a=rtpmap:18 G729/8000a=rtpmap:2 G726-32/8000a=rtpmap:3 GSM/8000a asterisk-price*CLI> =ptime:20 <-------------> --- (14 headers 14 lines) --- Sending to 10.200.26.105 : 5060 (NAT) Using INVITE request as basis request - c05b3f48efc16884@10.200.26.105 Found user '705' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 3 Peer audio RTP is at port 10.200.26.105:5004 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G723 for ID 4 asterisk-price*CLI> Found description format G729 for ID 18 Found description format G726-32 for ID 2 Found description format GSM for ID 3 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.105:5004 Looking for 760 in from-internal (domain 10.200.26.202) list_route: hop: <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKfd94ec7fcf16ddff;received=10.200.26.105From: "Main Office Conf Room" ;tag=4aed5c1b08e0dbdfTo: Call-ID: c05b3f48efc16884@10.200.26.105CSeq: 20224 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> -- Executing [760@from-internal:1] GotoIf("SIP/705-097a4ca0", "0?parkedcalls|200|1") in new stack asterisk-price*CLI> -- Executing [760@from-internal:2] SIPAddHeader("SIP/705-097a4ca0", "Call-Info: answer-after=0") in new stack asterisk-price*CLI> -- Executing [760@from-internal:3] Set("SIP/705-097a4ca0", "TIMEOUT(absolute)=10") in new stack asterisk-price*CLI> -- Channel will hangup at 2007-07-20 18:16:19 UTC. asterisk-price*CLI> -- Executing [760@from-internal:4] Page("SIP/705-097a4ca0", "SIP/701&SIP/702&SIP/703&SIP/704&SIP/705&SIP/706&SIP/707&SIP/708&SIP/709&SIP/710&SIP/711&SIP/712&SIP/713&SIP/715&SIP/716&SIP/717&SIP/718&SIP/719&SIP/720&SIP/721&SIP/722&SIP/724&SIP/725&SIP/726&SIP/727&SIP/728&SIP/729&SIP/730&SIP/731&SIP/739|10") in new stack asterisk-price*CLI> Audio is at 10.200.26.202 port 10568 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.101:5060: INVITE sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1e145269;rportFrom: "Conference Room" ;tag=as5ac1abdaTo: Contact: Call-ID: 68e28dd61a2046a0577d488b2efa0df1@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10568 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 701 asterisk-price*CLI> Audio is at 10.200.26.202 port 10644 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1e145269;rportFrom: "Conference Room" ;tag=as5ac1abdaTo: Call-ID: 68e28dd61a2046a0577d488b2efa0df1@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1e145269;rportFrom: "Conference Room" ;tag=as5ac1abdaTo: ;tag=37f37336cb895347Call-ID: 68e28dd61a2046a0577d488b2efa0df1@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.102:5060: INVITE sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1e911855;rportFrom: "Conference Room" ;tag=as72155756To: Contact: Call-ID: 1596367e0f152ef076c94b595c7c8163@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10644 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 702 asterisk-price*CLI> Audio is at 10.200.26.202 port 10470 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1e911855;rportFrom: "Conference Room" ;tag=as72155756To: Call-ID: 1596367e0f152ef076c94b595c7c8163@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1e911855;rportFrom: "Conference Room" ;tag=as72155756To: ;tag=5309a8024e0fee0bCall-ID: 1596367e0f152ef076c94b595c7c8163@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.103:5060: INVITE sip:703@10.200.26.103:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6576daea;rportFrom: "Conference Room" ;tag=as3490df47To: Contact: Call-ID: 734d61614b6f97c77e38342616025178@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10470 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 703 asterisk-price*CLI> Audio is at 10.200.26.202 port 10870 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6576daea;rportFrom: "Conference Room" ;tag=as3490df47To: Call-ID: 734d61614b6f97c77e38342616025178@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6576daea;rportFrom: "Conference Room" ;tag=as3490df47To: ;tag=b609ab02e01f617bCall-ID: 734d61614b6f97c77e38342616025178@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.104:5060: INVITE sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK09b72e92;rportFrom: "Conference Room" ;tag=as2457465eTo: Contact: Call-ID: 6aae31bb3ea8c4c17178015a36fb4fdf@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10870 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 704 asterisk-price*CLI> -- SIP/701-09835ed0 is ringing asterisk-price*CLI> -- SIP/702-09864628 is ringing asterisk-price*CLI> -- SIP/703-0983adc8 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK09b72e92;rportFrom: "Conference Room" ;tag=as2457465eTo: Call-ID: 6aae31bb3ea8c4c17178015a36fb4fdf@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK09b72e92;rportFrom: "Conference Room" ;tag=as2457465eTo: ;tag=069f8bc9ccfb2dd5Call-ID: 6aae31bb3ea8c4c17178015a36fb4fdf@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> -- SIP/704-09939d20 is ringing asterisk-price*CLI> Audio is at 10.200.26.202 port 10666 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.106:5060: INVITE sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK168a6c47;rportFrom: "Conference Room" ;tag=as787c678cTo: Contact: Call-ID: 60e8999a1938524d2ec9c00e6001926f@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10666 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 706 asterisk-price*CLI> Audio is at 10.200.26.202 port 10406 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK168a6c47;rportFrom: "Conference Room" ;tag=as787c678cTo: Call-ID: 60e8999a1938524d2ec9c00e6001926f@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK168a6c47;rportFrom: "Conference Room" ;tag=as787c678cTo: ;tag=208954d6317f9c69Call-ID: 60e8999a1938524d2ec9c00e6001926f@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.107:5060: INVITE sip:707@10.200.26.107:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5d5f210d;rportFrom: "Conference Room" ;tag=as6de7994fTo: Contact: Call-ID: 2c60d1190546aed93106a1fe5f8e2e4f@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10406 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 707 asterisk-price*CLI> Audio is at 10.200.26.202 port 10622 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5d5f210d;rportFrom: "Conference Room" ;tag=as6de7994fTo: Call-ID: 2c60d1190546aed93106a1fe5f8e2e4f@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5d5f210d;rportFrom: "Conference Room" ;tag=as6de7994fTo: ;tag=b65273658d3725beCall-ID: 2c60d1190546aed93106a1fe5f8e2e4f@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.108:5060: INVITE sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK46907d4f;rportFrom: "Conference Room" ;tag=as1df61756To: Contact: Call-ID: 31f848b95945bdc1657450c72b0b5030@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10622 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 708 asterisk-price*CLI> Audio is at 10.200.26.202 port 10044 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK46907d4f;rportFrom: "Conference Room" ;tag=as1df61756To: Call-ID: 31f848b95945bdc1657450c72b0b5030@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK46907d4f;rportFrom: "Conference Room" ;tag=as1df61756To: ;tag=9cf84f1c622fe1cbCall-ID: 31f848b95945bdc1657450c72b0b5030@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.109:5060: INVITE sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2eb626b9;rportFrom: "Conference Room" ;tag=as7acb1113To: Contact: Call-ID: 3e16a7734eeccef320ca05ff419906f6@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10044 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 709 asterisk-price*CLI> Audio is at 10.200.26.202 port 10228 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> -- SIP/706-0991d8c0 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2eb626b9;rportFrom: "Conference Room" ;tag=as7acb1113To: Call-ID: 3e16a7734eeccef320ca05ff419906f6@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> -- SIP/707-09842c20 is ringing asterisk-price*CLI> -- SIP/708-097b1d58 is ringing asterisk-price*CLI> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2eb626b9;rportFrom: "Conference Room" ;tag=as7acb1113To: ;tag=8d987e03bf1e6c3aCall-ID: 3e16a7734eeccef320ca05ff419906f6@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.110:5060: INVITE sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK37663b27;rportFrom: "Conference Room" ;tag=as15bbf524To: Contact: Call-ID: 1a89fb794e3e024057514cf718c53356@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10228 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 710 asterisk-price*CLI> Audio is at 10.200.26.202 port 10526 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK37663b27;rportFrom: "Conference Room" ;tag=as15bbf524To: Call-ID: 1a89fb794e3e024057514cf718c53356@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.111:5060: INVITE sip:711@10.200.26.111:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5c3a410d;rportFrom: "Conference Room" ;tag=as196ea359To: Contact: Call-ID: 2831eb4d191c591361840f6458e5d3a6@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10526 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK37663b27;rportFrom: "Conference Room" ;tag=as15bbf524To: ;tag=c04a265e3f704f1aCall-ID: 1a89fb794e3e024057514cf718c53356@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> -- Called 711 asterisk-price*CLI> Audio is at 10.200.26.202 port 10384 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.112:5060: INVITE sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1ec4e490;rportFrom: "Conference Room" ;tag=as2b565667To: Contact: Call-ID: 56bea4841960214952cae96768b232e9@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10384 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> <--- SIP read from 10.200.26.111:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5c3a410d;rportFrom: "Conference Room" ;tag=as196ea359To: Call-ID: 2831eb4d191c591361840f6458e5d3a6@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.2.23Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> -- Called 712 asterisk-price*CLI> <--- SIP read from 10.200.26.111:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5c3a410d;rportFrom: "Conference Room" ;tag=as196ea359To: ;tag=a50d4bf7a237bb99Call-ID: 2831eb4d191c591361840f6458e5d3a6@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.2.23Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1ec4e490;rportFrom: "Conference Room" ;tag=as2b565667To: Call-ID: 56bea4841960214952cae96768b232e9@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1ec4e490;rportFrom: "Conference Room" ;tag=as2b565667To: ;tag=3309e7020d0fcd0bCall-ID: 56bea4841960214952cae96768b232e9@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Audio is at 10.200.26.202 port 10220 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.113:5060: INVITE sip:713@10.200.26.113:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4b3a09dd;rportFrom: "Conference Room" ;tag=as071911baTo: Contact: Call-ID: 3d42cb2e76cad56a2e8b13cd7c4da3a8@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10220 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 713 asterisk-price*CLI> -- SIP/709-098625b8 is ringing asterisk-price*CLI> -- SIP/710-09825a50 is ringing asterisk-price*CLI> -- SIP/711-097bf390 is ringing asterisk-price*CLI> -- SIP/712-0984a588 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.113:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4b3a09dd;rportFrom: "Conference Room" ;tag=as071911baTo: Call-ID: 3d42cb2e76cad56a2e8b13cd7c4da3a8@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.113:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4b3a09dd;rportFrom: "Conference Room" ;tag=as071911baTo: ;tag=e175f56af1235b7bCall-ID: 3d42cb2e76cad56a2e8b13cd7c4da3a8@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Audio is at 10.200.26.202 port 10638 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.115:5060: INVITE sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5376ce11;rportFrom: "Conference Room" ;tag=as0b501a50To: Contact: Call-ID: 7768e384367bcf246706bc1e4e8cf8f7@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10638 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 715 asterisk-price*CLI> Audio is at 10.200.26.202 port 10476 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5376ce11;rportFrom: "Conference Room" ;tag=as0b501a50To: Call-ID: 7768e384367bcf246706bc1e4e8cf8f7@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5376ce11;rportFrom: "Conference Room" ;tag=as0b501a50To: ;tag=f419e9422f9f408eCall-ID: 7768e384367bcf246706bc1e4e8cf8f7@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.116:5060: INVITE sip:716@10.200.26.116:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7a8f3fd9;rportFrom: "Conference Room" ;tag=as224a4d42To: Contact: Call-ID: 2095ee5434257d94767ecce318d9aea1@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:09 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10476 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 716 asterisk-price*CLI> Audio is at 10.200.26.202 port 10484 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.116:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7a8f3fd9;rportFrom: "Conference Room" ;tag=as224a4d42To: Call-ID: 2095ee5434257d94767ecce318d9aea1@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.116:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7a8f3fd9;rportFrom: "Conference Room" ;tag=as224a4d42To: ;tag=a289a10e1d865b5eCall-ID: 2095ee5434257d94767ecce318d9aea1@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.117:5060: INVITE sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7a0a9167;rportFrom: "Conference Room" ;tag=as53bc76acTo: Contact: Call-ID: 315c17210e59ed7b1c5e65d209f10c79@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10484 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 717 asterisk-price*CLI> -- SIP/713-09854a00 is ringing asterisk-price*CLI> -- SIP/715-09840390 is ringing asterisk-price*CLI> -- SIP/716-09811930 is ringing asterisk-price*CLI> Audio is at 10.200.26.202 port 10420 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.118:5060: INVITE sip:718@10.200.26.118:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK252302e5;rportFrom: "Conference Room" ;tag=as20ccb9eaTo: Contact: Call-ID: 388a13b34ae12be842ac94e465834e11@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10420 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 718 asterisk-price*CLI> Audio is at 10.200.26.202 port 10492 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK252302e5;rportFrom: "Conference Room" ;tag=as20ccb9eaTo: Call-ID: 388a13b34ae12be842ac94e465834e11@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK252302e5;rportFrom: "Conference Room" ;tag=as20ccb9eaTo: ;tag=5ef8622c264fa58cCall-ID: 388a13b34ae12be842ac94e465834e11@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7a0a9167;rportFrom: "Conference Room" ;tag=as53bc76acTo: Call-ID: 315c17210e59ed7b1c5e65d209f10c79@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.119:5060: INVITE sip:719@10.200.26.119:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK655a03d0;rportFrom: "Conference Room" ;tag=as7c43023cTo: Contact: Call-ID: 31c5322a4a18c2e1033659987decf745@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10492 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 719 asterisk-price*CLI> Audio is at 10.200.26.202 port 10604 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7a0a9167;rportFrom: "Conference Room" ;tag=as53bc76acTo: ;tag=39e792d76ec98df4Call-ID: 315c17210e59ed7b1c5e65d209f10c79@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.200.26.119:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK655a03d0;rportFrom: "Conference Room" ;tag=as7c43023cTo: Call-ID: 31c5322a4a18c2e1033659987decf745@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.119:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK655a03d0;rportFrom: "Conference Room" ;tag=as7c43023cTo: ;tag=0f835da74832f462Call-ID: 31c5322a4a18c2e1033659987decf745@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> -- SIP/717-098691d0 is ringing asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.120:5060: INVITE sip:720@10.200.26.120:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4236697d;rportFrom: "Conference Room" ;tag=as20842e79To: Contact: Call-ID: 66606fd57172bad153d652c34eff1e0c@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10604 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 720 asterisk-price*CLI> Audio is at 10.200.26.202 port 10976 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.120:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4236697d;rportFrom: "Conference Room" ;tag=as20842e79To: Call-ID: 66606fd57172bad153d652c34eff1e0c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> asterisk-price*CLI> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.120:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4236697d;rportFrom: "Conference Room" ;tag=as20842e79To: ;tag=6b66ac8fe4ef7091Call-ID: 66606fd57172bad153d652c34eff1e0c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.121:5060: INVITE sip:721@10.200.26.121:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK32d1c0d5;rportFrom: "Conference Room" ;tag=as1ec1d811To: Contact: Call-ID: 73a9d25308b48d607f4357d04eb7f5e7@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10976 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 721 asterisk-price*CLI> Audio is at 10.200.26.202 port 10386 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.121:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK32d1c0d5;rportFrom: "Conference Room" ;tag=as1ec1d811To: Call-ID: 73a9d25308b48d607f4357d04eb7f5e7@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> -- SIP/718-097cf358 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.121:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK32d1c0d5;rportFrom: "Conference Room" ;tag=as1ec1d811To: ;tag=63a9650eb3b6b34cCall-ID: 73a9d25308b48d607f4357d04eb7f5e7@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> -- SIP/719-09816518 is ringing asterisk-price*CLI> -- SIP/720-097b8718 is ringing asterisk-price*CLI> -- SIP/721-097b32d8 is ringing asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.122:5060: INVITE sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK78278d2d;rportFrom: "Conference Room" ;tag=as52466f3fTo: Contact: Call-ID: 469c859f578805563b3e6e7507a70a7e@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10386 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 722 asterisk-price*CLI> Audio is at 10.200.26.202 port 10454 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK78278d2d;rportFrom: "Conference Room" ;tag=as52466f3fTo: Call-ID: 469c859f578805563b3e6e7507a70a7e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK78278d2d;rportFrom: "Conference Room" ;tag=as52466f3fTo: ;tag=08ad89fdfe20bbc5Call-ID: 469c859f578805563b3e6e7507a70a7e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.124:5060: INVITE sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK21bc8f92;rportFrom: "Conference Room" ;tag=as785f5377To: Contact: Call-ID: 5f9551110fe5f3e66ec1aa8849eb04d2@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10454 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 724 asterisk-price*CLI> Audio is at 10.200.26.202 port 10008 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK21bc8f92;rportFrom: "Conference Room" ;tag=as785f5377To: Call-ID: 5f9551110fe5f3e66ec1aa8849eb04d2@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK21bc8f92;rportFrom: "Conference Room" ;tag=as785f5377To: ;tag=7758fd3287ae981dCall-ID: 5f9551110fe5f3e66ec1aa8849eb04d2@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.125:5060: INVITE sip:725@10.200.26.125:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK46d4df6d;rportFrom: "Conference Room" ;tag=as1ec6dc28To: Contact: Call-ID: 46017eb0146a5fd149d9fd114dd69347@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10008 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1e145269;rportFrom: "Conference Room" ;tag=as5ac1abdaTo: ;tag=37f37336cb895347Call-ID: 68e28dd61a2046a0577d488b2efa0df1@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=701 8000 8000 IN IP4 10.200.26.101s=SIP Callc=IN IP4 10.200.26.101t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> asterisk-price*CLI> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.101:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.101:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.101, port 5060 Transmitting (NAT) to 10.200.26.101:5060: ACK sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK53206161;rportFrom: "Conference Room" ;tag=as5ac1abdaTo: ;tag=37f37336cb895347Contact: Call-ID: 68e28dd61a2046a0577d488b2efa0df1@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/701-09835ed0 answered -- Created MeetMe conference 1023 for conference '1196561824d' asterisk-price*CLI> -- Called 725 asterisk-price*CLI> <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1e911855;rportFrom: "Conference Room" ;tag=as72155756To: ;tag=5309a8024e0fee0bCall-ID: 1596367e0f152ef076c94b595c7c8163@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=702 8000 8000 IN IP4 10.200.26.102s=SIP Callc=IN IP4 10.200.26.102t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.102:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.102:5004 asterisk-price*CLI> list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.102, port 5060 Transmitting (NAT) to 10.200.26.102:5060: ACK sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK41fce52e;rportFrom: "Conference Room" ;tag=as72155756To: ;tag=5309a8024e0fee0bContact: Call-ID: 1596367e0f152ef076c94b595c7c8163@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK46d4df6d;rportFrom: "Conference Room" ;tag=as1ec6dc28To: Call-ID: 46017eb0146a5fd149d9fd114dd69347@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> -- SIP/702-09864628 answered asterisk-price*CLI> <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK46d4df6d;rportFrom: "Conference Room" ;tag=as1ec6dc28To: ;tag=31a7260fa05e0860Call-ID: 46017eb0146a5fd149d9fd114dd69347@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Audio is at 10.200.26.202 port 10258 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.126:5060: INVITE sip:726@10.200.26.126:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK768c18eb;rportFrom: "Conference Room" ;tag=as0069906eTo: Contact: Call-ID: 7c36f46e6fe0c6c17895542177d8abf2@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10258 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 726 asterisk-price*CLI> -- SIP/722-0993bc70 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6576daea;rportFrom: "Conference Room" ;tag=as3490df47To: ;tag=b609ab02e01f617bCall-ID: 734d61614b6f97c77e38342616025178@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=703 8000 8000 IN IP4 10.200.26.103s=SIP Callc=IN IP4 10.200.26.103t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> asterisk-price*CLI> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.103:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.103:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.103, port 5060 asterisk-price*CLI> Transmitting (NAT) to 10.200.26.103:5060: ACK sip:703@10.200.26.103:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK63661547;rportFrom: "Conference Room" ;tag=as3490df47To: ;tag=b609ab02e01f617bContact: Call-ID: 734d61614b6f97c77e38342616025178@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK768c18eb;rportFrom: "Conference Room" ;tag=as0069906eTo: Call-ID: 7c36f46e6fe0c6c17895542177d8abf2@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> -- SIP/703-0983adc8 answered asterisk-price*CLI> <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK768c18eb;rportFrom: "Conference Room" ;tag=as0069906eTo: ;tag=99587faa32547b7bCall-ID: 7c36f46e6fe0c6c17895542177d8abf2@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> -- SIP/724-0993fbd8 is ringing asterisk-price*CLI> -- SIP/725-09941158 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK09b72e92;rportFrom: "Conference Room" ;tag=as2457465eTo: ;tag=069f8bc9ccfb2dd5Call-ID: 6aae31bb3ea8c4c17178015a36fb4fdf@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=704 8000 8000 IN IP4 10.200.26.104s=SIP Callc=IN IP4 10.200.26.104t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.104:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.104:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.104, port 5060 Transmitting (NAT) to 10.200.26.104:5060: ACK sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6e16e8db;rportFrom: "Conference Room" ;tag=as2457465eTo: ;tag=069f8bc9ccfb2dd5Contact: Call-ID: 6aae31bb3ea8c4c17178015a36fb4fdf@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/704-09939d20 answered asterisk-price*CLI> Audio is at 10.200.26.202 port 10396 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.127:5062: INVITE sip:727@10.200.26.127:5062;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK569753ac;rportFrom: "Conference Room" ;tag=as58e7b3f8To: Contact: Call-ID: 14327a4224186e2812da18b15cc51d83@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10396 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 727 asterisk-price*CLI> <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK168a6c47;rportFrom: "Conference Room" ;tag=as787c678cTo: ;tag=208954d6317f9c69Call-ID: 60e8999a1938524d2ec9c00e6001926f@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=706 8000 8000 IN IP4 10.200.26.106s=SIP Callc=IN IP4 10.200.26.106t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.106:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.106:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.106, port 5060 Transmitting (NAT) to 10.200.26.106:5060: ACK sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5bf7e06d;rportFrom: "Conference Room" ;tag=as787c678cTo: ;tag=208954d6317f9c69Contact: Call-ID: 60e8999a1938524d2ec9c00e6001926f@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/706-0991d8c0 answered asterisk-price*CLI> Audio is at 10.200.26.202 port 10532 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5d5f210d;rportFrom: "Conference Room" ;tag=as6de7994fTo: ;tag=b65273658d3725beCall-ID: 2c60d1190546aed93106a1fe5f8e2e4f@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=707 8000 8000 IN IP4 10.200.26.107s=SIP Callc=IN IP4 10.200.26.107t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.107:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) asterisk-price*CLI> Peer audio RTP is at port 10.200.26.107:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.107, port 5060 Transmitting (NAT) to 10.200.26.107:5060: ACK sip:707@10.200.26.107:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK61bc80d5;rportFrom: "Conference Room" ;tag=as6de7994fTo: ;tag=b65273658d3725beContact: Call-ID: 2c60d1190546aed93106a1fe5f8e2e4f@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/707-09842c20 answered asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK763621dc;rportFrom: "Conference Room" ;tag=as0b4e7b16To: Contact: Call-ID: 15d39d7a2a3dfd4b7417fa5c41ee8da4@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10532 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK46907d4f;rportFrom: "Conference Room" ;tag=as1df61756To: ;tag=9cf84f1c622fe1cbCall-ID: 31f848b95945bdc1657450c72b0b5030@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=708 8000 8000 IN IP4 10.200.26.108s=SIP Callc=IN IP4 10.200.26.108t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.108:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.108:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.108, port 5060 Transmitting (NAT) to 10.200.26.108:5060: ACK sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK283a9765;rportFrom: "Conference Room" ;tag=as1df61756To: ;tag=9cf84f1c622fe1cbContact: Call-ID: 31f848b95945bdc1657450c72b0b5030@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/708-097b1d58 answered asterisk-price*CLI> <--- SIP read from 10.200.26.127:5062 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK569753ac;rportFrom: "Conference Room" ;tag=as58e7b3f8To: Call-ID: 14327a4224186e2812da18b15cc51d83@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> -- Called 728 asterisk-price*CLI> <--- SIP read from 10.200.26.127:5062 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK569753ac;rportFrom: "Conference Room" ;tag=as58e7b3f8To: ;tag=4bf7695894a7b9c7Call-ID: 14327a4224186e2812da18b15cc51d83@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> Audio is at 10.200.26.202 port 10544 asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2eb626b9;rportFrom: "Conference Room" ;tag=as7acb1113To: ;tag=8d987e03bf1e6c3aCall-ID: 3e16a7734eeccef320ca05ff419906f6@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=709 8000 8000 IN IP4 10.200.26.109s=SIP Callc=IN IP4 10.200.26.109t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.109:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.109:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.109, port 5060 Transmitting (NAT) to 10.200.26.109:5060: ACK sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK63571fc1;rportFrom: "Conference Room" ;tag=as7acb1113To: ;tag=8d987e03bf1e6c3aContact: Call-ID: 3e16a7734eeccef320ca05ff419906f6@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/709-098625b8 answered asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.129:5060: INVITE sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7abec3b7;rportFrom: "Conference Room" ;tag=as5ac2a55aTo: Contact: Call-ID: 58cdbd4b7893085050a4397c595f74b2@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10544 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 729 asterisk-price*CLI> Audio is at 10.200.26.202 port 10196 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7abec3b7;rportFrom: "Conference Room" ;tag=as5ac2a55aTo: Call-ID: 58cdbd4b7893085050a4397c595f74b2@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7abec3b7;rportFrom: "Conference Room" ;tag=as5ac2a55aTo: ;tag=0a2c87726ef8efdfCall-ID: 58cdbd4b7893085050a4397c595f74b2@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> asterisk-price*CLI> --- (10 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK37663b27;rportFrom: "Conference Room" ;tag=as15bbf524To: ;tag=c04a265e3f704f1aCall-ID: 1a89fb794e3e024057514cf718c53356@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=710 8000 8000 IN IP4 10.200.26.110s=SIP Callc=IN IP4 10.200.26.110t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.110:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.110:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.110, port 5060 Transmitting (NAT) to 10.200.26.110:5060: ACK sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4dc3798a;rportFrom: "Conference Room" ;tag=as15bbf524To: ;tag=c04a265e3f704f1aContact: Call-ID: 1a89fb794e3e024057514cf718c53356@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/710-09825a50 answered asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.130:5060: INVITE sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4fd4eeb5;rportFrom: "Conference Room" ;tag=as160b6042To: Contact: Call-ID: 526bf2d5669e6d170ab6db401d90f3f1@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10196 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> <--- SIP read from 10.200.26.111:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5c3a410d;rportFrom: "Conference Room" ;tag=as196ea359To: ;tag=a50d4bf7a237bb99Call-ID: 2831eb4d191c591361840f6458e5d3a6@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.2.23Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 212v=0o=711 8000 8000 IN IP4 10.200.26.111s=SIP Callc=IN IP4 10.200.26.111t=0 0m=audio 5004 RTP/AVP 0 101a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-11 <-------------> --- (12 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.200.26.111:5004 Found description format PCMU for ID 0 asterisk-price*CLI> Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.200.26.111:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.111, port 5060 Transmitting (NAT) to 10.200.26.111:5060: ACK sip:711@10.200.26.111:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18e5f55d;rportFrom: "Conference Room" ;tag=as196ea359To: ;tag=a50d4bf7a237bb99Contact: Call-ID: 2831eb4d191c591361840f6458e5d3a6@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/711-097bf390 answered asterisk-price*CLI> <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4fd4eeb5;rportFrom: "Conference Room" ;tag=as160b6042To: Call-ID: 526bf2d5669e6d170ab6db401d90f3f1@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk-price*CLI> -- Called 730 asterisk-price*CLI> <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1ec4e490;rportFrom: "Conference Room" ;tag=as2b565667To: ;tag=3309e7020d0fcd0bCall-ID: 56bea4841960214952cae96768b232e9@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=712 8000 8000 IN IP4 10.200.26.112s=SIP Callc=IN IP4 10.200.26.112t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.112:5004 asterisk-price*CLI> Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.112:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.112, port 5060 Transmitting (NAT) to 10.200.26.112:5060: ACK sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7367fdf6;rportFrom: "Conference Room" ;tag=as2b565667To: ;tag=3309e7020d0fcd0bContact: Call-ID: 56bea4841960214952cae96768b232e9@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4fd4eeb5;rportFrom: "Conference Room" ;tag=as160b6042To: ;tag=26990bc34bf52bdfCall-ID: 526bf2d5669e6d170ab6db401d90f3f1@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> -- SIP/712-0984a588 answered asterisk-price*CLI> <--- SIP read from 10.200.26.113:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4b3a09dd;rportFrom: "Conference Room" ;tag=as071911baTo: ;tag=e175f56af1235b7bCall-ID: 3d42cb2e76cad56a2e8b13cd7c4da3a8@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=713 8000 8000 IN IP4 10.200.26.113s=SIP Callc=IN IP4 10.200.26.113t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.113:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.113:5004 list_route: hop: asterisk-price*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.113, port 5060 Transmitting (NAT) to 10.200.26.113:5060: ACK sip:713@10.200.26.113:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6bea6e2d;rportFrom: "Conference Room" ;tag=as071911baTo: ;tag=e175f56af1235b7bContact: Call-ID: 3d42cb2e76cad56a2e8b13cd7c4da3a8@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- -- SIP/713-09854a00 answered asterisk-price*CLI> -- SIP/726-09925e10 is ringing asterisk-price*CLI> -- SIP/727-09915158 is ringing asterisk-price*CLI> -- SIP/729-098e6890 is ringing asterisk-price*CLI> -- SIP/730-098f9f10 is ringing asterisk-price*CLI> Audio is at 10.200.26.202 port 10998 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5376ce11;rportFrom: "Conference Room" ;tag=as0b501a50To: ;tag=f419e9422f9f408eCall-ID: 7768e384367bcf246706bc1e4e8cf8f7@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=715 8000 8000 IN IP4 10.200.26.115s=SIP Callc=IN IP4 10.200.26.115t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.115:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.115:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.115, port 5060 Transmitting (NAT) to 10.200.26.115:5060: ACK sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3217581b;rportFrom: "Conference Room" ;tag=as0b501a50To: ;tag=f419e9422f9f408eContact: Call-ID: 7768e384367bcf246706bc1e4e8cf8f7@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/715-09840390 answered asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.131:5060: INVITE sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK46cc7546;rportFrom: "Conference Room" ;tag=as1680f23bTo: Contact: Call-ID: 2d4bea95182dc92e72f06e8846486803@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10998 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 731 asterisk-price*CLI> Audio is at 10.200.26.202 port 10348 asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK46cc7546;rportFrom: "Conference Room" ;tag=as1680f23bTo: Call-ID: 2d4bea95182dc92e72f06e8846486803@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.200.26.116:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7a8f3fd9;rportFrom: "Conference Room" ;tag=as224a4d42To: ;tag=a289a10e1d865b5eCall-ID: 2095ee5434257d94767ecce318d9aea1@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=716 8000 8000 IN IP4 10.200.26.116s=SIP Callc=IN IP4 10.200.26.116t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.116:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.116:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.116, port 5060 Transmitting (NAT) to 10.200.26.116:5060: ACK sip:716@10.200.26.116:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK14db8b38;rportFrom: "Conference Room" ;tag=as224a4d42To: ;tag=a289a10e1d865b5eContact: Call-ID: 2095ee5434257d94767ecce318d9aea1@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK46cc7546;rportFrom: "Conference Room" ;tag=as1680f23bTo: ;tag=ee9961c063ce91d9Call-ID: 2d4bea95182dc92e72f06e8846486803@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/716-09811930 answered asterisk-price*CLI> -- SIP/731-098930d0 is ringing asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Adding non-codec 0x1 (telephone-event) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.139:5060: INVITE sip:739@10.200.26.139:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6fdd669f;rportFrom: "Conference Room" ;tag=as32c25ba5To: Contact: Call-ID: 22ac922c582f91ee6e3af2305d2e066b@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10348 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> -- Called 739 asterisk-price*CLI> <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK252302e5;rportFrom: "Conference Room" ;tag=as20ccb9eaTo: ;tag=5ef8622c264fa58cCall-ID: 388a13b34ae12be842ac94e465834e11@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=718 8000 8000 IN IP4 10.200.26.118s=SIP Callc=IN IP4 10.200.26.118t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.118:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.118:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.118, port 5060 Transmitting (NAT) to 10.200.26.118:5060: ACK sip:718@10.200.26.118:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6cdff796;rportFrom: "Conference Room" ;tag=as20ccb9eaTo: ;tag=5ef8622c264fa58cContact: Call-ID: 388a13b34ae12be842ac94e465834e11@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> <--- SIP read from 10.200.26.139:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6fdd669f;rportFrom: "Conference Room" ;tag=as32c25ba5To: Call-ID: 22ac922c582f91ee6e3af2305d2e066b@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- SIP/718-097cf358 answered <--- SIP read from 10.200.26.139:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6fdd669f;rportFrom: "Conference Room" ;tag=as32c25ba5To: ;tag=40a68a6b78ecbd37Call-ID: 22ac922c582f91ee6e3af2305d2e066b@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> Audio is at 10.200.26.202 port 10650 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKfd94ec7fcf16ddff;received=10.200.26.105From: "Main Office Conf Room" ;tag=4aed5c1b08e0dbdfTo: ;tag=as38609ba8Call-ID: c05b3f48efc16884@10.200.26.105CSeq: 20224 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Type: application/sdpContent-Length: 209v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10650 RTP/AVP 0 3a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=silenceSupp:off - - - -a=ptime:20a=sendrecv <------------> asterisk-price*CLI> <--- SIP read from 10.200.26.119:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK655a03d0;rportFrom: "Conference Room" ;tag=as7c43023cTo: ;tag=0f835da74832f462Call-ID: 31c5322a4a18c2e1033659987decf745@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=719 8000 8000 IN IP4 10.200.26.119s=SIP Callc=IN IP4 10.200.26.119t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.119:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.119:5004 list_route: hop: asterisk-price*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.119, port 5060 Transmitting (NAT) to 10.200.26.119:5060: ACK sip:719@10.200.26.119:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5bdfa4da;rportFrom: "Conference Room" ;tag=as7c43023cTo: ;tag=0f835da74832f462Contact: Call-ID: 31c5322a4a18c2e1033659987decf745@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- -- SIP/719-09816518 answered asterisk-price*CLI> -- Playing 'beep' (language 'en') asterisk-price*CLI> -- SIP/739-098a31f0 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.120:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4236697d;rportFrom: "Conference Room" ;tag=as20842e79To: ;tag=6b66ac8fe4ef7091Call-ID: 66606fd57172bad153d652c34eff1e0c@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=720 8000 8000 IN IP4 10.200.26.120s=SIP Callc=IN IP4 10.200.26.120t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.120:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.120:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.120, port 5060 Transmitting (NAT) to 10.200.26.120:5060: ACK sip:720@10.200.26.120:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5bd36289;rportFrom: "Conference Room" ;tag=as20842e79To: ;tag=6b66ac8fe4ef7091Contact: Call-ID: 66606fd57172bad153d652c34eff1e0c@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/720-097b8718 answered asterisk-price*CLI> <--- SIP read from 10.200.26.121:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK32d1c0d5;rportFrom: "Conference Room" ;tag=as1ec1d811To: ;tag=63a9650eb3b6b34cCall-ID: 73a9d25308b48d607f4357d04eb7f5e7@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=721 8000 8000 IN IP4 10.200.26.121s=SIP Callc=IN IP4 10.200.26.121t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.121:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.121:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.121, port 5060 Transmitting (NAT) to 10.200.26.121:5060: ACK sip:721@10.200.26.121:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK29fc2617;rportFrom: "Conference Room" ;tag=as1ec1d811To: ;tag=63a9650eb3b6b34cContact: Call-ID: 73a9d25308b48d607f4357d04eb7f5e7@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/721-097b32d8 answered asterisk-price*CLI> <--- SIP read from 10.200.26.139:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6fdd669f;rportFrom: "Conference Room" ;tag=as32c25ba5To: ;tag=40a68a6b78ecbd37Call-ID: 22ac922c582f91ee6e3af2305d2e066b@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> -- SIP/739-098a31f0 is ringing asterisk-price*CLI> <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK78278d2d;rportFrom: "Conference Room" ;tag=as52466f3fTo: ;tag=08ad89fdfe20bbc5Call-ID: 469c859f578805563b3e6e7507a70a7e@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=722 8000 8000 IN IP4 10.200.26.122s=SIP Callc=IN IP4 10.200.26.122t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.122:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.122:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.122, port 5060 Transmitting (NAT) to 10.200.26.122:5060: ACK sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0fabe3c0;rportFrom: "Conference Room" ;tag=as52466f3fTo: ;tag=08ad89fdfe20bbc5Contact: Call-ID: 469c859f578805563b3e6e7507a70a7e@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/722-0993bc70 answered asterisk-price*CLI> <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK21bc8f92;rportFrom: "Conference Room" ;tag=as785f5377To: ;tag=7758fd3287ae981dCall-ID: 5f9551110fe5f3e66ec1aa8849eb04d2@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=724 8000 8000 IN IP4 10.200.26.124s=SIP Callc=IN IP4 10.200.26.124t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.124:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.124:5004 asterisk-price*CLI> list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.124, port 5060 Transmitting (NAT) to 10.200.26.124:5060: ACK sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4135d8b6;rportFrom: "Conference Room" ;tag=as785f5377To: ;tag=7758fd3287ae981dContact: Call-ID: 5f9551110fe5f3e66ec1aa8849eb04d2@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/724-0993fbd8 answered asterisk-price*CLI> <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK46d4df6d;rportFrom: "Conference Room" ;tag=as1ec6dc28To: ;tag=31a7260fa05e0860Call-ID: 46017eb0146a5fd149d9fd114dd69347@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=725 8000 8000 IN IP4 10.200.26.125s=SIP Callc=IN IP4 10.200.26.125t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.125:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.125:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.125, port 5060 Transmitting (NAT) to 10.200.26.125:5060: ACK sip:725@10.200.26.125:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2d501e20;rportFrom: "Conference Room" ;tag=as1ec6dc28To: ;tag=31a7260fa05e0860Contact: Call-ID: 46017eb0146a5fd149d9fd114dd69347@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/725-09941158 answered asterisk-price*CLI> <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK768c18eb;rportFrom: "Conference Room" ;tag=as0069906eTo: ;tag=99587faa32547b7bCall-ID: 7c36f46e6fe0c6c17895542177d8abf2@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=726 8000 8000 IN IP4 10.200.26.126s=SIP Callc=IN IP4 10.200.26.126t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.126:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.126:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.126, port 5060 Transmitting (NAT) to 10.200.26.126:5060: ACK sip:726@10.200.26.126:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0712dbc4;rportFrom: "Conference Room" ;tag=as0069906eTo: ;tag=99587faa32547b7bContact: Call-ID: 7c36f46e6fe0c6c17895542177d8abf2@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/726-09925e10 answered asterisk-price*CLI> <--- SIP read from 10.200.26.127:5062 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK569753ac;rportFrom: "Conference Room" ;tag=as58e7b3f8To: ;tag=4bf7695894a7b9c7Call-ID: 14327a4224186e2812da18b15cc51d83@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=727 8000 8000 IN IP4 10.200.26.127s=SIP Callc=IN IP4 10.200.26.127t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- asterisk-price*CLI> Found RTP audio format 0 Peer audio RTP is at port 10.200.26.127:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.127:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.127, port 5062 Transmitting (NAT) to 10.200.26.127:5062: ACK sip:727@10.200.26.127:5062;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK698145d6;rportFrom: "Conference Room" ;tag=as58e7b3f8To: ;tag=4bf7695894a7b9c7Contact: Call-ID: 14327a4224186e2812da18b15cc51d83@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/727-09915158 answered asterisk-price*CLI> <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7abec3b7;rportFrom: "Conference Room" ;tag=as5ac2a55aTo: ;tag=0a2c87726ef8efdfCall-ID: 58cdbd4b7893085050a4397c595f74b2@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=729 8000 8000 IN IP4 10.200.26.129s=SIP Callc=IN IP4 10.200.26.129t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.129:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.129:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.129, port 5060 Transmitting (NAT) to 10.200.26.129:5060: ACK sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK38018840;rportFrom: "Conference Room" ;tag=as5ac2a55aTo: ;tag=0a2c87726ef8efdfContact: Call-ID: 58cdbd4b7893085050a4397c595f74b2@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/729-098e6890 answered asterisk-price*CLI> <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4fd4eeb5;rportFrom: "Conference Room" ;tag=as160b6042To: ;tag=26990bc34bf52bdfCall-ID: 526bf2d5669e6d170ab6db401d90f3f1@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=730 8000 8000 IN IP4 10.200.26.130s=SIP Callc=IN IP4 10.200.26.130t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.130:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.130:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.130, port 5060 Transmitting (NAT) to 10.200.26.130:5060: ACK sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK28caff2b;rportFrom: "Conference Room" ;tag=as160b6042To: ;tag=26990bc34bf52bdfContact: Call-ID: 526bf2d5669e6d170ab6db401d90f3f1@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/730-098f9f10 answered asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK46cc7546;rportFrom: "Conference Room" ;tag=as1680f23bTo: ;tag=ee9961c063ce91d9Call-ID: 2d4bea95182dc92e72f06e8846486803@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=731 8000 8000 IN IP4 10.200.26.131s=SIP Callc=IN IP4 10.200.26.131t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 10.200.26.131:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.131:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.131, port 5060 Transmitting (NAT) to 10.200.26.131:5060: ACK sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK355c6fc1;rportFrom: "Conference Room" ;tag=as1680f23bTo: ;tag=ee9961c063ce91d9Contact: Call-ID: 2d4bea95182dc92e72f06e8846486803@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/731-098930d0 answered asterisk-price*CLI> <--- SIP read from 10.200.26.139:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6fdd669f;rportFrom: "Conference Room" ;tag=as32c25ba5To: ;tag=40a68a6b78ecbd37Call-ID: 22ac922c582f91ee6e3af2305d2e066b@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 212v=0o=739 8000 8000 IN IP4 10.200.26.139s=SIP Callc=IN IP4 10.200.26.139t=0 0m=audio 5004 RTP/AVP 0 101a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-11 <-------------> --- (12 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.200.26.139:5004 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.200.26.139:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.139, port 5060 Transmitting (NAT) to 10.200.26.139:5060: ACK sip:739@10.200.26.139:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK341258d6;rportFrom: "Conference Room" ;tag=as32c25ba5To: ;tag=40a68a6b78ecbd37Contact: Call-ID: 22ac922c582f91ee6e3af2305d2e066b@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/739-098a31f0 answered asterisk-price*CLI> Audio is at 10.200.26.202 port 10650 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP <--- Reliably Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKfd94ec7fcf16ddff;received=10.200.26.105From: "Main Office Conf Room" ;tag=4aed5c1b08e0dbdfTo: ;tag=as38609ba8Call-ID: c05b3f48efc16884@10.200.26.105CSeq: 20224 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Type: application/sdpContent-Length: 209v=0o=root 10024 10025 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10650 RTP/AVP 0 3a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=silenceSupp:off - - - -a=ptime:20a=sendrecv <------------> asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> ACK sip:760@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKc40f629c54d05ee8From: "Main Office Conf Room" ;tag=4aed5c1b08e0dbdfTo: ;tag=as38609ba8Contact: Proxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:760@10.200.26.202", nonce="0c3f45ae", response="10daf425cc59318cf937abc8bee0abaf"Call-ID: c05b3f48efc16884@10.200.26.105CSeq: 20224 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (12 headers 0 lines) --- asterisk-price*CLI> Retransmitting #1 (NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK763621dc;rportFrom: "Conference Room" ;tag=as0b4e7b16To: Contact: Call-ID: 15d39d7a2a3dfd4b7417fa5c41ee8da4@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10532 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> Retransmitting #2 (NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK763621dc;rportFrom: "Conference Room" ;tag=as0b4e7b16To: Contact: Call-ID: 15d39d7a2a3dfd4b7417fa5c41ee8da4@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10532 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> Retransmitting #5 (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK024acab2;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbContact: Call-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=59651Subscription-state: terminated;reason=noresourceC asterisk-price*CLI> ontent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 16SIP/2.0 200 OK --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK024acab2;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbCall-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> Retransmitting #3 (NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK763621dc;rportFrom: "Conference Room" ;tag=as0b4e7b16To: Contact: Call-ID: 15d39d7a2a3dfd4b7417fa5c41ee8da4@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:16:10 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesCall-Info: answer-after=0Content-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10532 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> BYE sip:760@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKaa89d5b1487d6158From: "Main Office Conf Room" ;tag=4aed5c1b08e0dbdfTo: ;tag=as38609ba8Proxy-Authorization: Digest username="705", realm="asterisk", algorithm=MD5, uri="sip:760@10.200.26.202", nonce="0c3f45ae", response="c13aae057032c59da2664b3507fe893b"Call-ID: c05b3f48efc16884@10.200.26.105CSeq: 20225 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.200.26.105 : 5060 (NAT) <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKaa89d5b1487d6158;received=10.200.26.105From: "Main Office Conf Room" ;tag=4aed5c1b08e0dbdfTo: ;tag=as38609ba8Call-ID: c05b3f48efc16884@10.200.26.105CSeq: 20225 BYEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> Scheduling destruction of SIP dialog '68e28dd61a2046a0577d488b2efa0df1@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.101, port 5060 Reliably Transmitting (NAT) to 10.200.26.101:5060: BYE sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5ec54f36;rportFrom: "Conference Room" ;tag=as5ac1abdaTo: ;tag=37f37336cb895347Call-ID: 68e28dd61a2046a0577d488b2efa0df1@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5ec54f36;rportFrom: "Conference Room" ;tag=as5ac1abdaTo: ;tag=37f37336cb895347Call-ID: 68e28dd61a2046a0577d488b2efa0df1@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> Really destroying SIP dialog '68e28dd61a2046a0577d488b2efa0df1@66.111.122.20' Method: INVITE asterisk-price*CLI> Scheduling destruction of SIP dialog '1596367e0f152ef076c94b595c7c8163@66.111.122.20' in 32000 ms (Method: INVITE) asterisk-price*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.102, port 5060 Reliably Transmitting (NAT) to 10.200.26.102:5060: BYE sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1c70f778;rportFrom: "Conference Room" ;tag=as72155756To: ;tag=5309a8024e0fee0bCall-ID: 1596367e0f152ef076c94b595c7c8163@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> Scheduling destruction of SIP dialog '734d61614b6f97c77e38342616025178@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.103, port 5060 Reliably Transmitting (NAT) to 10.200.26.103:5060: BYE sip:703@10.200.26.103:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK44ef128d;rportFrom: "Conference Room" ;tag=as3490df47To: ;tag=b609ab02e01f617bCall-ID: 734d61614b6f97c77e38342616025178@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '6aae31bb3ea8c4c17178015a36fb4fdf@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.104, port 5060 Reliably Transmitting (NAT) to 10.200.26.104:5060: BYE sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4feda904;rportFrom: "Conference Room" ;tag=as2457465eTo: ;tag=069f8bc9ccfb2dd5Call-ID: 6aae31bb3ea8c4c17178015a36fb4fdf@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1c70f778;rportFrom: "Conference Room" ;tag=as72155756To: ;tag=5309a8024e0fee0bCall-ID: 1596367e0f152ef076c94b595c7c8163@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK44ef128d;rportFrom: "Conference Room" ;tag=as3490df47To: ;tag=b609ab02e01f617bCall-ID: 734d61614b6f97c77e38342616025178@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- -- Hungup 'Zap/pseudo-473125140' Scheduling destruction of SIP dialog '60e8999a1938524d2ec9c00e6001926f@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.106, port 5060 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.106:5060: BYE sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK629dbfe8;rportFrom: "Conference Room" ;tag=as787c678cTo: ;tag=208954d6317f9c69Call-ID: 60e8999a1938524d2ec9c00e6001926f@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '2c60d1190546aed93106a1fe5f8e2e4f@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.107, port 5060 Reliably Transmitting (NAT) to 10.200.26.107:5060: BYE sip:707@10.200.26.107:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18985b07;rportFrom: "Conference Room" ;tag=as6de7994fTo: ;tag=b65273658d3725beCall-ID: 2c60d1190546aed93106a1fe5f8e2e4f@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '31f848b95945bdc1657450c72b0b5030@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.108, port 5060 Reliably Transmitting (NAT) to 10.200.26.108:5060: BYE sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK153fd74a;rportFrom: "Conference Room" ;tag=as1df61756To: ;tag=9cf84f1c622fe1cbCall-ID: 31f848b95945bdc1657450c72b0b5030@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4feda904;rportFrom: "Conference Room" ;tag=as2457465eTo: ;tag=069f8bc9ccfb2dd5Call-ID: 6aae31bb3ea8c4c17178015a36fb4fdf@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- Really destroying SIP dialog '6aae31bb3ea8c4c17178015a36fb4fdf@66.111.122.20' Method: INVITE Really destroying SIP dialog '734d61614b6f97c77e38342616025178@66.111.122.20' Method: INVITE Really destroying SIP dialog '1596367e0f152ef076c94b595c7c8163@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK629dbfe8;rportFrom: "Conference Room" ;tag=as787c678cTo: ;tag=208954d6317f9c69Call-ID: 60e8999a1938524d2ec9c00e6001926f@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18985b07;rportFrom: "Conference Room" ;tag=as6de7994fTo: ;tag=b65273658d3725beCall-ID: 2c60d1190546aed93106a1fe5f8e2e4f@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK153fd74a;rportFrom: "Conference Room" ;tag=as1df61756To: ;tag=9cf84f1c622fe1cbCall-ID: 31f848b95945bdc1657450c72b0b5030@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '31f848b95945bdc1657450c72b0b5030@66.111.122.20' Method: INVITE Really destroying SIP dialog '2c60d1190546aed93106a1fe5f8e2e4f@66.111.122.20' Method: INVITE Really destroying SIP dialog '60e8999a1938524d2ec9c00e6001926f@66.111.122.20' Method: INVITE asterisk-price*CLI> Scheduling destruction of SIP dialog '3e16a7734eeccef320ca05ff419906f6@66.111.122.20' in 32000 ms (Method: INVITE) asterisk-price*CLI> set_destination: Parsing for address/port to send to asterisk-price*CLI> set_destination: set destination to 10.200.26.109, port 5060 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.109:5060: BYE sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18ddf8f8;rportFrom: "Conference Room" ;tag=as7acb1113To: ;tag=8d987e03bf1e6c3aCall-ID: 3e16a7734eeccef320ca05ff419906f6@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> <--- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK18ddf8f8;rportFrom: "Conference Room" ;tag=as7acb1113To: ;tag=8d987e03bf1e6c3aCall-ID: 3e16a7734eeccef320ca05ff419906f6@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> Scheduling destruction of SIP dialog '1a89fb794e3e024057514cf718c53356@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.110, port 5060 Reliably Transmitting (NAT) to 10.200.26.110:5060: BYE sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK04d32fd3;rportFrom: "Conference Room" ;tag=as15bbf524To: ;tag=c04a265e3f704f1aCall-ID: 1a89fb794e3e024057514cf718c53356@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '2831eb4d191c591361840f6458e5d3a6@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.111, port 5060 asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.111:5060: BYE sip:711@10.200.26.111:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d5dc054;rportFrom: "Conference Room" ;tag=as196ea359To: ;tag=a50d4bf7a237bb99Call-ID: 2831eb4d191c591361840f6458e5d3a6@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '56bea4841960214952cae96768b232e9@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.112, port 5060 Reliably Transmitting (NAT) to 10.200.26.112:5060: BYE sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK46c80ade;rportFrom: "Conference Room" ;tag=as2b565667To: ;tag=3309e7020d0fcd0bCall-ID: 56bea4841960214952cae96768b232e9@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '3d42cb2e76cad56a2e8b13cd7c4da3a8@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.113, port 5060 Reliably Transmitting (NAT) to 10.200.26.113:5060: BYE sip:713@10.200.26.113:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK615f20e1;rportFrom: "Conference Room" ;tag=as071911baTo: ;tag=e175f56af1235b7bCall-ID: 3d42cb2e76cad56a2e8b13cd7c4da3a8@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '7768e384367bcf246706bc1e4e8cf8f7@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.115, port 5060 Reliably Transmitting (NAT) to 10.200.26.115:5060: BYE sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK168d02da;rportFrom: "Conference Room" ;tag=as0b501a50To: ;tag=f419e9422f9f408eCall-ID: 7768e384367bcf246706bc1e4e8cf8f7@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '3e16a7734eeccef320ca05ff419906f6@66.111.122.20' Method: INVITE asterisk-price*CLI> <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK04d32fd3;rportFrom: "Conference Room" ;tag=as15bbf524To: ;tag=c04a265e3f704f1aCall-ID: 1a89fb794e3e024057514cf718c53356@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.111:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d5dc054;rportFrom: "Conference Room" ;tag=as196ea359To: ;tag=a50d4bf7a237bb99Call-ID: 2831eb4d191c591361840f6458e5d3a6@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.2.23Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.113:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK615f20e1;rportFrom: "Conference Room" ;tag=as071911baTo: ;tag=e175f56af1235b7bCall-ID: 3d42cb2e76cad56a2e8b13cd7c4da3a8@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '3d42cb2e76cad56a2e8b13cd7c4da3a8@66.111.122.20' Method: INVITE Really destroying SIP dialog '2831eb4d191c591361840f6458e5d3a6@66.111.122.20' Method: INVITE Really destroying SIP dialog '1a89fb794e3e024057514cf718c53356@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK46c80ade;rportFrom: "Conference Room" ;tag=as2b565667To: ;tag=3309e7020d0fcd0bCall-ID: 56bea4841960214952cae96768b232e9@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Scheduling destruction of SIP dialog '2095ee5434257d94767ecce318d9aea1@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.116, port 5060 Reliably Transmitting (NAT) to 10.200.26.116:5060: BYE sip:716@10.200.26.116:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0de93347;rportFrom: "Conference Room" ;tag=as224a4d42To: ;tag=a289a10e1d865b5eCall-ID: 2095ee5434257d94767ecce318d9aea1@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK168d02da;rportFrom: "Conference Room" ;tag=as0b501a50To: ;tag=f419e9422f9f408eCall-ID: 7768e384367bcf246706bc1e4e8cf8f7@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> Scheduling destruction of SIP dialog '315c17210e59ed7b1c5e65d209f10c79@66.111.122.20' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.200.26.117:5060: CANCEL sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7a0a9167;rportFrom: "Conference Room" ;tag=as53bc76acTo: Call-ID: 315c17210e59ed7b1c5e65d209f10c79@66.111.122.20CSeq: 102 CANCELUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> Scheduling destruction of SIP dialog '315c17210e59ed7b1c5e65d209f10c79@66.111.122.20' in 32000 ms (Method: INVITE) asterisk-price*CLI> Really destroying SIP dialog '7768e384367bcf246706bc1e4e8cf8f7@66.111.122.20' Method: INVITE Really destroying SIP dialog '56bea4841960214952cae96768b232e9@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.116:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0de93347;rportFrom: "Conference Room" ;tag=as224a4d42To: ;tag=a289a10e1d865b5eCall-ID: 2095ee5434257d94767ecce318d9aea1@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> Scheduling destruction of SIP dialog '388a13b34ae12be842ac94e465834e11@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.118, port 5060 Reliably Transmitting (NAT) to 10.200.26.118:5060: BYE sip:718@10.200.26.118:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7e6392c5;rportFrom: "Conference Room" ;tag=as20ccb9eaTo: ;tag=5ef8622c264fa58cCall-ID: 388a13b34ae12be842ac94e465834e11@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '31c5322a4a18c2e1033659987decf745@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.119, port 5060 Reliably Transmitting (NAT) to 10.200.26.119:5060: BYE sip:719@10.200.26.119:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5e90ff43;rportFrom: "Conference Room" ;tag=as7c43023cTo: ;tag=0f835da74832f462Call-ID: 31c5322a4a18c2e1033659987decf745@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '66606fd57172bad153d652c34eff1e0c@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.120, port 5060 Reliably Transmitting (NAT) to 10.200.26.120:5060: BYE sip:720@10.200.26.120:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK69094a7c;rportFrom: "Conference Room" ;tag=as20842e79To: ;tag=6b66ac8fe4ef7091Call-ID: 66606fd57172bad153d652c34eff1e0c@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Really destroying SIP dialog '2095ee5434257d94767ecce318d9aea1@66.111.122.20' Method: INVITE asterisk-price*CLI> Scheduling destruction of SIP dialog '73a9d25308b48d607f4357d04eb7f5e7@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.121, port 5060 Reliably Transmitting (NAT) to 10.200.26.121:5060: BYE sip:721@10.200.26.121:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK67f429af;rportFrom: "Conference Room" ;tag=as1ec1d811To: ;tag=63a9650eb3b6b34cCall-ID: 73a9d25308b48d607f4357d04eb7f5e7@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '469c859f578805563b3e6e7507a70a7e@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.122, port 5060 Reliably Transmitting (NAT) to 10.200.26.122:5060: BYE sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0e290375;rportFrom: "Conference Room" ;tag=as52466f3fTo: ;tag=08ad89fdfe20bbc5Call-ID: 469c859f578805563b3e6e7507a70a7e@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '5f9551110fe5f3e66ec1aa8849eb04d2@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.124, port 5060 Reliably Transmitting (NAT) to 10.200.26.124:5060: BYE sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK584afad6;rportFrom: "Conference Room" ;tag=as785f5377To: ;tag=7758fd3287ae981dCall-ID: 5f9551110fe5f3e66ec1aa8849eb04d2@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7e6392c5;rportFrom: "Conference Room" ;tag=as20ccb9eaTo: ;tag=5ef8622c264fa58cCall-ID: 388a13b34ae12be842ac94e465834e11@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.119:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5e90ff43;rportFrom: "Conference Room" ;tag=as7c43023cTo: ;tag=0f835da74832f462Call-ID: 31c5322a4a18c2e1033659987decf745@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.120:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK69094a7c;rportFrom: "Conference Room" ;tag=as20842e79To: ;tag=6b66ac8fe4ef7091Call-ID: 66606fd57172bad153d652c34eff1e0c@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> Really destroying SIP dialog '66606fd57172bad153d652c34eff1e0c@66.111.122.20' Method: INVITE Really destroying SIP dialog '31c5322a4a18c2e1033659987decf745@66.111.122.20' Method: INVITE Really destroying SIP dialog '388a13b34ae12be842ac94e465834e11@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.121:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK67f429af;rportFrom: "Conference Room" ;tag=as1ec1d811To: ;tag=63a9650eb3b6b34cCall-ID: 73a9d25308b48d607f4357d04eb7f5e7@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7a0a9167;rportFrom: "Conference Room" ;tag=as53bc76acTo: ;tag=39e792d76ec98df4Call-ID: 315c17210e59ed7b1c5e65d209f10c79@66.111.122.20CSeq: 102 CANCELUser-Agent: Grandstream GXP2000 1.1.4.18Supported: replaces, timerContent-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0e290375;rportFrom: "Conference Room" ;tag=as52466f3fTo: ;tag=08ad89fdfe20bbc5Call-ID: 469c859f578805563b3e6e7507a70a7e@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> Really destroying SIP dialog '469c859f578805563b3e6e7507a70a7e@66.111.122.20' Method: INVITE Really destroying SIP dialog '73a9d25308b48d607f4357d04eb7f5e7@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 487 Request CancelledVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7a0a9167;rportFrom: "Conference Room" ;tag=as53bc76acTo: ;tag=39e792d76ec98df4Call-ID: 315c17210e59ed7b1c5e65d209f10c79@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 10.200.26.117:5060: ACK sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7a0a9167;rportFrom: "Conference Room" ;tag=as53bc76acTo: ;tag=39e792d76ec98df4Contact: Call-ID: 315c17210e59ed7b1c5e65d209f10c79@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK584afad6;rportFrom: "Conference Room" ;tag=as785f5377To: ;tag=7758fd3287ae981dCall-ID: 5f9551110fe5f3e66ec1aa8849eb04d2@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> Really destroying SIP dialog '5f9551110fe5f3e66ec1aa8849eb04d2@66.111.122.20' Method: INVITE Really destroying SIP dialog '315c17210e59ed7b1c5e65d209f10c79@66.111.122.20' Method: INVITE asterisk-price*CLI> Scheduling destruction of SIP dialog '46017eb0146a5fd149d9fd114dd69347@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.125, port 5060 Reliably Transmitting (NAT) to 10.200.26.125:5060: BYE sip:725@10.200.26.125:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK474c9ebf;rportFrom: "Conference Room" ;tag=as1ec6dc28To: ;tag=31a7260fa05e0860Call-ID: 46017eb0146a5fd149d9fd114dd69347@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> Scheduling destruction of SIP dialog '7c36f46e6fe0c6c17895542177d8abf2@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.126, port 5060 Reliably Transmitting (NAT) to 10.200.26.126:5060: BYE sip:726@10.200.26.126:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK47ca5c5f;rportFrom: "Conference Room" ;tag=as0069906eTo: ;tag=99587faa32547b7bCall-ID: 7c36f46e6fe0c6c17895542177d8abf2@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> Scheduling destruction of SIP dialog '14327a4224186e2812da18b15cc51d83@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.127, port 5062 Reliably Transmitting (NAT) to 10.200.26.127:5062: BYE sip:727@10.200.26.127:5062;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK51e5b35e;rportFrom: "Conference Room" ;tag=as58e7b3f8T asterisk-price*CLI> o: ;tag=4bf7695894a7b9c7Call-ID: 14327a4224186e2812da18b15cc51d83@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK474c9ebf;rportFrom: "Conference Room" ;tag=as1ec6dc28To: ;tag=31a7260fa05e0860Call-ID: 46017eb0146a5fd149d9fd114dd69347@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK47ca5c5f;rportFrom: "Conference Room" ;tag=as0069906eTo: ;tag=99587faa32547b7bCall-ID: 7c36f46e6fe0c6c17895542177d8abf2@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> Scheduling destruction of SIP dialog '15d39d7a2a3dfd4b7417fa5c41ee8da4@66.111.122.20' in 32000 ms (Method: INVITE) Really destroying SIP dialog '7c36f46e6fe0c6c17895542177d8abf2@66.111.122.20' Method: INVITE Really destroying SIP dialog '46017eb0146a5fd149d9fd114dd69347@66.111.122.20' Method: INVITE asterisk-price*CLI> Scheduling destruction of SIP dialog '58cdbd4b7893085050a4397c595f74b2@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.129, port 5060 Reliably Transmitting (NAT) to 10.200.26.129:5060: BYE sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3fa755ec;rportFrom: "Conference Room" ;tag=as5ac2a55aTo: ;tag=0a2c87726ef8efdfCall-ID: 58cdbd4b7893085050a4397c595f74b2@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> Scheduling destruction of SIP dialog '526bf2d5669e6d170ab6db401d90f3f1@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.130, port 5060 Reliably Transmitting (NAT) to 10.200.26.130:5060: BYE sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK45a34820;rportFrom: "Conference Room" ;tag=as160b6042To: ;tag=26990bc34bf52bdfCall-ID: 526bf2d5669e6d170ab6db401d90f3f1@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '2d4bea95182dc92e72f06e8846486803@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.131, port 5060 Reliably Transmitting (NAT) to 10.200.26.131:5060: BYE sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1dbee2d9;rportFrom: "Conference Room" ;tag=as1680f23bTo: ;tag=ee9961c063ce91d9Call-ID: 2d4bea95182dc92e72f06e8846486803@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '22ac922c582f91ee6e3af2305d2e066b@66.111.122.20' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.139, port 5060 Reliably Transmitting (NAT) to 10.200.26.139:5060: BYE sip:739@10.200.26.139:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2c638a12;rportFrom: "Conference Room" ;tag=as32c25ba5To: ;tag=40a68a6b78ecbd37Call-ID: 22ac922c582f91ee6e3af2305d2e066b@66.111.122.20CSeq: 103 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- == Spawn extension (from-internal, 760, 4) exited non-zero on 'SIP/705-097a4ca0' asterisk-price*CLI> <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3fa755ec;rportFrom: "Conference Room" ;tag=as5ac2a55aTo: ;tag=0a2c87726ef8efdfCall-ID: 58cdbd4b7893085050a4397c595f74b2@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK45a34820;rportFrom: "Conference Room" ;tag=as160b6042To: ;tag=26990bc34bf52bdfCall-ID: 526bf2d5669e6d170ab6db401d90f3f1@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 10.200.26.139:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2c638a12;rportFrom: "Conference Room" ;tag=as32c25ba5To: ;tag=40a68a6b78ecbd37Call-ID: 22ac922c582f91ee6e3af2305d2e066b@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '22ac922c582f91ee6e3af2305d2e066b@66.111.122.20' Method: INVITE Really destroying SIP dialog '526bf2d5669e6d170ab6db401d90f3f1@66.111.122.20' Method: INVITE Really destroying SIP dialog '58cdbd4b7893085050a4397c595f74b2@66.111.122.20' Method: INVITE <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1dbee2d9;rportFrom: "Conference Room" ;tag=as1680f23bTo: ;tag=ee9961c063ce91d9Call-ID: 2d4bea95182dc92e72f06e8846486803@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> -- Executing [h@from-internal:1] Hangup("SIP/705-097a4ca0", "") in new stack asterisk-price*CLI> == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/705-097a4ca0' asterisk-price*CLI> <--- SIP read from 10.200.26.127:5062 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK51e5b35e;rportFrom: "Conference Room" ;tag=as58e7b3f8To: ;tag=4bf7695894a7b9c7Call-ID: 14327a4224186e2812da18b15cc51d83@66.111.122.20CSeq: 103 BYEUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '2d4bea95182dc92e72f06e8846486803@66.111.122.20' Method: INVITE asterisk-price*CLI> Really destroying SIP dialog '14327a4224186e2812da18b15cc51d83@66.111.122.20' Method: INVITE asterisk-price*CLI> Really destroying SIP dialog 'c05b3f48efc16884@10.200.26.105' Method: BYE asterisk-price*CLI> Retransmitting #6 (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK024acab2;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbContact: Call-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=59651Subscription-state: terminated;reason=noresourceC asterisk-price*CLI> ontent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 16SIP/2.0 200 OK --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK024acab2;rportFrom: "8014525335" ;tag=as614446beTo: ;tag=5dc52ad73787efbbCall-ID: 2665885949c42d88012168484f1e39f1@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> Really destroying SIP dialog '1b85e9a7609ea1e74a336cd7023b28b5@66.111.122.20' Method: INVITE asterisk-price*CLI> [Jul 20 12:16:20] WARNING[10061]: chan_sip.c:1920 retrans_pkt: Maximum retries exceeded on transmission 2665885949c42d88012168484f1e39f1@66.111.122.20 for seqno 103 (Non-critical Request) asterisk-price*CLI> Really destroying SIP dialog '2665885949c42d88012168484f1e39f1@66.111.122.20' Method: REFER asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) Zap/2-1 900@from-zaptel:6 Up Parked Call() SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) 3 active channels 1 active call asterisk-price*CLI> Really destroying SIP dialog '15d39d7a2a3dfd4b7417fa5c41ee8da4@66.111.122.20' Method: INVITE asterisk-price*CLI> show channels -- Stopped music on hold on Zap/2-1 -- Added extension 'SIP/705' priority 1 to park-dial == Timeout for Zap/2-1 parked on 201. Returning to park-dial,SIP/705,1 asterisk-price*CLI> show channels -- Executing [SIP/705@park-dial:1] Dial("Zap/2-1", "SIP/705||t") in new stack asterisk-price*CLI> show channels Audio is at 10.200.26.202 port 10162 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.200.26.105:5060: INVITE sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK515776d2;rportFrom: "8014525335" ;tag=as5962a73dTo: Contact: Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 20 Jul 2007 18:17:15 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 265v=0o=root 10024 10024 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10162 RTP/AVP 0 3 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- Reliably Transmitting (NAT) to 10.200.26.127:5060: NOTIFY sip:727@10.200.26.127:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7e1918f1;rportFrom: ;tag=as0fac1be1To: "Conference Room" ;tag=1e94d3e74ab057e7Contact: Call-ID: cc74970ec691e7e1@10.200.26.127CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 727 Reliably Transmitting (NAT) to 10.200.26.131:5060: NOTIFY sip:731@10.200.26.131:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4831ec28;rportFrom: ;tag=as44d1397bTo: "Lobby" ;tag=99f4161d5df43d88Contact: Call-ID: 62279987519ceb08@10.200.26.131CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> show channels Extension Changed 201 new state Idle for Notify User 731 Reliably Transmitting (NAT) to 10.200.26.130:5060: NOTIFY sip:730@10.200.26.130:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK616716ca;rportFrom: ;tag=as73346657To: "Foyer" ;tag=bee43bdcf28e33c2Contact: Call-ID: 0c17b027f44a0b16@10.200.26.130CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK515776d2;rportFrom: "8014525335" ;tag=as5962a73dTo: C asterisk-price*CLI> show channels all-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Extension Changed 201 new state Idle for Notify User 730 Reliably Transmitting (NAT) to 10.200.26.129:5060: NOTIFY sip:729@10.200.26.129:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6327686b;rportFrom: ;tag=as0b80a390To: "Will Jaimez" ;tag=18b4544c3b75f992Contact: Call-ID: 46071ba670da2a07@10.200.26.129CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 729 Reliably Transmitting (NAT) to 10.200.26.117:5060: NOTIFY sip:717@10.200.26.117:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK78c24c89;rportFrom: ;tag=as5ab9c51aTo: "Ken Williams" ;tag=85b710412fc42025Contact: Call-ID: d416754dbbee5bda@10.200.26.117CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK515776d2;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- asterisk-price*CLI> show channels Extension Changed 201 new state Idle for Notify User 717 Reliably Transmitting (NAT) to 10.200.26.126:5060: NOTIFY sip:726@10.200.26.126:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2c9db4a3;rportFrom: ;tag=as3f56cc30To: "Bob Teny" ;tag=94c4908c7746b742Contact: Call-ID: 8e17d427fa4a6426@10.200.26.126CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 726 Reliably Transmitting (NAT) to 10.200.26.125:5060: NOTIFY sip:725@10.200.26.125:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK65a315f9;rportFrom: ;tag=as38f74437To: "Front Desk" ;tag=1bd437ccfd6ebe92Contact: Call-ID: 62473a6832fbae08@10.200.26.125CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 asterisk-price*CLI> show channels terminated --- Extension Changed 201 new state Idle for Notify User 725 Reliably Transmitting (NAT) to 10.200.26.124:5060: NOTIFY sip:724@10.200.26.124:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3ec99e12;rportFrom: ;tag=as35039685To: "Bobby Houston" ;tag=79c4178cb056f172Contact: Call-ID: 8d27b397d8fca0f7@10.200.26.124CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> show channels<--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4831ec28;rportFrom: ;tag=as44d1397bTo: "Lobby" ;tag=99f4161d5df43d88Call-ID: 62279987519ceb08@10.200.26.131CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> Extension Changed 201 new state Idle for Notify User 724 Reliably Transmitting (NAT) to 10.200.26.121:5060: NOTIFY sip:721@10.200.26.121:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7ee353fd;rportFrom: ;tag=as52c9f68fTo: "Danny D'Ambrosio" ;tag=91e4dffc190f5ae1Contact: Call-ID: 8c47d4781d5ccab6@10.200.26.121CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 721 Reliably Transmitting (NAT) to 10.200.26.122:5060: NOTIFY sip:722@10.200.26.122:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3d241cbf;rportFrom: ;tag=as3bd2c35dTo: "Jake Erramouspe" ;tag=d0e49efc170f18e1Contact: Call-ID: a94732785b5c49b6@10.200.26.122CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 722 Reliably Transmitting (NAT) to 10.200.26.120:5060: NOTIFY sip:720@10.200.26.120:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1ce0fcfa;rportFrom: ;tag=as26caf817To: "Darek Martinez" ;tag=1ae4770db0f4d3b8Contact: Call-ID: 0067f749305becc6@10.200.26.120CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 720 Reliably Transmitting (NAT) to 10.200.26.119:5060: NOTIFY sip:719@10.200.26.119:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK146011cd;rportFrom: ;tag=as4d16fad1To: "Randy Leffler" ;tag=9e94dba4740eb5d1Contact: Call-ID: 2f273697fcfc87f7@10.200.26.119CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- asterisk-price*CLI> show channels --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.130:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK616716ca;rportFrom: ;tag=as73346657To: "Foyer" ;tag=bee43bdcf28e33c2Call-ID: 0c17b027f44a0b16@10.200.26.130CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.129:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6327686b;rportFrom: ;tag=as0b80a390To: "Will Jaimez" ;tag=18b4544c3b75f992Call-ID: 46071ba670da2a07@10.200.26.129CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Extension Changed 201 new state Idle for Notify User 719 <--- SIP read from 10.200.26.126:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2c9db4a3;rportFrom: ;tag=as3f56cc30To: "Bob Teny" ;tag=94c4908c7746b742Call-ID: 8e17d427fa4a6426@10.200.26.126CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK65a315f9;rportFrom: ;tag=as38f74437To: "Front Desk" ;tag=1bd437ccfd6ebe92Call-ID: 62473a6832fbae08@10.200.26.125CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.124:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3ec99e12;rportFrom: ;tag=as35039685To: "Bobby Houston" ;tag=79c4178cb056f172Call-ID: 8d27b397d8fca0f7@10.200.26.124CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Reliably Transmitting (NAT) to 10.200.26.118:5060: NOTIFY sip:718@10.200.26.118:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7cf830b7;rportFrom: ;tag=as501b143eTo: "Chris / Ed" ;tag=f9a4973cd07533c2Contact: Call-ID: 8647dd68f4fb0f08@10.200.26.118CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 718 <--- SIP read from 10.200.26.121:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7ee353fd;rportFrom: ;tag=as52c9f68fTo: "Danny D'Ambrosio" ;tag=91e4dffc190f5ae1Call-ID: 8c47d4781d5ccab6@10.200.26.121CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.122:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3d241cbf;rportFrom: ;tag=as3bd2c35dTo: "Jake Erramouspe" ;tag=d0e49efc170f18e1Call-ID: a94732785b5c49b6@10.200.26.122CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Reliably Transmitting (NAT) to 10.200.26.116:5060: NOTIFY sip:716@10.200.26.116:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7fc8536f;rportFrom: ;tag=as71101529To: "Shop Kitchen" ;tag=3ea41b3c727533c2 asterisk-price*CLI> show channels Contact: Call-ID: 484791787a5c06b6@10.200.26.116CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 716 Reliably Transmitting (NAT) to 10.200.26.111:5060: NOTIFY sip:711@10.200.26.111:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4a030887;rportFrom: ;tag=as4dacd753To: "Center Shop" ;tag=cf7dedb2b53fa60fContact: Call-ID: 43ce35395af9fd6f@10.200.26.111CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 711 Reliably Transmitting (NAT) to 10.200.26.115:5060: NOTIFY sip:715@10.200.26.115:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6a397f9c;rportFrom: ;tag=as039135acTo: "Jerry Wright" ;tag=34b4537c3d36fe42Contact: Call-ID: af47d878926c2027@10.200.26.115CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- <--- SIP read from 10.200.26.120:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1ce0fcfa;rportFrom: ;tag=as26caf817To: "Darek Martinez" ;tag=1ae4770db0f4d3b8Call-ID: 0067f749305becc6@10.200.26.120CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Extension Changed 201 new state Idle for Notify User 715 <--- SIP read from 10.200.26.119:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK146011cd;rportFrom: ;tag=as4d16fad1To: "Randy Leffler" ;tag=9e94dba4740eb5d1Call-ID: 2f273697fcfc87f7@10.200.26.119CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Reliably Transmitting (NAT) to 10.200.26.113:5060: NOTIFY sip:713@10.200.26.113:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2e35b608;rportFrom: ;tag=as05b1e1acTo: "Front Shop" ;tag=37a152d9516ad0eeContact: Call-ID: 49e06e58c4d76003@10.200.26.113CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 713 Reliably Transmitting (NAT) to 10.200.26.112:5060: NOTIFY sip:712@10.200.26.112:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6a091190;rportFrom: ;tag=as0b0c2072To: "Lorrie Blake" ;tag=d3947f943676b5a2Contact: Call-ID: 0d175327994ac226@10.200.26.112CSeq: 117 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- <--- SIP read from 10.200.26.118:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7cf830b7;rportFrom: ;tag=as501b143eTo: "Chris / Ed" ;tag=f9a4973cd07533c2Call-ID: 8647dd68f4fb0f08@10.200.26.118CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Extension Changed 201 new state Idle for Notify User 712 Reliably Transmitting (NAT) to 10.200.26.110:5060: NOTIFY sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3468afb7;rportFrom: ;tag=as3b8bd8b5To: "Shawn Norton" ;tag=d481dbf1f79a741fContact: Call-ID: 0ca0cf97a319ed9e@10.200.26.110CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 710 Reliably Transmitting (NAT) to 10.200.26.109:5060: NOTIFY sip:709@10.200.26.109:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK284b98cd;rportFrom: ;tag=as24dbb862To: "Alan Young" ;tag=597133f1324431aeContact: Call-ID: ace04268693825af@10.200.26.109CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- <--- SIP read from 10.200.26.116:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7fc8536f;rportFrom: ;tag=as71101529To: "Shop Kitchen" ;tag=3ea41b3c727533c2Call-ID: 484791787a5c06b6@10.200.26.116CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 asterisk-price*CLI> show channels <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Extension Changed 201 new state Idle for Notify User 709 Reliably Transmitting (NAT) to 10.200.26.108:5060: NOTIFY sip:708@10.200.26.108:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK72b2ddf9;rportFrom: ;tag=as23529257To: "Jake Ori" ;tag=3b5457d35e7edd92Contact: Call-ID: ae07f2b6183b61b5@10.200.26.108CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 708 <--- SIP read from 10.200.26.111:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4a030887;rportFrom: ;tag=as4dacd753To: "Center Shop" ;tag=cf7dedb2b53fa60fCall-ID: 43ce35395af9fd6f@10.200.26.111CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.2.23Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.115:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6a397f9c;rportFrom: ;tag=as039135acTo: "Jerry Wright" ;tag=34b4537c3d36fe42Call-ID: af47d878926c2027@10.200.26.115CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.127:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7e1918f1;rportFrom: ;tag=as0fac1be1To: "Conference Room" ;tag=1e94d3e74ab057e7Call-ID: cc74970ec691e7e1@10.200.26.127CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.113:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2e35b608;rportFrom: ;tag=as05b1e1acTo: "Front Shop" ;tag=37a152d9516ad0eeCall-ID: 49e06e58c4d76003@10.200.26.113CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Reliably Transmitting (NAT) to 10.200.26.107:5060: NOTIFY sip:707@10.200.26.107:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK413f2bef;rportFrom: ;tag=as55ec459bTo: "Tom Akers" ;tag=b864d414bbf47a88Contact: Call-ID: 05271c87d29c4c08@10.200.26.107CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 707 Reliably Transmitting (NAT) to 10.200.26.106:5060: NOTIFY sip:706@10.200.26.106:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1b7b105e;rportFrom: ;tag=as12414dc8To: "Jenine Bentley" ;tag=19cef75d507ad439Contact: Call-ID: ba019788181b78b6@10.200.26.106CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 706 Reliably Transmitting (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5a322275;rportFrom: ;tag=as30d10f77To: "Main Office Conf Room" ;tag=5b5475d3ba7e3992Contact: Call-ID: a2171917ddd9a517@10.200.26.105CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- <--- SIP read from 10.200.26.112:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6a091190;rportFrom: ;tag=as0b0c2072To: "Lorrie Blake" ;tag=d3947f943676b5a2Call-ID: 0d175327994ac226@10.200.26.112CSeq: 117 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3468afb7;rportFrom: ;tag=as3b8bd8b5To: "Shawn Norton" ;tag=d481dbf1f79a741fCall-ID: 0ca0cf97a319ed9e@10.200.26.110CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Extension Changed 201 new state Idle for Notify User 705 < asterisk-price*CLI> show channels --- SIP read from 10.200.26.109:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK284b98cd;rportFrom: ;tag=as24dbb862To: "Alan Young" ;tag=597133f1324431aeCall-ID: ace04268693825af@10.200.26.109CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.108:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK72b2ddf9;rportFrom: ;tag=as23529257To: "Jake Ori" ;tag=3b5457d35e7edd92Call-ID: ae07f2b6183b61b5@10.200.26.108CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Reliably Transmitting (NAT) to 10.200.26.104:5060: NOTIFY sip:704@10.200.26.104:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK15011482;rportFrom: ;tag=as6afb6310To: "Suzy Iorg" ;tag=06f2f98b43a41526Contact: Call-ID: 97b4dff5782a54a2@10.200.26.104CSeq: 117 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 704 asterisk-price*CLI> show channels Reliably Transmitting (NAT) to 10.200.26.103:5060: NOTIFY sip:703@10.200.26.103:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK461ac264;rportFrom: ;tag=as0f08e20bTo: "Shelli Marvidakis" ;tag=9984b794f0a6b218Contact: Call-ID: 0a5774e87eaceb46@10.200.26.103CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: active asterisk-price*CLI> show channels Content-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 703 <--- SIP read from 10.200.26.117:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK78c24c89;rportFrom: ;tag=as5ab9c51aTo: "Ken Williams" ;tag=85b710412fc42025Call-ID: d416754dbbee5bda@10.200.26.117CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.4.18Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.107:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK413f2bef;rportFrom: ;tag=as55ec459bTo: "Tom Akers" ;tag=b864d414bbf47a88Call-ID: 05271c87d29c4c08@10.200.26.107CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.106:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1b7b105e;rportFrom: ;tag=as12414dc8To: "Jenine Bentley" ;tag=19cef75d507ad439Call-ID: ba019788181b78b6@10.200.26.106CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Reliably Transmitting (NAT) to 10.200.26.102:5060: NOTIFY sip:702@10.200.26.102:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK317c61e2;rportFrom: ;tag=as414071ddTo: "Jason Worley" ;tag=5b84999412a69418Contact: Call-ID: 8a5755e830bc6ea6@10.200.26.102CSeq: 116 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 702 Reliably Transmitting (NAT) to 10.200.26.101:5060: NOTIFY sip:701@10.200.26.101:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK65b6d1ed;rportFrom: ;tag=as15c13c13To: "John Houston" ;tag=1d74fa5434d55668Contact: Call-ID: 88479078785cc3b6@10.200.26.101CSeq: 117 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 205 terminated --- <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5a322275;rportFrom: ;tag=as30d10f77To: "Main Office Conf Room" ;tag=5b5475d3ba7e3992Call-ID: a2171917ddd9a517@10.200.26.105CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> Extension Changed 201 new state Idle for Notify User 701 --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.104:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK15011482;rportFrom: ;tag=as6afb6310 asterisk-price*CLI> show channels To: "Suzy Iorg" ;tag=06f2f98b43a41526Call-ID: 97b4dff5782a54a2@10.200.26.104CSeq: 117 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.103:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK461ac264;rportFrom: ;tag=as0f08e20bTo: "Shelli Marvidakis" ;tag=9984b794f0a6b218Call-ID: 0a5774e87eaceb46@10.200.26.103CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.102:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK317c61e2;rportFrom: ;tag=as414071ddTo: "Jason Worley" ;tag=5b84999412a69418Call-ID: 8a5755e830bc6ea6@10.200.26.102CSeq: 116 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.101:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK65b6d1ed;rportFrom: ;tag=as15c13c13To: "John Houston" ;tag=1d74fa5434d55668Call-ID: 88479078785cc3b6@10.200.26.101CSeq: 117 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> show channels -- Called 705 -- SIP/705-097a4ca0 is ringing asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) asterisk-price*CLI> SIP/705-097a4ca0 SIP/705@from-interna Ringing AppDial((Outgoing Line)) asterisk-price*CLI> Zap/2-1 SIP/705@park-dial:1 Up Dial(SIP/705||t) asterisk-price*CLI> SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) asterisk-price*CLI> Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) asterisk-price*CLI> 4 active channels asterisk-price*CLI> 2 active calls asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK515776d2;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 102 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpSupported: replaces, timerContent-Length: 156v=0o=705 8000 8000 IN IP4 10.200.26.105s=SIP Callc=IN IP4 10.200.26.105t=0 0m=audio 5004 RTP/AVP 0a=sendrecva=rtpmap:0 PCMU/8000a=ptime:20 <-------------> asterisk-price*CLI> --- (12 headers 9 lines) --- asterisk-price*CLI> Found RTP audio format 0 asterisk-price*CLI> Peer audio RTP is at port 10.200.26.105:5004 asterisk-price*CLI> Found description format PCMU for ID 0 asterisk-price*CLI> Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) asterisk-price*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) asterisk-price*CLI> Peer audio RTP is at port 10.200.26.105:5004 asterisk-price*CLI> list_route: hop: asterisk-price*CLI> set_destination: Parsing for address/port to send to asterisk-price*CLI> set_destination: set destination to 10.200.26.105, port 5060 asterisk-price*CLI> Transmitting (NAT) to 10.200.26.105:5060: ACK sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0e11b67c;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Contact: Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- asterisk-price*CLI> -- SIP/705-097a4ca0 answered Zap/2-1 asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) asterisk-price*CLI> SIP/705-097a4ca0 (None) Up Bridged Call(Zap/2-1) asterisk-price*CLI> Zap/2-1 SIP/705@park-dial:1 Up Dial(SIP/705||t) asterisk-price*CLI> SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) asterisk-price*CLI> Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) asterisk-price*CLI> 4 active channels asterisk-price*CLI> 2 active calls asterisk-price*CLI> show channelsip shasterisk-price*CLI> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message asterisk-price*CLI> 10.200.26.105 705 3383a482252 00102/00000 ulaw No Tx: ACK asterisk-price*CLI> 10.200.26.117 717 4b560a236b5 00102/00000 ulaw No Tx: ACK asterisk-price*CLI> 2 active SIP channels asterisk-price*CLI> sip show channels<--- SIP read from 10.200.26.105:5060 ---> INVITE sip:8014525335@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK2aafc4eedb61be72From: ;tag=11e5563813de9b31To: "8014525335" ;tag=as5962a73dContact: Supported: replaces, timerCall-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 64408 INVITEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 280v=0o=705 8000 8001 IN IP4 10.200.26.105s=SIP Callc=IN IP4 10.200.26.105t=0 0m=audio 5004 RTP/AVP 0 8 4 18 2 3a=sendonlya=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:4 G723/8000a=rtpmap:18 G729/8000a=rtpmap:2 G726-32/8000a=rtpmap:3 GSM/8000a=ptime:20 <-------------> --- (13 headers 14 lines) --- Sending to 10.200.26.105 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 3 Peer audio RTP is at port 10.200.26.105:5004 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G723 for ID 4 Found description format G729 for ID 18 Found description format G726-32 for ID 2 Found description format GSM for ID 3 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.105:5004 Audio is at 10.200.26.202 port 10162 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK2aafc4eedb61be72;received=10.200.26.105From: ;tag=11e5563813de9b31To: "8014525335" ;tag=as5962a73dCall-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 64408 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Type: application/sdpContent-Length: 209v=0o=root 10024 10025 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10162 RTP/AVP 0 3a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=silenceSupp:off - - - -a=ptime:20a=recvonly <------------> -- Started music on hold, class 'default', on Zap/2-1 asterisk-price*CLI> [Jul 20 12:17:30] WARNING[11325]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> ACK sip:8014525335@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK92545996b6d8c871From: ;tag=11e5563813de9b31To: "8014525335" ;tag=as5962a73dContact: Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 64408 ACKUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> sip show channelsasterisk-price*CLI> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message asterisk-price*CLI> 10.200.26.105 705 3383a482252 00102/64408 ulaw Yes Rx: ACK asterisk-price*CLI> 10.200.26.117 717 4b560a236b5 00102/00000 ulaw No Tx: ACK asterisk-price*CLI> 2 active SIP channels asterisk-price*CLI> sip show channelsh<--- SIP read from 10.200.26.105:5060 ---> REFER sip:8014525335@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKa6b80ff4fcdfe5e9From: ;tag=11e5563813de9b31To: "8014525335" ;tag=as5962a73dContact: Supported: replacesRefer-To: Referred-By: Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 64409 REFERUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (14 headers 0 lines) --- Call 3383a482252267b15ca04f1b06273221@66.111.122.20 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 8710@from-internal by 705@10.200.26.202 <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 202 AcceptedVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bKa6b80ff4fcdfe5e9;received=10.200.26.105From: ;tag=11e5563813de9b31To: "8014525335" ;tag=as5962a73dCall-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 64409 REFERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.105, port 5060 Reliably Transmitting (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31c555d4;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Contact: Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=64409Subscription-state: activeContent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 21SIP/2.0 183 Ringing --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.105, port 5060 Reliably Transmitting (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK436a594a;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Contact: Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 104 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=64409Subscription-state: terminated;reason=noresourceContent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 16SIP/2.0 200 Ok --- -- Stopped music on hold on Zap/2-1 Scheduling destruction of SIP dialog '3383a482252267b15ca04f1b06273221@66.111.122.20' in 32000 ms (Method: REFER) == Spawn extension (from-internal, 8710, 0) exited non-zero on 'Zap/2-1' -- Executing [8710@from-internal:1] VoiceMail("Zap/2-1", "710|u") in new stack <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31c555d4;rport asterisk-price*CLI> show channels From: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- -- Playing '/var/spool/asterisk/voicemail/default/710/unavail' (language 'en') asterisk-price*CLI> show channels<--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK436a594a;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 104 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) asterisk-price*CLI> Zap/2-1 8710@from-internal:1 Up VoiceMail(710|u) asterisk-price*CLI> SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) asterisk-price*CLI> Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) asterisk-price*CLI> 3 active channels asterisk-price*CLI> 2 active calls asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> BYE sip:8014525335@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK1861ef45fc195e90From: ;tag=11e5563813de9b31To: "8014525335" ;tag=as5962a73dCall-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 64410 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.200.26.105 : 5060 (NAT) Scheduling destruction of SIP dialog '3383a482252267b15ca04f1b06273221@66.111.122.20' in 32000 ms (Method: BYE) <--- Transmitting (NAT) to 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.105:5060;branch=z9hG4bK1861ef45fc195e90;received=10.200.26.105From: ;tag=11e5563813de9b31To: "8014525335" ;tag=as5962a73dCall-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 64410 BYEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Length: 0 <------------> asterisk-price*CLI> show channels Retransmitting #1 (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31c555d4;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Contact: Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=64409Subscription-state: activeContent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 21SIP/2.0 183 Ringing --- asterisk-price*CLI> show channels<--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31c555d4;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> show channelsasterisk-price*CLI> Channel Location State Application(Data) Zap/2-1 8710@from-internal:1 Up VoiceMail(710|u) SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) 3 active channels 2 active calls asterisk-price*CLI> Retransmitting #2 (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31c555d4;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Contact: Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=64409Subscription-state: activeContent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 21SIP/2.0 183 Ringing --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31c555d4;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> --- (11 headers 0 lines) --- asterisk-price*CLI> Retransmitting #3 (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31c555d4;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Contact: Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=64409Subscription-state: activeContent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 21SIP/2.0 183 Ringing --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31c555d4;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> Retransmitting #4 (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31c555d4;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Contact: Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=64409Subscription-state: activeContent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 21SIP/2.0 183 Ringing --- asterisk-price*CLI> <--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31c555d4;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> show channels Retransmitting #5 (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31c555d4;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Contact: Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=64409Subscription-state: activeContent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 21SIP/2.0 183 Ringing --- asterisk-price*CLI> show channels<--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31c555d4;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> asterisk-price*CLI> show channels --- (11 headers 0 lines) --- asterisk-price*CLI> show channels Retransmitting #6 (NAT) to 10.200.26.105:5060: NOTIFY sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31c555d4;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Contact: Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: refer;id=64409Subscription-state: activeContent-Type: message/sipfrag;version=2.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 21SIP/2.0 183 Ringing --- asterisk-price*CLI> show channels<--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK31c555d4;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 103 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> show channels -- Playing 'vm-intro' (language 'en') asterisk-price*CLI> show channels [Jul 20 12:17:53] WARNING[10061]: chan_sip.c:1920 retrans_pkt: Maximum retries exceeded on transmission 3383a482252267b15ca04f1b06273221@66.111.122.20 for seqno 103 (Non-critical Request) asterisk-price*CLI> show channels -- Playing 'beep' (language 'en') asterisk-price*CLI> show channels -- Recording the message asterisk-price*CLI> show channels -- x=0, open writing: /var/spool/asterisk/voicemail/default/710/tmp/y3oE7N format: wav49, 0x9703840 -- x=1, open writing: /var/spool/asterisk/voicemail/default/710/tmp/y3oE7N format: gsm, 0x9846d98 -- x=2, open writing: /var/spool/asterisk/voicemail/default/710/tmp/y3oE7N format: wav, 0x979a150 asterisk-price*CLI> show channels set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.105, port 5060 Reliably Transmitting (NAT) to 10.200.26.105:5060: BYE sip:705@10.200.26.105:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0153a441;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 105 BYEUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --- Scheduling destruction of SIP dialog '3383a482252267b15ca04f1b06273221@66.111.122.20' in 32000 ms (Method: BYE) asterisk-price*CLI> show channels<--- SIP read from 10.200.26.105:5060 ---> SIP/2.0 481 No Such CallVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0153a441;rportFrom: "8014525335" ;tag=as5962a73dTo: ;tag=11e5563813de9b31Call-ID: 3383a482252267b15ca04f1b06273221@66.111.122.20CSeq: 105 BYEUser-Agent: Grandstream GXP2000 1.1.1.14Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived asterisk-price*CLI> show channels Really destroying SIP dialog '3383a482252267b15ca04f1b06273221@66.111.122.20' Method: BYE asterisk-price*CLI> show channels -- Recording automatically stopped after a silence of 5 seconds asterisk-price*CLI> show channels -- Playing 'auth-thankyou' (language 'en') asterisk-price*CLI> show channels -- Playing 'vm-review' (language 'en') asterisk-price*CLI> show channels [Jul 20 12:18:30] WARNING[11325]: file.c:626 ast_readaudio_callback: Failed to write frame -- Playing 'vm-goodbye' (language 'en') == Spawn extension (from-internal, 8710, 1) exited non-zero on 'Zap/2-1' -- Executing [h@from-internal:1] Hangup("Zap/2-1", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' asterisk-price*CLI> show channels Scheduling destruction of SIP dialog '757812c36e733cac47d5e3e94c5ec71e@66.111.122.20' in 32000 ms (Method: NOTIFY) Reliably Transmitting (NAT) to 10.200.26.110:5060: NOTIFY sip:710@10.200.26.110:5060 SIP/2.0Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK45d5a00b;rportFrom: "Unknown Caller" ;tag=as06ae27a6To: Contact: Call-ID: 757812c36e733cac47d5e3e94c5ec71e@66.111.122.20CSeq: 102 NOTIFYUser-Agent: Asterisk PBXMax-Forwards: 70Event: message-summaryContent-Type: application/simple-message-summaryContent-Length: 94Messages-Waiting: yesMessage-Account: sip:asterisk@66.111.122.20Voice-Message: 1/1 (0/0) --- asterisk-price*CLI> show channels<--- SIP read from 10.200.26.110:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK45d5a00b;rportFrom: "Unknown Caller" ;tag=as06ae27a6To: ;tag=f80a3f0febd96cd2Call-ID: 757812c36e733cac47d5e3e94c5ec71e@66.111.122.20CSeq: 102 NOTIFYUser-Agent: Grandstream GXP2000 1.1.1.14Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGESupported: replaces, timerContent-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> show channels Really destroying SIP dialog '757812c36e733cac47d5e3e94c5ec71e@66.111.122.20' Method: NOTIFY asterisk-price*CLI> show channelsip shhasterisk-price*CLI> Channel Location State Application(Data) asterisk-price*CLI> SIP/717-09731328 (None) Up Bridged Call(Zap/3-1) asterisk-price*CLI> Zap/3-1 SIP/717@park-dial:1 Up Dial(SIP/717||t) asterisk-price*CLI> 2 active channels asterisk-price*CLI> 1 active call asterisk-price*CLI> show channelsip shasterisk-price*CLI> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message asterisk-price*CLI> 10.200.26.117 717 4b560a236b5 00102/00000 ulaw No Tx: ACK asterisk-price*CLI> 1 active SIP channel asterisk-price*CLI> <--- SIP read from 10.200.26.117:5060 ---> INVITE sip:8019187318@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.117:5060;branch=z9hG4bKadece82c28b48ae4From: ;tag=9dcc5ef335a53591To: "WIRELESS CALLER" ;tag=as4ec3b36bContact: Supported: replaces, timer, pathCall-ID: 4b560a236b52eb1b768e3c3a48222ddf@66.111.122.20CSeq: 44763 INVITEUser-Agent: Grandstream GXP2000 1.1.4.18Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Type: application/sdpContent-Length: 253v=0o=717 8000 8001 IN IP4 10.200.26.117s=SIP Callc=IN IP4 10.200.26.117t=0 0m=audio 5004 RTP/AVP 0 8 4 18 3a=sendonlya=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:4 G723/8000a=rtpmap:18 G729/8000a=rtpmap:3 GSM/8000a=ptime:20 <-------------> --- (13 headers 13 lines) --- Sending to 10.200.26.117 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 3 Peer audio RTP is at port 10.200.26.117:5004 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G723 for ID 4 Found description format G729 for ID 18 Found description format GSM for ID 3 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.117:5004 Audio is at 10.200.26.202 port 10876 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP <--- Transmitting (NAT) to 10.200.26.117:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 10.200.26.117:5060;branch=z9hG4bKadece82c28b48ae4;received=10.200.26.117From: ;tag=9dcc5ef335a53591To: "WIRELESS CALLER" ;tag=as4ec3b36bCall-ID: 4b560a236b52eb1b768e3c3a48222ddf@66.111.122.20CSeq: 44763 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: Content-Type: application/sdpContent-Length: 209v=0o=root 10024 10025 IN IP4 10.200.26.202s=sessionc=IN IP4 10.200.26.202t=0 0m=audio 10876 RTP/AVP 0 3a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=silenceSupp:off - - - -a=ptime:20a=recvonly <------------> -- Started music on hold, class 'default', on Zap/3-1 asterisk-price*CLI> <--- SIP read from 10.200.26.117:5060 ---> ACK sip:8019187318@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.117:5060;branch=z9hG4bKe968223b443b2038From: ;tag=9dcc5ef335a53591To: "WIRELESS CALLER" ;tag=as4ec3b36bContact: Supported: pathCall-ID: 4b560a236b52eb1b768e3c3a48222ddf@66.111.122.20CSeq: 44763 ACKUser-Agent: Grandstream GXP2000 1.1.4.18Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (12 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.117:5060 ---> REFER sip:8019187318@10.200.26.202 SIP/2.0Via: SIP/2.0/UDP 10.200.26.117:5060;branch=z9hG4bK79e7f1c732183804From: ;tag=9dcc5ef335a53591To: "WIRELESS CALLER" ;tag=as4ec3b36bContact: Supported: replaces, pathRefer-To: Referred-By: Call-ID: 4b560a236b52eb1b768e3c3a48222ddf@66.111.122.20CSeq: 44764 REFERUser-Agent: Grandstream GXP2000 1.1.4.18Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGEContent-Length: 0 <-------------> --- (14 headers 0 lines) --- Call 4b560a236b52eb1b768e3c3a48222ddf@66.111.122.20 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 200@from-internal by 717@10.200.26.202 asterisk-price*CLI> quitsip show channelsasterisk-price*CLI> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message asterisk-price*CLI> 10.200.26.117 717 4b560a236b5 00102/44764 ulaw Yes Rx: REFER Done asterisk-price*CLI> 1 active SIP channel asterisk-price*CLI> sip show history 104b560a236b52eb1b768e3c3a48222ddf@66.111.122 .20 asterisk-price*CLI> * SIP Call 1. NewChan Channel SIP/717-09731328 - from 4b560a236b52eb1b768e3c3a48222dd 2. TxReqRel INVITE / 102 INVITE - -UNKNOWN- 3. Rx SIP/2.0 / 102 INVITE / 100 Trying 4. Rx SIP/2.0 / 102 INVITE / 180 Ringing 5. Rx SIP/2.0 / 102 INVITE / 200 OK 6. TxReq ACK / 102 ACK - -UNKNOWN- 7. Rx INVITE / 44763 INVITE / sip:8019187318@10.200.26.202 8. Hold INVITE 9. ReInv Re-invite received 10. TxResp SIP/2.0 / 44763 INVITE - 200 OK 11. Rx ACK / 44763 ACK / sip:8019187318@10.200.26.202 12. Rx REFER / 44764 REFER / sip:8019187318@10.200.26.202 13. Xfer REFER to call parking. asterisk-price*CLI> show chwannelsasterisk-price*CLI> Channel Location State Application(Data) asterisk-price*CLI> SIPPeer/SIP/717-0973 s@from-internal:1 Down (None) asterisk-price*CLI> Parking/Zap/3-1 SIP/717@park-dial:1 Down (None) asterisk-price*CLI> SIPPeer/SIP/717-0973 s@from-internal:1 Down (None) asterisk-price*CLI> Parking/Zap/3-1 SIP/717@park-dial:1 Down (None) asterisk-price*CLI> SIPPeer/SIP/717-0973 s@from-internal:1 Down (None) asterisk-price*CLI> Parking/Zap/3-1 SIP/717@park-dial:1 Down (None) asterisk-price*CLI> SIPPeer/SIP/717-0973 s@from-internal:1 Down (None) asterisk-price*CLI> Parking/Zap/3-1 SIP/717@park-dial:1 Down (None) asterisk-price*CLI> SIPPeer/SIP/717-0973 s@from-internal:1 Down (None) asterisk-price*CLI> Parking/Zap/3-1 SIP/717@park-dial:1 Down (None) asterisk-price*CLI> SIPPeer/SIP/717-0973 s@from-internal:1 Down (None) asterisk-price*CLI> Parking/Zap/3-1 SIP/717@park-dial:1 Down (None) asterisk-price*CLI> SIPPeer/SIP/717-0973 s@from-internal:1 Down (None) asterisk-price*CLI> Parking/Zap/3-1 SIP/717@park-dial:1 Down (None) asterisk-price*CLI> SIPPeer/SIP/717-0973 s@from-internal:1 Down (None) asterisk-price*CLI> Parking/Zap/3-1 SIP/717@park-dial:1 Down (None) asterisk-price*CLI> SIPPeer/SIP/717-0973 s@from-internal:1 Down (None) asterisk-price*CLI> Parking/Zap/3-1 SIP/717@park-dial:1 Down (None) asterisk-price*CLI> SIPPeer/SIP/717-0973 s@from-internal:1 Down (None) asterisk-price*CLI> Parking/Zap/3-1 SIP/717@park-dial:1 Down (None) asterisk-price*CLI> SIPPeer/SIP/717-0973 s@from-internal:1 Down (None) asterisk-price*CLI> Parking/Zap/3-1 SIP/717@park-dial:1 Down (None) asterisk-price*CLI> SIPPeer/SIP/717-0973 s@from-internal:1 Down (None) asterisk-price*CLI> Parking/Zap/3-1 SIP/717@park-dial:1 Down (None) asterisk-price*CLI> SIPPeer/SIP/717-0973 s@from-internal:1 Down (None) asterisk-price*CLI> Parking/Zap/3-1 SIP/717@park-dial:1 Down (None) asterisk-price*CLI> SIPPeer/SIP/717-0973 s@from-internal:1 Down (None) asterisk-price*CLI> Parking/Zap/3-1 SIP/717@park-dial:1 Down (None) asterisk-price*CLI> SIPPeer/SIP/717-0973 s@from-internal:1 Down (None) asterisk-price*CLI> Parking/Zap/3-1 SIP/717@park-dial:1 Down (None) asterisk-price*CLI> SIPPeer/SIP/717-0973 s@from-internal:1 Down (None) ****** asterisk-price*CLI> exit