--- Extension Changed 201 new state Idle for Notify User 709 Reliably Transmitting (no NAT) to 10.200.26.108:5060: NOTIFY sip:708@10.200.26.108:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5ff73a18;rport From: ;tag=as0768f506 To: "Jake Ori" ;tag=be0477e4d9de1ce8 Contact: Call-ID: c4e6add429ec84c9@10.200.26.108 CSeq: 341 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 206

terminated

--- Extension Changed 201 new state Idle for Notify User 708 Reliably Transmitting (no NAT) to 10.200.26.106:5060: NOTIFY sip:706@10.200.26.106:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK14239094;rport From: ;tag=as20c3ac97 To: "Jenine Bentley" ;tag=baeede5cd711c58d Contact: Call-ID: 18e077a378e2795d@10.200.26.106 CSeq: 341 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 206

terminated

--- Extension Changed 201 new state Idle for Notify User 706 Reliably Transmitting (no NAT) to 10.200.26.102:5060: NOTIFY sip:702@10.200.26.102:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK432b4686;rport From: ;tag=as0484f6c5 To: "Jason Worley" ;tag=b5f39cee3bbe3988 Contact: Call-ID: 48b6af0a6a3a05d5@10.200.26.102 CSeq: 339 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 206

terminated

--- Extension Changed 201 new state Idle for Notify User 702 Reliably Transmitting (no NAT) to 10.200.26.103:5060: NOTIFY sip:703@10.200.26.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK239d6035;rport From: ;tag=as667459b9 To: "Shelli Marvidakis" ;tag=b57492549bc5dd38 Contact: Call-ID: 6347f968bfeba8a7@10.200.26.103 CSeq: 342 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 206

terminated

--- Extension Changed 201 new state Idle for Notify User 703 --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK07914444;rport From: ;tag=as5ca73217 To: "Front Desk" ;tag=d1545ab5bb2e7e82 Call-ID: 40c6a37a29dcaf29@10.200.26.125 CSeq: 340 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-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 OK*CLI> Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK70c45329;rport From: ;tag=as13895db2 To: "Bobby Houston" ;tag=d8b3b38d4795aaa2 Call-ID: 26b5fff87c5c0a65@10.200.26.124 CSeq: 341 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-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 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4d71ba7d;rport From: ;tag=as13708049 To: "Danny D'Ambrosio" ;tag=e6df33b968f87d73 Call-ID: f93f2591a365d26e@10.200.26.121 CSeq: 339 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-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 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK32475fd9;rport From: ;tag=as2735cb1b To: "Darek Martinez" ;tag=aab3a58d09955ca2 Call-ID: 68b5c402ef192ee3@10.200.26.120 CSeq: 341 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-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 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK68200351;rport From: ;tag=as57b2832d To: "Randy Leffler" ;tag=5b1451f430eedfd8 Call-ID: a9a6ce99678d21a8@10.200.26.119 CSeq: 341 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-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 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2bb7b173;rport From: ;tag=as6b565f3e To: "Ed Hickman" ;tag=d93452755336b4c2 Call-ID: 8dd6a5fac0ad2c38@10.200.26.118 CSeq: 340 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-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 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK77ca5621;rport From: ;tag=as5071efdf To: "Shop Kitchen" ;tag=39241f255e25fe82 Call-ID: 4fb6661ac0aaeb18@10.200.26.116 CSeq: 341 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-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 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1bce4132;rport From: ;tag=as37255bb0 To: "Ken Williams" ;tag=93445ea5122e16b2 Call-ID: 62f6c9cb249c2f49@10.200.26.117 CSeq: 342 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-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 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6eac6e9c;rport From: ;tag=as1e0c5551 To: "Jerry Wright" ;tag=7924b135f1653213 Call-ID: c0c6677a21eccb99@10.200.26.115 CSeq: 342 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK77797a6f;rport From: ;tag=as27c650d0 To: "Jamie Stofko" ;tag=f7a4d12e93557513 Call-ID: e4f66dcb499c4659@10.200.26.131 CSeq: 340 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-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 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2309d396;rport From: ;tag=as128448a2 To: "Front Shop" ;tag=39a172d9b124b16a Call-ID: 10f0f4c8a0e01520@10.200.26.113 CSeq: 277 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-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 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK37e0c0b8;rport From: ;tag=as7143f3b1 To: "Shawn Norton" ;tag=f0b13b0adaad3b8c Call-ID: bbf090d8cc415b76@10.200.26.110 CSeq: 290 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-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 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK72d3d670;rport From: ;tag=as56d51660 To: "Alan Young" ;tag=f081fd22ffedb9c2 Call-ID: 3df0f1d8cc413a76@10.200.26.109 CSeq: 269 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-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 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5ff73a18;rport From: ;tag=as0768f506 To: "Jake Ori" ;tag=be0477e4d9de1ce8 Call-ID: c4e6add429ec84c9@10.200.26.108 CSeq: 341 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 10.200.26.106:5060 ---> IP/2.0 200 OKe*CLI> Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK14239094;rport From: ;tag=as20c3ac97 To: "Jenine Bentley" ;tag=baeede5cd711c58d Call-ID: 18e077a378e2795d@10.200.26.106 CSeq: 341 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-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 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK432b4686;rport From: ;tag=as0484f6c5 To: "Jason Worley" ;tag=b5f39cee3bbe3988 Call-ID: 48b6af0a6a3a05d5@10.200.26.102 CSeq: 339 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-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 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK239d6035;rport From: ;tag=as667459b9 To: "Shelli Marvidakis" ;tag=b57492549bc5dd38 Call-ID: 6347f968bfeba8a7@10.200.26.103 CSeq: 342 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK506fe804;rport From: "CODALE ELECTRIC" ;tag=as7bb384c3 To: ;tag=a451f806154d10a8 Call-ID: 2bbac31a7b7235df3d67faf1158f0170@10.200.26.202 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 v=0 o=731 8000 8000 IN IP4 10.200.26.131 s=SIP Call c=IN IP4 10.200.26.131 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (12 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.200.26.131:5004 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 10.200.26.131:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.131, port 5060 Transmitting (no NAT) to 10.200.26.131:5060: ACK sip:731@10.200.26.131:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK622a4f66;rport From: "CODALE ELECTRIC" ;tag=as7bb384c3 To: ;tag=a451f806154d10a8 Contact: Call-ID: 2bbac31a7b7235df3d67faf1158f0170@10.200.26.202 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Call on SIP/731-08325720 left from hold -- SIP/731-08325720 answered Zap/3-1 asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> INVITE sip:4356362900@10.200.26.202 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.131:5060;branch=z9hG4bKa4d72a9dd8e41ebd From: ;tag=a451f806154d10a8 To: "CODALE ELECTRIC" ;tag=as7bb384c3 Contact: Supported: replaces, timer Call-ID: 2bbac31a7b7235df3d67faf1158f0170@10.200.26.202 CSeq: 21363 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 309 v=0 o=731 8000 8001 IN IP4 10.200.26.131 s=SIP Call c=IN IP4 10.200.26.131 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 3 101 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (13 headers 15 lines) --- Sending to 10.200.26.131 : 5060 (no 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 Found RTP audio format 101 Peer audio RTP is at port 10.200.26.131:5004 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 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer 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 - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.200.26.131:5004 Audio is at 10.200.26.202 port 17606 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.200.26.131:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.26.131:5060;branch=z9hG4bKa4d72a9dd8e41ebd;received=10.200.26.131 From: ;tag=a451f806154d10a8 To: "CODALE ELECTRIC" ;tag=as7bb384c3 Call-ID: 2bbac31a7b7235df3d67faf1158f0170@10.200.26.202 CSeq: 21363 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10914 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 17606 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> -- Started music on hold, class 'default', on Zap/3-1 asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> ACK sip:4356362900@10.200.26.202 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.131:5060;branch=z9hG4bK032da6905c4f42cd From: ;tag=a451f806154d10a8 To: "CODALE ELECTRIC" ;tag=as7bb384c3 Contact: Call-ID: 2bbac31a7b7235df3d67faf1158f0170@10.200.26.202 CSeq: 21363 ACK User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> REFER sip:4356362900@10.200.26.202 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.131:5060;branch=z9hG4bKeabb758bc4cd3e1a From: ;tag=a451f806154d10a8 To: "CODALE ELECTRIC" ;tag=as7bb384c3 Contact: Supported: replaces Refer-To: Referred-By: Call-ID: 2bbac31a7b7235df3d67faf1158f0170@10.200.26.202 CSeq: 21364 REFER User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Call 2bbac31a7b7235df3d67faf1158f0170@10.200.26.202 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 200@from-internal by 731@10.200.26.202 -- Hungup 'Zap/2-1' == Spawn extension (from-internal, 201, 2) exited non-zero on 'SIP/719-b66ae0f0' -- Executing [h@from-internal:1] Hangup("SIP/719-b66ae0f0", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/719-b66ae0f0' Scheduling destruction of SIP dialog '4fc77bdadb11f6d6@10.200.26.119' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 10.200.26.119, port 5060 Reliably Transmitting (no NAT) to 10.200.26.119:5060: BYE sip:719@10.200.26.119:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3e1fa8a7;rport From: ;tag=as19cd4e01 To: "Randy Leffler" ;tag=656d6624e6a0345d Call-ID: 4fc77bdadb11f6d6@10.200.26.119 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk-price*CLI> quit [root@asterisk-price asterisk]# rasterisk Asterisk 1.4.4, Copyright (C) 1999 - 2006 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 1.4.4 currently running on asterisk-price (pid = 10913) Verbosity is at least 4 Core debug is at least 4 asterisk-price*CLI> sip show history Usage: sip show history Provides detailed dialog history on a given SIP channel. asterisk-price*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 10.200.26.131 731 2bbac31a7b7 00102/21364 ulaw Yes Rx: REFER Done 10.200.26.119 719 4fc77bdadb1 00102/18402 unkn No Tx: BYE 2 active SIP channels -- Starting simple switch on 'Zap/2-1' -- Executing [s@from-pstn:1] Answer("Zap/2-1", "") in new stack -- Executing [s@from-pstn:2] Goto("Zap/2-1", "790|1") in new stack -- Goto (from-pstn,790,1) -- Executing [790@from-pstn:1] Dial("Zap/2-1", "SIP/701&SIP/703&SIP/704&SIP/705&SIP/706&SIP/707&SIP/708&SIP/709&SIP/712&SIP/715&SIP/716&SIP/717&SIP/718&SIP/720&SIP/721&SIP/724&SIP/725&SIP/726&SIP/727&SIP/728&SIP/729&SIP/730&SIP/731|20|tr") in new stack Audio is at 10.200.26.202 port 24574 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.101:5060: INVITE sip:701@10.200.26.101:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1ff217c4;rport From: "AGAPITO ASOC IN" ;tag=as24aa5261 To: Contact: Call-ID: 128d35131c26129c5c390e1f4f347da4@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 24574 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 701 Audio is at 10.200.26.202 port 11968 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.103:5060: INVITE sip:703@10.200.26.103:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6970b2aa;rport From: "AGAPITO ASOC IN" ;tag=as52d99935 To: Contact: Call-ID: 75cddfc80b5ce1382df168b9456d2b77@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 11968 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 703 Audio is at 10.200.26.202 port 22598 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.104:5060: INVITE sip:704@10.200.26.104:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK20e42cdc;rport From: "AGAPITO ASOC IN" ;tag=as74e8cc7a To: Contact: Call-ID: 46470e5c280b8465634a08e41604975e@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 22598 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 704 Audio is at 10.200.26.202 port 27026 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.105:5060: INVITE sip:705@10.200.26.105:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK663fb03c;rport From: "AGAPITO ASOC IN" ;tag=as3d5ec784 To: Contact: Call-ID: 58bec537066553833a2b7a281aa4cb42@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 27026 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 705 Audio is at 10.200.26.202 port 15540 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.106:5060: INVITE sip:706@10.200.26.106:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3d1b007c;rport From: "AGAPITO ASOC IN" ;tag=as26847141 To: Contact: Call-ID: 5e8b8fe42565002736303fdd6701f856@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 15540 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 706I> sip Audio is at 10.200.26.202 port 13374 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.107:5060: INVITE sip:707@10.200.26.107:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4ce2bb5e;rport From: "AGAPITO ASOC IN" ;tag=as540c1c09 To: Contact: Call-ID: 5ddfb5f01cf8328d1b6f5595302ff805@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 13374 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ---erisk-price*CLI> sip -- Called 707 Audio is at 10.200.26.202 port 10128 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.108:5060: INVITE sip:708@10.200.26.108:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK417fcdbe;rport From: "AGAPITO ASOC IN" ;tag=as36c277b0 To: Contact: Call-ID: 6fbcefa17cd69f1c1f5d9acc485391af@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10128 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ---erisk-price*CLI> sip -- Called 708I> sip Audio is at 10.200.26.202 port 9544 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.109:5060: INVITE sip:709@10.200.26.109:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK10d43eab;rport From: "AGAPITO ASOC IN" ;tag=as0389501a To: Contact: Call-ID: 386e7b661e48c9dd7adae8b8071e8850@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 264 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 9544 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 709 Audio is at 10.200.26.202 port 17972 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.112:5060: INVITE sip:712@10.200.26.112:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1524bb0a;rport From: "AGAPITO ASOC IN" ;tag=as02c160aa To: Contact: Call-ID: 09f9819541fe934f1097383b3dc4f45c@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 17972 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 712I> sip Audio is at 10.200.26.202 port 27400 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.115:5060: INVITE sip:715@10.200.26.115:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK131c6568;rport From: "AGAPITO ASOC IN" ;tag=as1a23a487 To: Contact: Call-ID: 4c9bfc4c4f5b60052e53d1dc5c80c158@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 27400 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ---erisk-price*CLI> sip -- Called 715 Audio is at 10.200.26.202 port 13244 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.116:5060: INVITE sip:716@10.200.26.116:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK68f3ca47;rport From: "AGAPITO ASOC IN" ;tag=as6ed7aca1 To: Contact: Call-ID: 5867289e1054614e763302b7217ecf98@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 13244 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 716I> sip Audio is at 10.200.26.202 port 21726 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.117:5060: INVITE sip:717@10.200.26.117:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK298916a8;rport From: "AGAPITO ASOC IN" ;tag=as52daf9ed To: Contact: Call-ID: 6883e4a85a5fc7842f85b3441ab03663@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 21726 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ---erisk-price*CLI> sip -- Called 717I> sip Audio is at 10.200.26.202 port 23926 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.118:5060: INVITE sip:718@10.200.26.118:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK76dc7ea5;rport From: "AGAPITO ASOC IN" ;tag=as048e4be3 To: Contact: Call-ID: 62728963445802c76068f654171eb523@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 23926 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 718I> sip Audio is at 10.200.26.202 port 23586 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.120:5060: INVITE sip:720@10.200.26.120:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6f5ac30a;rport From: "AGAPITO ASOC IN" ;tag=as0d75f93b To: Contact: Call-ID: 18c1448036d579140b19ed8546605a4c@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 23586 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 720I> sip Audio is at 10.200.26.202 port 15186 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.121:5060: INVITE sip:721@10.200.26.121:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5296965c;rport From: "AGAPITO ASOC IN" ;tag=as42c5cb0d To: Contact: Call-ID: 738060575d2ff576624ee02a179dc2e4@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 15186 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 721 Audio is at 10.200.26.202 port 30792 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.124:5060: INVITE sip:724@10.200.26.124:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7b675f08;rport From: "AGAPITO ASOC IN" ;tag=as2f40b41d To: Contact: Call-ID: 4c71482a0bdfd0d56aabf44c11cba972@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 30792 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ---erisk-price*CLI> sip -- Called 724 Audio is at 10.200.26.202 port 15072 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.125:5060: INVITE sip:725@10.200.26.125:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK111604c9;rport From: "AGAPITO ASOC IN" ;tag=as31ec7a48 To: Contact: Call-ID: 7217c25a4d5d5381237dcd6000bccf8f@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 15072 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 725I> sip Audio is at 10.200.26.202 port 29344 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.126:5060: INVITE sip:726@10.200.26.126:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK02a26939;rport From: "AGAPITO ASOC IN" ;tag=as0d52573a To: Contact: Call-ID: 5a5b762433c984431dceca9e45990555@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 29344 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ---erisk-price*CLI> sip -- Called 726I> sip Audio is at 10.200.26.202 port 27824 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.127:5060: INVITE sip:727@10.200.26.127:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK50a1a18b;rport From: "AGAPITO ASOC IN" ;tag=as2afe162f To: Contact: Call-ID: 45c65e91248385f031947f2c1b2639c0@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 27824 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 727 Audio is at 10.200.26.202 port 6458 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.128:5060: INVITE sip:728@10.200.26.128:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK70304d80;rport From: "AGAPITO ASOC IN" ;tag=as3907f67a To: Contact: Call-ID: 4c4ceca664492aaf72786b7022267bc9@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 264 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 6458 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 728 Audio is at 10.200.26.202 port 18740 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.129:5060: INVITE sip:729@10.200.26.129:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5c280104;rport From: "AGAPITO ASOC IN" ;tag=as22e67880 To: Contact: Call-ID: 4f53ba8d4423e1026879babc277fcad8@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 18740 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 729I> sip Audio is at 10.200.26.202 port 22766 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.130:5060: INVITE sip:730@10.200.26.130:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6df844a7;rport From: "AGAPITO ASOC IN" ;tag=as2accc31c To: Contact: Call-ID: 14229254483f75b95cc54f720869de4e@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 22766 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 730 Audio is at 10.200.26.202 port 10438 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.200.26.131:5060: INVITE sip:731@10.200.26.131:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK29b24c2f;rport From: "AGAPITO ASOC IN" ;tag=as10ab8a19 To: Contact: Call-ID: 00df65fc7b7439c00122fb094d32169f@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 14 May 2007 17:39:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 10913 10913 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10438 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 731 asterisk-price*CLI> reload chan_sip.so -- Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP)) asterisk-price*CLI> unload chan_sip.so Unable to unload resource chan_sip.so The 'unload' command is deprecated and will be removed in a future release. Please use 'module unload' instead. [May 14 11:40:42] WARNING[6314]: loader.c:458 ast_unload_resource: Soft unload failed, 'chan_sip.so' has use count 32 asterisk-price*CLI> unload chan_sip.so Unable to unload resource chan_sip.so [May 14 11:40:47] WARNING[6314]: loader.c:458 ast_unload_resource: Soft unload failed, 'chan_sip.so' has use count 32 asterisk-price*CLI> sip show inuse * User name In use Limit 739 0 9 731 0 9 730 0 9 729 0 9 728 0 9 727 0 9 726 0 9 725 0 9 724 0 9 721 0 9 720 0 9 719 0 9 718 0 9 717 0 9 716 0 9 715 0 9 713 0 9 712 0 9 710 0 9 709 0 9 708 0 9 707 0 9 706 0 9 705 0 9 704 0 9 703 0 9 702 0 9 701 0 9 * Peer name In use Limit 739 0/0 9 731 2/1 9 730 1/1 9 729 1/1 9 728 1/1 9 727 1/1 9 726 1/1 9 725 1/1 9 724 1/1 9 721 1/1 9 720 1/1 9 719 0/0 9 718 1/1 9 717 1/1 9 716 1/1 9 715 1/1 9 713 0/0 9 712 1/1 9 710 0/0 9 709 1/1 9 708 1/1 9 707 1/1 9 706 1/1 9 705 1/1 9 704 1/1 9 703 1/1 9 702 0/0 9 701 1/1 9 asterisk-price*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 10.200.26.131 731 00df65fc7b7 00102/00000 ulaw No Init: INVITE 10.200.26.130 730 14229254483 00102/00000 ulaw No Init: INVITE 10.200.26.129 729 4f53ba8d442 00102/00000 ulaw No Init: INVITE 10.200.26.128 728 4c4ceca6644 00102/00000 ulaw No Init: INVITE 10.200.26.127 727 45c65e91248 00102/00000 ulaw No Init: INVITE 10.200.26.126 726 5a5b762433c 00102/00000 ulaw No Init: INVITE 10.200.26.125 725 7217c25a4d5 00102/00000 ulaw No Init: INVITE 10.200.26.124 724 4c71482a0bd 00102/00000 ulaw No Init: INVITE 10.200.26.121 721 738060575d2 00102/00000 ulaw No Init: INVITE 10.200.26.120 720 18c1448036d 00102/00000 ulaw No Init: INVITE 10.200.26.118 718 62728963445 00102/00000 ulaw No Init: INVITE 10.200.26.117 717 6883e4a85a5 00102/00000 ulaw No Init: INVITE 10.200.26.116 716 5867289e105 00102/00000 ulaw No Init: INVITE 10.200.26.115 715 4c9bfc4c4f5 00102/00000 ulaw No Init: INVITE 10.200.26.112 712 09f9819541f 00102/00000 ulaw No Init: INVITE 10.200.26.109 709 386e7b661e4 00102/00000 ulaw No Init: INVITE 10.200.26.108 708 6fbcefa17cd 00102/00000 ulaw No Init: INVITE 10.200.26.107 707 5ddfb5f01cf 00102/00000 ulaw No Init: INVITE 10.200.26.106 706 5e8b8fe4256 00102/00000 ulaw No Init: INVITE 10.200.26.105 705 58bec537066 00102/00000 ulaw No Init: INVITE 10.200.26.104 704 46470e5c280 00102/00000 ulaw No Init: INVITE 10.200.26.103 703 75cddfc80b5 00102/00000 ulaw No Init: INVITE 10.200.26.101 701 128d35131c2 00102/00000 ulaw No Init: INVITE 10.200.26.131 731 2bbac31a7b7 00102/21364 ulaw Yes Rx: REFER Done 10.200.26.119 719 4fc77bdadb1 00102/18402 unkn No Tx: BYE 25 active SIP channels asterisk-price*CLI> sip show history 2bbac31a7b7 asterisk-price*CLI> * SIP Callce*CLI> 1. NewChan Channel SIP/731-08325720 - from 2bbac31a7b7235df3d67faf1158f017 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 / 21363 INVITE / sip:4356362900@10.200.26.202 8. Hold INVITE 9. ReInv Re-invite received 10. TxRespRel SIP/2.0 / 21363 INVITE - 200 OK 11. Rx ACK / 21363 ACK / sip:4356362900@10.200.26.202 12. Rx REFER / 21364 REFER / sip:4356362900@10.200.26.202 13. Xfer REFER to call parking. asterisk-price*CLI> sip show history 4fc77bdadb1 asterisk-price*CLI> * SIP Callce*CLI> 1. Rx INVITE / 18401 INVITE / sip:201@10.200.26.202 2. AuthChal Auth challenge sent for - nc 0 3. TxRespRel SIP/2.0 / 18401 INVITE - 407 Proxy Authentication Required 4. SchedDestroy 32000 ms 5. Rx ACK / 18401 ACK / sip:201@10.200.26.202 6. Rx INVITE / 18402 INVITE / sip:201@10.200.26.202 7. CancelDestroy 8. Invite New call: 4fc77bdadb11f6d6@10.200.26.119 9. AuthOK Auth challenge succesful for 719 10. NewChan Channel SIP/719-b66ae0f0 - from 4fc77bdadb11f6d6@10.200.26.119 11. TxResp SIP/2.0 / 18402 INVITE - 100 Trying 12. TxRespRel SIP/2.0 / 18402 INVITE - 200 OK 13. Rx ACK / 18402 ACK / sip:201@10.200.26.202 14. SchedDestroy 32000 ms 15. TxReqRel BYE / 102 BYE - -UNKNOWN- 16. RTCPaudio Quality:ssrc=323951117;themssrc=4099729936;lp=0;rxjitter=0.0000 asterisk-price*CLI>