--- -- SIP/731-086a6fc8 is ringing <--- SIP read from 10.200.26.111:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0986aba9;rport From: ;tag=as0ce83172 To: "Center Shop" ;tag=dc80b02254ef636e Call-ID: 238e87bc8ca85b7f@10.200.26.111 CSeq: 156 NOTIFY User-Agent: Grandstream GXP2000 1.1.2.23 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 10.200.26.130:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK6e83fdd5;rport From: ;tag=as50911967 To: "Foyer" ;tag=93947c863e56fff2 Call-ID: e5d60dea26b6ee88@10.200.26.130 CSeq: 159 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 10.200.26.129:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK63d1bae6;rport From: ;tag=as70b89961 To: "Will Jaimez" ;tag=5e54b9c55d6e1023 Call-ID: 42d6a7ea4f2d0785@10.200.26.129 CSeq: 158 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 10.200.26.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK503add2f;rport From: ;tag=as7784c871 To: "Upstairs Kitchen" ;tag=3864d106d3e41509 Call-ID: 49d6e0fa2c9d8839@10.200.26.128 CSeq: 159 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: asterisk-price*CLI> 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 10.200.26.127:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK53df4d78;rport From: ;tag=as14531158 To: "Conference Room" ;tag=5b7417469ab55db8 Call-ID: 0346041a00646d38@10.200.26.127 CSeq: 159 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 10.200.26.126:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK0c272aab;rport From: ;tag=as5c23ca42 To: "Bob Teny" ;tag=de64b806fbe4bf09 Call-ID: cae6816bcaac2548@10.200.26.126 CSeq: 159 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 10.200.26.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK34167a45;rport From: ;tag=as7eaa31fa To: "Front Desk" ;tag=6672e71402d2fb76 Call-ID: aa933b68de09d221@10.200.26.125 CSeq: 159 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 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1632b759;rport From: ;tag=as393a8b8d To: "Bobby Houston" ;tag=716499f5d9fe3a19 Call-ID: 00d608eae13d4ce5@10.200.26.124 CSeq: 159 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=z9hG4bK0b025241;rport From: ;tag=as3a824987 To: "Danny D'Ambrosio" ;tag=f85471c5726e5313 Call-ID: 04c6a97a42ec6c99@10.200.26.121 CSeq: 159 NOTIFY asterisk-price*CLI> 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 <-------------> Extension Changed 201 new state Idle for Notify User 716 --- (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=z9hG4bK2d1885bd;rport From: ;tag=as4754a142 To: "Darek Martinez" ;tag=31b3ab7d9e557122 Call-ID: f9a5c2982f6c4cb5@10.200.26.120 CSeq: 159 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=z9hG4bK283ef2ce;rport From: ;tag=as08f79e46 To: "Randy Leffler" ;tag=754430b5f35ef713 Call-ID: 85f62dcb879ca259@10.200.26.119 CSeq: 159 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=z9hG4bK0e9e9c3b;rport From: ;tag=as0d667105 To: "Ed Hickman" ;tag=90341765f6f5d519 Call-ID: 88c6a08a4c4da856@10.200.26.118 CSeq: 159 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 Reliably Transmitting (NAT) to 10.200.26.113:5060: NOTIFY sip:713@10.200.26.113:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK74d3d67d;rport From: ;tag=as2d050bd0 To: "Front Shop" ;tag=f8b1941af5dd7737 Contact: Call-ID: 19015f950c32f256@10.200.26.113 CSeq: 147 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- <--- SIP read from 10.200.26.116:5060 ---> SIP/2.0 200 OK asterisk-price*CLI> Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK7f8aa7f6;rport From: ;tag=as355fe848 To: "Shop Kitchen" ;tag=3b24d3351565f513 Call-ID: e2c6697ac366eef7@10.200.26.116 CSeq: 159 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) --- asterisk-price*CLI> 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=z9hG4bK79ab5cd6;rport From: ;tag=as45f2d1df To: "Jamie Stofko" ;tag=3194b9863a565af2 Call-ID: 0cc6438a0e4d0a46@10.200.26.131 CSeq: 159 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 Extension Changed 201 new state Idle for Notify User 713 Reliably Transmitting (NAT) to 10.200.26.110:5060: NOTIFY sip:710@10.200.26.110:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK2e52e5d5;rport From: ;tag=as2c0e012c To: "Shawn Norton" ;tag=1bc1d75a5a6edc0c Contact: Call-ID: 18217f76cd919487@10.200.26.110 CSeq: 147 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-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/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK1d0c7054;rport From: ;tag=as567b6b40 To: "Alan Young" ;tag=7381de223fed1f0e Contact: Call-ID: 1df0f2d8cf41dc76@10.200.26.109 CSeq: 147 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-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/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK05223b22;rport From: ;tag=as389d6a48 To: "Jake Ori" ;tag=7004f8d4f99efa73 Contact: Call-ID: 20d666eaee2de885@10.200.26.108 CSeq: 158 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 708 Reliably Transmitting (NAT) to 10.200.26.101:5060: NOTIFY sip:701@10.200.26.101:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK53a28483;rport From: ;tag=as719e083e To: "John Houston" ;tag=1764f314fbf43c88 Contact: Call-ID: 212777871e8c28a7@10.200.26.101 CSeq: 159 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- Extension Changed 201 new state Idle for Notify User 701 <--- SIP read from 10.200.26.113:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK74d3d67d;rport From: ;tag=as2d050bd0 To: "Front Shop" ;tag=f8b1941af5dd7737 Call-ID: 19015f950c32f256@10.200.26.113 CSeq: 147 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=z9hG4bK2e52e5d5;rport From: ;tag=as2c0e012c To: "Shawn Norton" ;tag=1bc1d75a5a6edc0c Call-ID: 18217f76cd919487@10.200.26.110 CSeq: 147 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=z9hG4bK1d0c7054;rport From: ;tag=as567b6b40 To: "Alan Young" ;tag=7381de223fed1f0e Call-ID: 1df0f2d8cf41dc76@10.200.26.109 CSeq: 147 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=z9hG4bK05223b22;rport From: ;tag=as389d6a48 To: "Jake Ori" ;tag=7004f8d4f99efa73 Call-ID: 20d666eaee2de885@10.200.26.108 CSeq: 158 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 10.200.26.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK53a28483;rport From: ;tag=as719e083e To: "John Houston" ;tag=1764f314fbf43c88 Call-ID: 212777871e8c28a7@10.200.26.101 CSeq: 159 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> Retransmitting #1 (NAT) to 12.152.163.196:1460: NOTIFY sip:734@192.168.2.25:5060 SIP/2.0 Via: SIP/2.0/UDP 66.111.122.20:5060;branch=z9hG4bK62c9d64c;rport From: ;tag=as00491aad To: "Ken Williams" ;tag=db2a99436c6a53c8 Contact: Call-ID: 9e1d884504858fe2@192.168.2.25 CSeq: 158 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- asterisk-price*CLI> Retransmitting #2 (NAT) to 12.152.163.196:1460: NOTIFY sip:734@192.168.2.25:5060 SIP/2.0 Via: SIP/2.0/UDP 66.111.122.20:5060;branch=z9hG4bK62c9d64c;rport From: ;tag=as00491aad To: "Ken Williams" ;tag=db2a99436c6a53c8 Contact: Call-ID: 9e1d884504858fe2@192.168.2.25 CSeq: 158 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- asterisk-price*CLI> Retransmitting #3 (NAT) to 12.152.163.196:1460: NOTIFY sip:734@192.168.2.25:5060 SIP/2.0 Via: SIP/2.0/UDP 66.111.122.20:5060;branch=z9hG4bK62c9d64c;rport From: ;tag=as00491aad To: "Ken Williams" ;tag=db2a99436c6a53c8 Contact: Call-ID: 9e1d884504858fe2@192.168.2.25 CSeq: 158 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- asterisk-price*CLI> Retransmitting #4 (NAT) to 12.152.163.196:1460: OPTIONS sip:734@192.168.2.25:5060 SIP/2.0 Via: SIP/2.0/UDP 66.111.122.20:5060;branch=z9hG4bK6e9ca907;rport From: "Unknown Caller" ;tag=as51aeb9b8 To: Contact: Call-ID: 22d9d10807e7b68c0e5e952e34d62658@66.111.122.20 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 14:59:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Really destroying SIP dialog '22d9d10807e7b68c0e5e952e34d62658@66.111.122.20' Method: OPTIONS asterisk-price*CLI> Retransmitting #4 (NAT) to 12.152.163.196:1460: NOTIFY sip:734@192.168.2.25:5060 SIP/2.0 Via: SIP/2.0/UDP 66.111.122.20:5060;branch=z9hG4bK62c9d64c;rport From: ;tag=as00491aad To: "Ken Williams" ;tag=db2a99436c6a53c8 Contact: Call-ID: 9e1d884504858fe2@192.168.2.25 CSeq: 158 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- asterisk-price*CLI> Retransmitting #5 (NAT) to 12.152.163.196:1460: NOTIFY sip:734@192.168.2.25:5060 SIP/2.0 Via: SIP/2.0/UDP 66.111.122.20:5060;branch=z9hG4bK62c9d64c;rport From: ;tag=as00491aad To: "Ken Williams" ;tag=db2a99436c6a53c8 Contact: Call-ID: 9e1d884504858fe2@192.168.2.25 CSeq: 158 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- 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=z9hG4bK075178ec;rport From: "EIS INC" ;tag=as22028614 To: ;tag=e23b44165df86673 Call-ID: 151dba617b76b64146da6ca06d1060f2@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: 156 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 a=sendrecv a=rtpmap:0 PCMU/8000 a=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 - 0x0 (nothing), 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/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4033d618;rport From: "EIS INC" ;tag=as22028614 To: ;tag=e23b44165df86673 Contact: Call-ID: 151dba617b76b64146da6ca06d1060f2@10.200.26.202 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/731-086a6fc8 answered Zap/7-1 asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> INVITE sip:6782553661@10.200.26.202 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.131:5060;branch=z9hG4bKdf6a36ac6be5e351 From: ;tag=e23b44165df86673 To: "EIS INC" ;tag=as22028614 Contact: Supported: replaces, timer Call-ID: 151dba617b76b64146da6ca06d1060f2@10.200.26.202 CSeq: 34243 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: 253 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 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 <-------------> --- (13 headers 13 lines) --- Sending to 10.200.26.131 : 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 10.200.26.131:5004 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 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 - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.200.26.131:5004 Audio is at 10.200.26.202 port 10226 Adding codec 0x4 (ulaw) to SDP asterisk-price*CLI> Adding codec 0x2 (gsm) to SDP <--- Reliably Transmitting (NAT) to 10.200.26.131:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.200.26.131:5060;branch=z9hG4bKdf6a36ac6be5e351;received=10.200.26.131 From: ;tag=e23b44165df86673 To: "EIS INC" ;tag=as22028614 Call-ID: 151dba617b76b64146da6ca06d1060f2@10.200.26.202 CSeq: 34243 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1381 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10226 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> asterisk-price*CLI> -- Started music on hold, class 'default', on Zap/7-1 asterisk-price*CLI> <--- SIP read from 10.200.26.131:5060 ---> ACK sip:6782553661@10.200.26.202 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.131:5060;branch=z9hG4bKe10723893a349621 From: ;tag=e23b44165df86673 To: "EIS INC" ;tag=as22028614 Contact: Call-ID: 151dba617b76b64146da6ca06d1060f2@10.200.26.202 CSeq: 34243 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:6782553661@10.200.26.202 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.131:5060;branch=z9hG4bK5946ca300c00ffec From: ;tag=e23b44165df86673 To: "EIS INC" ;tag=as22028614 Contact: Supported: replaces Refer-To: Referred-By: Call-ID: 151dba617b76b64146da6ca06d1060f2@10.200.26.202 CSeq: 34244 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 151dba617b76b64146da6ca06d1060f2@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 asterisk-price*CLI> -- Hungup 'Zap/3-1' == Spawn extension (from-internal, 201, 3) exited non-zero on 'SIP/706-08625f40' -- Executing [h@from-internal:1] Hangup("SIP/706-08625f40", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/706-08625f40' Scheduling destruction of SIP dialog '32e433cab3f588e6@10.200.26.106' in 32000 ms (Method: ACK) 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/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK3a39cfea;rport From: ;tag=as3f15e550 To: "Jenine Bentley" ;tag=019d58687157fc47 Call-ID: 32e433cab3f588e6@10.200.26.106 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk-price*CLI> -- Remote UNIX connection asterisk-price*CLI> -- Starting simple switch on 'Zap/2-1' asterisk-price*CLI> -- 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] ChanIsAvail("Zap/2-1", "SIP/710|s") in new stack asterisk-price*CLI> Scheduling destruction of SIP dialog '018455be524b9b12744e212605abc320@10.200.26.202' in 32000 ms (Method: INVITE) asterisk-price*CLI> -- Executing [790@from-pstn:2] NoOp("Zap/2-1", "Shawn is ") in new stack asterisk-price*CLI> -- Executing [790@from-pstn:3] GotoIf("Zap/2-1", "0?noshawn|1") in new stack -- Executing [790@from-pstn:4] ChanIsAvail("Zap/2-1", "SIP/702|s") in new stack Scheduling destruction of SIP dialog '4b28cf2a22bf0ce206f208ff70bdf04f@10.200.26.202' in 32000 ms (Method: INVITE) -- Executing [790@from-pstn:5] NoOp("Zap/2-1", "Jason is ") in new stack asterisk-price*CLI> -- Executing [790@from-pstn:6] GotoIf("Zap/2-1", "0?nojason|1") in new stack -- Executing [790@from-pstn:7] Dial("Zap/2-1", "SIP/701&SIP/702&SIP/703&SIP/704&SIP/705&SIP/706&SIP/707&SIP/708&SIP/709&SIP/710&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|tTK") in new stack Audio is at 10.200.26.202 port 10512 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK3a2bc9ea;rport From: "EIS INC" ;tag=as22141cb6 To: Contact: Call-ID: 10359b5323fd5fd84b1c287349edc3a4@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10512 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 701 asterisk-price*CLI> Audio is at 10.200.26.202 port 10546 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Reliably Transmitting (NAT) to 10.200.26.102:5060: INVITE sip:702@10.200.26.102:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK232fa4ae;rport From: "EIS INC" ;tag=as0da0f65f To: Contact: Call-ID: 5c76eb8309d40c6d528e696b3eae7568@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10546 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-price*CLI> -- Called 702 Audio is at 10.200.26.202 port 10418 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK499faece;rport From: "EIS INC" ;tag=as4b5f1017 To: Contact: Call-ID: 3f21b574034a3f6d1b496b9677e9167a@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10418 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-price*CLI> -- Called 703 Audio is at 10.200.26.202 port 10972 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK4e4746d4;rport From: "EIS INC" ;tag=as2c53ab43 To: Contact: Call-ID: 7ac1d80146b7c36613f677e341076ba9@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10972 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 704 asterisk-price*CLI> Audio is at 10.200.26.202 port 10070 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK178cb4c2;rport From: "EIS INC" ;tag=as7ab6498c To: Contact: Call-ID: 42fae1bf3cd7d6c25c37a5c71b3d06b1@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10070 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-price*CLI> -- Called 705 asterisk-price*CLI> Audio is at 10.200.26.202 port 10782 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/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK059759ae;rport From: "EIS INC" ;tag=as74d90593 To: Contact: Call-ID: 45927ae3599842d727f3e2e934f4249f@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 263 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10782 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 --- asterisk-price*CLI> -- Called 706 asterisk-price*CLI> Audio is at 10.200.26.202 port 10840 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK3647c15c;rport From: "EIS INC" ;tag=as4854ce14 To: Contact: Call-ID: 10c90ffb0c189fc526e9861b35a7b41d@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10840 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 707 asterisk-price*CLI> Audio is at 10.200.26.202 port 10326 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK004fc62d;rport From: "EIS INC" ;tag=as62153097 To: Contact: Call-ID: 3a90be5f7d2f9c474152cfed57c19ea5@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10326 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 708 asterisk-price*CLI> Audio is at 10.200.26.202 port 10316 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK465e058e;rport From: "EIS INC" ;tag=as28d63a88 To: Contact: Call-ID: 3dfbff5e61f3864c28aa1a1d7652b6ed@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10316 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-price*CLI> -- Called 709 asterisk-price*CLI> Audio is at 10.200.26.202 port 10176 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (NAT) to 10.200.26.110:5060: INVITE sip:710@10.200.26.110:5060 SIP/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK5d509bef;rport From: "EIS INC" ;tag=as5d28016c To: Contact: Call-ID: 526901127b7501d16f9d2b042db317db@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10176 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 710 asterisk-price*CLI> Audio is at 10.200.26.202 port 10044 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Reliably Transmitting (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=z9hG4bK7c7c4071;rport From: "EIS INC" ;tag=as03b51063 To: Contact: Call-ID: 710e6c6d1fae327220dc61e260b906ba@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10044 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-price*CLI> -- Called 712 Audio is at 10.200.26.202 port 10514 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Reliably Transmitting (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=z9hG4bK71e3958f;rport From: "EIS INC" ;tag=as3a72a2ec To: Contact: Call-ID: 693f60ce0dc086474d8b1756053ebee0@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10514 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 715 asterisk-price*CLI> Audio is at 10.200.26.202 port 10090 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP asterisk-price*CLI> Reliably Transmitting (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=z9hG4bK01160962;rport From: "EIS INC" ;tag=as3b2754f0 To: Contact: Call-ID: 2dd03b3f1820112f16665f0255b4187e@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10090 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-price*CLI> -- Called 716 Audio is at 10.200.26.202 port 10680 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/2.0 Via: SIP/2.0/UDP 10.200.26.202:5060;branch=z9hG4bK4e06cd39;rport From: "EIS INC" ;tag=as2c571441 To: Contact: Call-ID: 552e27fb3228e1e86b5b419d3fcaa803@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 263 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10680 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 --- asterisk-price*CLI> -- Called 717 asterisk-price*CLI> Audio is at 10.200.26.202 port 10132 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK1bae19c5;rport From: "EIS INC" ;tag=as2f0abc62 To: Contact: Call-ID: 0250c228253090467004414e6fcc16f6@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10132 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-price*CLI> -- Called 718 asterisk-price*CLI> Audio is at 10.200.26.202 port 10166 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK67349787;rport From: "EIS INC" ;tag=as4a4edd89 To: Contact: Call-ID: 0bff035a6353560761912ed01576bbb6@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10166 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-price*CLI> -- Called 720 Audio is at 10.200.26.202 port 10960 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK12bd48be;rport From: "EIS INC" ;tag=as47348223 To: Contact: Call-ID: 4bf855f13ee00587731a850a7f16a332@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10960 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 721 asterisk-price*CLI> Audio is at 10.200.26.202 port 10402 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK5cc76bb7;rport From: "EIS INC" ;tag=as592b66c1 To: Contact: Call-ID: 45bcedcf120387bb06f43bc47237a6ba@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10402 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-price*CLI> -- Called 724 asterisk-price*CLI> Audio is at 10.200.26.202 port 10928 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK1f83ff98;rport From: "EIS INC" ;tag=as1f22ddee To: Contact: Call-ID: 6cfda91e69e66864566c52ae70a554d3@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10928 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-price*CLI> -- Called 725 asterisk-price*CLI> Audio is at 10.200.26.202 port 10280 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK6f800d32;rport From: "EIS INC" ;tag=as63f1c3e6 To: Contact: Call-ID: 691a06be44258327400d61731cae4679@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10280 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-price*CLI> -- Called 726 asterisk-price*CLI> Audio is at 10.200.26.202 port 10320 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK50f3170e;rport From: "EIS INC" ;tag=as0c7c7e2f To: Contact: Call-ID: 02e3ebc2411854fb4cd7d58b5dd614fb@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10320 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-price*CLI> -- Called 727 Audio is at 10.200.26.202 port 10700 asterisk-price*CLI> Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK587646a2;rport From: "EIS INC" ;tag=as0ca36b08 To: Contact: Call-ID: 2658856a4574389a7af64acc062c757c@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10700 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-price*CLI> -- Called 728 asterisk-price*CLI> Audio is at 10.200.26.202 port 10232 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK20d3dbd1;rport From: "EIS INC" ;tag=as78920ee2 To: Contact: Call-ID: 593e399d7fb8216f118d4ade632d1c50@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10232 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-price*CLI> -- Called 729 asterisk-price*CLI> Audio is at 10.200.26.202 port 10648 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK3c408f5f;rport From: "EIS INC" ;tag=as66050a55 To: Contact: Call-ID: 6eac00d97a1f172e75966ed160d9dd6a@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10648 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk-price*CLI> -- Called 730 asterisk-price*CLI> Audio is at 10.200.26.202 port 10858 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (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=z9hG4bK46395de4;rport From: "EIS INC" ;tag=as57662dfc To: Contact: Call-ID: 177b0d7b1349e50b70a9eb6c6380ad84@10.200.26.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 May 2007 15:01:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 207 v=0 o=root 1380 1380 IN IP4 10.200.26.202 s=session c=IN IP4 10.200.26.202 t=0 0 m=audio 10858 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 731 asterisk-price*CLI> -- Remote UNIX connection disconnected asterisk-price*CLI>