PR-EGUK-IPPBX-1*CLI> == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL CoS mark 5 -- Executing [2999@DLPN_StandardUser:1] Dial("SIP/2000-00000011", "SIP/2999") in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL CoS mark 5 Audio is at 212.62.8.168 port 19000 Adding codec 0x1000 (g722) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 195.59.152.66:27678: INVITE sip:2999@192.168.1.165:5060 SIP/2.0 Via: SIP/2.0/UDP 212.62.8.168:5060;branch=z9hG4bK5e531981;rport Max-Forwards: 70 From: "Stephanie CEO" ;tag=as1c402420 To: Contact: Call-ID: 1a3b004a74a371587888018130103d96@212.62.8.168 CSeq: 102 INVITE User-Agent: Asterisk PBX Remote-Party-ID: "Stephanie CEO" ;privacy=off;screen=no Date: Mon, 22 Feb 2010 13:51:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 311 v=0 o=root 1503367970 1503367970 IN IP4 192.168.1.60 s=Asterisk PBX 1.6.1.11 c=IN IP4 192.168.1.60 t=0 0 m=audio 2238 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 2999 PR-EGUK-IPPBX-1*CLI> <--- SIP read from UDP://195.59.152.66:27678 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.62.8.168:5060;branch=z9hG4bK5e531981;rport From: "Stephanie CEO" ;tag=as1c402420 To: "NAP test 550" ;tag=7ED69C0A-97839EEF CSeq: 102 INVITE Call-ID: 1a3b004a74a371587888018130103d96@212.62.8.168 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 Accept-Language: en Content-Length: 0 <-------------> --- (10 headers 0 lines) --- PR-EGUK-IPPBX-1*CLI> <--- SIP read from UDP://195.59.152.66:27678 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 212.62.8.168:5060;branch=z9hG4bK5e531981;rport From: "Stephanie CEO" ;tag=as1c402420 To: "NAP test 550" ;tag=7ED69C0A-97839EEF CSeq: 102 INVITE Call-ID: 1a3b004a74a371587888018130103d96@212.62.8.168 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 Allow-Events: talk,hold,conference Accept-Language: en Content-Length: 0 <-------------> --- (11 headers 0 lines) --- -- SIP/2999-00000012 is ringing PR-EGUK-IPPBX-1*CLI> <--- SIP read from UDP://195.59.152.66:27678 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 212.62.8.168:5060;branch=z9hG4bK5e531981;rport From: "Stephanie CEO" ;tag=as1c402420 To: "NAP test 550" ;tag=7ED69C0A-97839EEF CSeq: 102 INVITE Call-ID: 1a3b004a74a371587888018130103d96@212.62.8.168 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 Accept-Language: en Content-Type: application/sdp Content-Length: 201 v=0 o=- 1266761225 1266761225 IN IP4 192.168.1.165 s=Polycom IP Phone c=IN IP4 192.168.1.165 t=0 0 m=audio 2226 RTP/AVP 9 127 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 <-------------> --- (13 headers 9 lines) --- Found RTP audio format 9 Found RTP audio format 127 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 127 Capabilities: us - 0x100c (ulaw|alaw|g722), peer - audio=0x1000 (g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1000 (g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.165:2226 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.165, port 5060 Transmitting (NAT) to 195.59.152.66:27678: ACK sip:2999@192.168.1.165:5060 SIP/2.0 Via: SIP/2.0/UDP 212.62.8.168:5060;branch=z9hG4bK00e63090;rport Max-Forwards: 70 From: "Stephanie CEO" ;tag=as1c402420 To: ;tag=7ED69C0A-97839EEF Contact: Call-ID: 1a3b004a74a371587888018130103d96@212.62.8.168 CSeq: 102 ACK User-Agent: Asterisk PBX Remote-Party-ID: "Stephanie CEO" ;privacy=off;screen=no Content-Length: 0 --- -- SIP/2999-00000012 answered SIP/2000-00000011 -- Native bridging SIP/2000-00000011 and SIP/2999-00000012 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.165, port 5060 Audio is at 212.62.8.168 port 19000 Adding codec 0x1000 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 195.59.152.66:27678: INVITE sip:2999@192.168.1.165:5060 SIP/2.0 Via: SIP/2.0/UDP 212.62.8.168:5060;branch=z9hG4bK230aa713;rport Max-Forwards: 70 From: "Stephanie CEO" ;tag=as1c402420 To: ;tag=7ED69C0A-97839EEF Contact: Call-ID: 1a3b004a74a371587888018130103d96@212.62.8.168 CSeq: 103 INVITE User-Agent: Asterisk PBX Remote-Party-ID: "Stephanie CEO" ;privacy=off;screen=no Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 264 v=0 o=root 1503367970 1503367971 IN IP4 212.62.8.168 s=Asterisk PBX 1.6.1.11 c=IN IP4 212.62.8.168 t=0 0 m=audio 19000 RTP/AVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Started music on hold, class 'default', on SIP/2999-00000012 PR-EGUK-IPPBX-1*CLI> <--- SIP read from UDP://195.59.152.66:27678 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 212.62.8.168:5060;branch=z9hG4bK230aa713;rport From: "Stephanie CEO" ;tag=as1c402420 To: "NAP test 550" ;tag=7ED69C0A-97839EEF CSeq: 103 INVITE Call-ID: 1a3b004a74a371587888018130103d96@212.62.8.168 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 Accept-Language: en Content-Type: application/sdp Content-Length: 201 v=0 o=- 1266761225 1266761226 IN IP4 192.168.1.165 s=Polycom IP Phone c=IN IP4 192.168.1.165 t=0 0 m=audio 2226 RTP/AVP 9 127 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 <-------------> --- (13 headers 9 lines) --- Found RTP audio format 9 Found RTP audio format 127 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 127 Capabilities: us - 0x100c (ulaw|alaw|g722), peer - audio=0x1000 (g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1000 (g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.165:2226 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.165, port 5060 Transmitting (NAT) to 195.59.152.66:27678: ACK sip:2999@192.168.1.165:5060 SIP/2.0 Via: SIP/2.0/UDP 212.62.8.168:5060;branch=z9hG4bK0a377b68;rport Max-Forwards: 70 From: "Stephanie CEO" ;tag=as1c402420 To: ;tag=7ED69C0A-97839EEF Contact: Call-ID: 1a3b004a74a371587888018130103d96@212.62.8.168 CSeq: 103 ACK User-Agent: Asterisk PBX Remote-Party-ID: "Stephanie CEO" ;privacy=off;screen=no Content-Length: 0 --- -- Stopped music on hold on SIP/2999-00000012 -- Started music on hold, class 'default', on SIP/2999-00000012 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.165, port 5060 Audio is at 212.62.8.168 port 19000 Adding codec 0x1000 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 195.59.152.66:27678: INVITE sip:2999@192.168.1.165:5060 SIP/2.0 Via: SIP/2.0/UDP 212.62.8.168:5060;branch=z9hG4bK0b40ee55;rport Max-Forwards: 70 From: "Stephanie CEO" ;tag=as1c402420 To: ;tag=7ED69C0A-97839EEF Contact: Call-ID: 1a3b004a74a371587888018130103d96@212.62.8.168 CSeq: 104 INVITE User-Agent: Asterisk PBX Remote-Party-ID: "Stephanie CEO" ;privacy=off;screen=no Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 263 v=0 o=root 1503367970 1503367972 IN IP4 192.168.1.60 s=Asterisk PBX 1.6.1.11 c=IN IP4 192.168.1.60 t=0 0 m=audio 2238 RTP/AVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Stopped music on hold on SIP/2999-00000012 PR-EGUK-IPPBX-1*CLI> <--- SIP read from UDP://195.59.152.66:27678 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 212.62.8.168:5060;branch=z9hG4bK0b40ee55;rport From: "Stephanie CEO" ;tag=as1c402420 To: "NAP test 550" ;tag=7ED69C0A-97839EEF CSeq: 104 INVITE Call-ID: 1a3b004a74a371587888018130103d96@212.62.8.168 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 Accept-Language: en Content-Type: application/sdp Content-Length: 201 v=0 o=- 1266761225 1266761227 IN IP4 192.168.1.165 s=Polycom IP Phone c=IN IP4 192.168.1.165 t=0 0 m=audio 2226 RTP/AVP 9 127 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 <-------------> --- (13 headers 9 lines) --- Found RTP audio format 9 Found RTP audio format 127 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 127 Capabilities: us - 0x100c (ulaw|alaw|g722), peer - audio=0x1000 (g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1000 (g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.165:2226 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.165, port 5060 Transmitting (NAT) to 195.59.152.66:27678: ACK sip:2999@192.168.1.165:5060 SIP/2.0 Via: SIP/2.0/UDP 212.62.8.168:5060;branch=z9hG4bK2715e433;rport Max-Forwards: 70 From: "Stephanie CEO" ;tag=as1c402420 To: ;tag=7ED69C0A-97839EEF Contact: Call-ID: 1a3b004a74a371587888018130103d96@212.62.8.168 CSeq: 104 ACK User-Agent: Asterisk PBX Remote-Party-ID: "Stephanie CEO" ;privacy=off;screen=no Content-Length: 0 --- PR-EGUK-IPPBX-1*CLI> <--- SIP read from UDP://195.59.152.66:27678 ---> BYE sip:2000@212.62.8.168 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK6c23bd51BB0BE2EA From: "NAP test 550" ;tag=7ED69C0A-97839EEF To: "Stephanie CEO" ;tag=as1c402420 CSeq: 1 BYE Call-ID: 1a3b004a74a371587888018130103d96@212.62.8.168 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 195.59.152.66 : 27678 (NAT) PR-EGUK-IPPBX-1*CLI> sip show history <--- Transmitting (NAT) to 195.59.152.66:27678 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.165:5060;branch=z9hG4bK6c23bd51BB0BE2EA;received=195.59.152.66 From: "NAP test 550" ;tag=7ED69C0A-97839EEF To: "Stephanie CEO" ;tag=as1c402420 Call-ID: 1a3b004a74a371587888018130103d96@212.62.8.168 CSeq: 1 BYE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 ------------>-1*CLI> sip show history == Spawn extension (DLPN_StandardUser, 2999, 1) exited non-zero on 'SIP/2000-00000011' Really destroying SIP dialog '1a3b004a74a371587888018130103d96@212.62.8.168' Method: BYE