[Feb 1 20:19:25] VERBOSE[13909] logger.c: <--- SIP read from 12.37.6.160:5060 ---> INVITE sip:17185387024@sip-us.ziggoo.net SIP/2.0 Via: SIP/2.0/UDP 12.37.6.160:5060;branch=z9hG4bK1944e697;rport From: "Espacio" ;tag=as79712a0e To: Contact: Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 02 Feb 2007 01:19:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 309 v=0 o=root 2617 2617 IN IP4 12.37.6.160 s=session c=IN IP4 12.37.6.160 t=0 0 m=audio 15580 RTP/AVP 0 18 8 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> [Feb 1 20:19:25] VERBOSE[13909] logger.c: --- (13 headers 14 lines) --- [Feb 1 20:19:25] VERBOSE[13909] logger.c: Sending to 12.37.6.160 : 5060 (NAT) [Feb 1 20:19:25] VERBOSE[13909] logger.c: Using INVITE request as basis request - 6ab3df5d298822913fa87a2f788751d0@12.37.6.160 [Feb 1 20:19:25] VERBOSE[13909] logger.c: Found peer 'slogica' [Feb 1 20:19:25] VERBOSE[13909] logger.c: Found RTP audio format 0 [Feb 1 20:19:25] VERBOSE[13909] logger.c: Found RTP audio format 18 [Feb 1 20:19:25] VERBOSE[13909] logger.c: Found RTP audio format 8 [Feb 1 20:19:25] VERBOSE[13909] logger.c: Found RTP audio format 97 [Feb 1 20:19:25] VERBOSE[13909] logger.c: Found RTP audio format 101 [Feb 1 20:19:25] DEBUG[13909] chan_sip.c: Peer doesn't provide T.38 UDPTL [Feb 1 20:19:25] VERBOSE[13909] logger.c: Peer audio RTP is at port 12.37.6.160:15580 [Feb 1 20:19:25] VERBOSE[13909] logger.c: Found description format PCMU for ID 0 [Feb 1 20:19:25] VERBOSE[13909] logger.c: Found description format G729 for ID 18 [Feb 1 20:19:25] VERBOSE[13909] logger.c: Found description format PCMA for ID 8 [Feb 1 20:19:25] VERBOSE[13909] logger.c: Found description format iLBC for ID 97 [Feb 1 20:19:25] VERBOSE[13909] logger.c: Found description format telephone-event for ID 101 [Feb 1 20:19:25] VERBOSE[13909] logger.c: Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) [Feb 1 20:19:25] VERBOSE[13909] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 1 20:19:25] VERBOSE[13909] logger.c: Peer audio RTP is at port 12.37.6.160:15580 [Feb 1 20:19:25] VERBOSE[13909] logger.c: Looking for 17185387024 in slogica (domain sip-us.ziggoo.net) [Feb 1 20:19:25] VERBOSE[13909] logger.c: list_route: hop: [Feb 1 20:19:25] VERBOSE[13909] logger.c: <--- Transmitting (no NAT) to 12.37.6.160:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 12.37.6.160:5060;branch=z9hG4bK1944e697;received=12.37.6.160;rport=5060 From: "Espacio" ;tag=as79712a0e To: Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160 CSeq: 102 INVITE User-Agent: Ziggoo Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 1 20:19:25] VERBOSE[14426] logger.c: -- Executing [17185387024@slogica:1] Set("SIP/12.37.6.160-0849a3b0", "CDR(userfield)="Espacio" <2655209225>,0.013,6,6") in new stack [Feb 1 20:19:25] VERBOSE[14426] logger.c: -- Executing [17185387024@slogica:2] Macro("SIP/12.37.6.160-0849a3b0", "usa|17185387024") in new stack [Feb 1 20:19:25] VERBOSE[14426] logger.c: -- Executing [s@macro-usa:1] Set("SIP/12.37.6.160-0849a3b0", "CALLERID(all)=2655209225") in new stack [Feb 1 20:19:25] VERBOSE[14426] logger.c: -- Executing [s@macro-usa:2] Macro("SIP/12.37.6.160-0849a3b0", "dial|SIP/ibell-usa/17185387024") in new stack [Feb 1 20:19:25] VERBOSE[14426] logger.c: -- Executing [s@macro-dial:1] Dial("SIP/12.37.6.160-0849a3b0", "SIP/ibell-usa/17185387024||") in new stack [Feb 1 20:19:25] VERBOSE[14426] logger.c: Audio is at 208.116.0.218 port 15746 [Feb 1 20:19:25] VERBOSE[14426] logger.c: Adding codec 0x100 (g729) to SDP [Feb 1 20:19:25] VERBOSE[14426] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Feb 1 20:19:25] VERBOSE[14426] logger.c: Reliably Transmitting (no NAT) to 69.28.243.3:5060: INVITE sip:17185387024@69.28.243.3 SIP/2.0 Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK0cdb89fe;rport From: "2655209225" ;tag=as1e6f28ed To: Contact: Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218 CSeq: 102 INVITE User-Agent: Ziggoo Max-Forwards: 70 Date: Fri, 02 Feb 2007 01:19:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 13853 13853 IN IP4 208.116.0.218 s=session c=IN IP4 208.116.0.218 t=0 0 m=audio 15746 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 1 20:19:25] VERBOSE[14426] logger.c: -- Called ibell-usa/17185387024 [Feb 1 20:19:25] VERBOSE[13909] logger.c: <--- SIP read from 69.28.243.3:5060 ---> SIP/2.0 407 Proxy Authentication Required CSeq: 102 INVITE Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK0cdb89fe;rport From: "2655209225" ;tag=as1e6f28ed Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218 To: Contact: Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="117037916501200803202192504405 Content-Length: 0 <-------------> [Feb 1 20:19:25] VERBOSE[13909] logger.c: --- (9 headers 0 lines) --- [Feb 1 20:19:25] VERBOSE[13909] logger.c: Transmitting (no NAT) to 69.28.243.3:5060: ACK sip:17185387024@69.28.243.3 SIP/2.0 Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK0cdb89fe;rport From: "2655209225" ;tag=as1e6f28ed To: Contact: Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218 CSeq: 102 ACK User-Agent: Ziggoo Max-Forwards: 70 Content-Length: 0 --- [Feb 1 20:19:25] VERBOSE[13909] logger.c: Audio is at 208.116.0.218 port 15746 [Feb 1 20:19:25] VERBOSE[13909] logger.c: Adding codec 0x100 (g729) to SDP [Feb 1 20:19:25] VERBOSE[13909] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Feb 1 20:19:25] VERBOSE[13909] logger.c: Reliably Transmitting (no NAT) to 69.28.243.3:5060: INVITE sip:17185387024@69.28.243.3 SIP/2.0 Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK4677ce3c;rport From: "2655209225" ;tag=as1e6f28ed To: Contact: Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218 CSeq: 103 INVITE User-Agent: Ziggoo Max-Forwards: 70 Proxy-Authorization: Digest username="ziggoo", realm="VoipSwitch", algorithm=MD5, uri="sip:17185387024@69.28.243.3", nonce="117037916501200803202192504405", response="bfb56cc7bef14b42f436c61e437676c8", opaque="" Date: Fri, 02 Feb 2007 01:19:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 13853 13854 IN IP4 208.116.0.218 s=session c=IN IP4 208.116.0.218 t=0 0 m=audio 15746 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 1 20:19:25] VERBOSE[13909] logger.c: <--- SIP read from 69.28.243.3:5060 ---> SIP/2.0 100 Trying CSeq: 103 INVITE Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK0cdb89fe;rport From: "2655209225" ;tag=as1e6f28ed Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218 To: ;tag=0102190720256601 Contact: Content-Length: 0 <-------------> [Feb 1 20:19:28] VERBOSE[13909] logger.c: --- (10 headers 0 lines) --- [Feb 1 20:19:28] VERBOSE[13909] logger.c: Really destroying SIP dialog '4ec7cb1073acb95b7fb2952e54f9003f@208.116.0.218' Method: OPTIONS [Feb 1 20:19:29] VERBOSE[13909] logger.c: <--- SIP read from 69.28.243.3:5060 ---> SIP/2.0 180 Ringing CSeq: 103 INVITE Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK0cdb89fe;rport From: "2655209225" ;tag=as1e6f28ed Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218 To: ;tag=0102190720256601 Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=VoipSwitch 7600 7600 IN IP4 69.28.243.3 s=VoipSIP i=Audio Session c=IN IP4 69.28.243.3 t=0 0 m=audio 6600 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Feb 1 20:19:29] VERBOSE[13909] logger.c: --- (9 headers 10 lines) --- [Feb 1 20:19:29] VERBOSE[13909] logger.c: Found RTP audio format 18 [Feb 1 20:19:29] VERBOSE[13909] logger.c: Found RTP audio format 101 [Feb 1 20:19:29] DEBUG[13909] chan_sip.c: Peer doesn't provide T.38 UDPTL [Feb 1 20:19:29] VERBOSE[13909] logger.c: Peer audio RTP is at port 69.28.243.3:6600 [Feb 1 20:19:29] VERBOSE[13909] logger.c: Found description format G729 for ID 18 [Feb 1 20:19:29] VERBOSE[13909] logger.c: Found description format telephone-event for ID 101 [Feb 1 20:19:29] VERBOSE[13909] logger.c: Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) [Feb 1 20:19:29] VERBOSE[13909] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 1 20:19:29] VERBOSE[13909] logger.c: Peer audio RTP is at port 69.28.243.3:6600 [Feb 1 20:19:29] VERBOSE[14426] logger.c: -- SIP/ibell-usa-084baad8 is ringing [Feb 1 20:19:29] VERBOSE[14426] logger.c: <--- Transmitting (no NAT) to 12.37.6.160:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 12.37.6.160:5060;branch=z9hG4bK1944e697;received=12.37.6.160;rport=5060 From: "Espacio" ;tag=as79712a0e To: ;tag=as4c0b22b8 Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160 CSeq: 102 INVITE User-Agent: Ziggoo Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 1 20:19:29] VERBOSE[14426] logger.c: -- SIP/ibell-usa-084baad8 is making progress passing it to SIP/12.37.6.160-0849a3b0 [Feb 1 20:19:29] VERBOSE[14426] logger.c: Audio is at 208.116.0.218 port 12838 [Feb 1 20:19:29] VERBOSE[14426] logger.c: Adding codec 0x100 (g729) to SDP [Feb 1 20:19:29] VERBOSE[14426] logger.c: Adding codec 0x4 (ulaw) to SDP [Feb 1 20:19:29] VERBOSE[14426] logger.c: Adding codec 0x8 (alaw) to SDP [Feb 1 20:19:29] VERBOSE[14426] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Feb 1 20:19:29] VERBOSE[14426] logger.c: <--- Transmitting (no NAT) to 12.37.6.160:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 12.37.6.160:5060;branch=z9hG4bK1944e697;received=12.37.6.160;rport=5060 From: "Espacio" ;tag=as79712a0e To: ;tag=as4c0b22b8 Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160 CSeq: 102 INVITE User-Agent: Ziggoo Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 313 v=0 o=root 13853 13853 IN IP4 208.116.0.218 s=session c=IN IP4 208.116.0.218 t=0 0 m=audio 12838 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Feb 1 20:19:37] VERBOSE[13909] logger.c: <--- SIP read from 69.28.243.3:5060 ---> SIP/2.0 200 OK CSeq: 103 INVITE Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK0cdb89fe;rport From: "2655209225" ;tag=as1e6f28ed Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218 To: ;tag=0102190720256601 Contact: Content-Type: application/sdp Content-Length: 209 v=0 o=VoipSwitch 7600 7600 IN IP4 69.28.243.3 s=VoipSIP i=Audio Session c=IN IP4 69.28.243.3 t=0 0 m=audio 6600 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Feb 1 20:19:37] VERBOSE[13909] logger.c: --- (9 headers 10 lines) --- [Feb 1 20:19:37] VERBOSE[13909] logger.c: Found RTP audio format 18 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Found RTP audio format 101 [Feb 1 20:19:37] DEBUG[13909] chan_sip.c: Peer doesn't provide T.38 UDPTL [Feb 1 20:19:37] VERBOSE[13909] logger.c: Peer audio RTP is at port 69.28.243.3:6600 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Found description format G729 for ID 18 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Found description format telephone-event for ID 101 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) [Feb 1 20:19:37] VERBOSE[13909] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 1 20:19:37] VERBOSE[13909] logger.c: Peer audio RTP is at port 69.28.243.3:6600 [Feb 1 20:19:37] VERBOSE[13909] logger.c: list_route: hop: [Feb 1 20:19:37] DEBUG[13909] chan_sip.c: Strict routing enforced for session 32295ead63e2966d57419f5b386f6e7a@208.116.0.218 [Feb 1 20:19:37] VERBOSE[13909] logger.c: set_destination: Parsing for address/port to send to [Feb 1 20:19:37] VERBOSE[13909] logger.c: set_destination: set destination to 69.28.243.3, port 5060 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Transmitting (no NAT) to 69.28.243.3:5060: ACK sip:69.28.243.3:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK76a7a2f9;rport From: "2655209225" ;tag=as1e6f28ed To: ;tag=0102190720256601 Contact: Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218 CSeq: 103 ACK User-Agent: Ziggoo Max-Forwards: 70 Content-Length: 0 --- [Feb 1 20:19:37] VERBOSE[14426] logger.c: -- SIP/ibell-usa-084baad8 answered SIP/12.37.6.160-0849a3b0 [Feb 1 20:19:37] VERBOSE[14426] logger.c: Audio is at 208.116.0.218 port 12838 [Feb 1 20:19:37] VERBOSE[14426] logger.c: Adding codec 0x100 (g729) to SDP [Feb 1 20:19:37] VERBOSE[14426] logger.c: Adding codec 0x4 (ulaw) to SDP [Feb 1 20:19:37] VERBOSE[14426] logger.c: Adding codec 0x8 (alaw) to SDP [Feb 1 20:19:37] VERBOSE[14426] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Feb 1 20:19:37] VERBOSE[14426] logger.c: <--- Reliably Transmitting (no NAT) to 12.37.6.160:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 12.37.6.160:5060;branch=z9hG4bK1944e697;received=12.37.6.160;rport=5060 From: "Espacio" ;tag=as79712a0e To: ;tag=as4c0b22b8 Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160 CSeq: 102 INVITE User-Agent: Ziggoo Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 313 v=0 o=root 13853 13854 IN IP4 208.116.0.218 s=session c=IN IP4 208.116.0.218 t=0 0 m=audio 12838 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Feb 1 20:19:37] VERBOSE[14426] logger.c: -- Native bridging SIP/12.37.6.160-0849a3b0 and SIP/ibell-usa-084baad8 [Feb 1 20:19:37] DEBUG[14426] chan_sip.c: Strict routing enforced for session 32295ead63e2966d57419f5b386f6e7a@208.116.0.218 [Feb 1 20:19:37] VERBOSE[14426] logger.c: set_destination: Parsing for address/port to send to [Feb 1 20:19:37] VERBOSE[14426] logger.c: set_destination: set destination to 69.28.243.3, port 5060 [Feb 1 20:19:37] VERBOSE[14426] logger.c: Audio is at 208.116.0.218 port 15746 [Feb 1 20:19:37] VERBOSE[14426] logger.c: Adding codec 0x100 (g729) to SDP [Feb 1 20:19:37] VERBOSE[14426] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Feb 1 20:19:37] VERBOSE[14426] logger.c: Reliably Transmitting (no NAT) to 69.28.243.3:5060: INVITE sip:69.28.243.3:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK754c72cc;rport From: "2655209225" ;tag=as1e6f28ed To: ;tag=0102190720256601 Contact: Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218 CSeq: 104 INVITE User-Agent: Ziggoo Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 261 v=0 o=root 13853 13855 IN IP4 12.37.6.160 s=session c=IN IP4 12.37.6.160 t=0 0 m=audio 15580 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 1 20:19:37] VERBOSE[13909] logger.c: <--- SIP read from 12.37.6.160:5060 ---> ACK sip:17185387024@208.116.0.218 SIP/2.0 Via: SIP/2.0/UDP 12.37.6.160:5060;branch=z9hG4bK0f52c978;rport From: "Espacio" ;tag=as79712a0e To: ;tag=as4c0b22b8 Contact: Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Feb 1 20:19:37] VERBOSE[13909] logger.c: --- (10 headers 0 lines) --- [Feb 1 20:19:37] DEBUG[13909] chan_sip.c: Strict routing enforced for session 6ab3df5d298822913fa87a2f788751d0@12.37.6.160 [Feb 1 20:19:37] VERBOSE[13909] logger.c: set_destination: Parsing for address/port to send to [Feb 1 20:19:37] VERBOSE[13909] logger.c: set_destination: set destination to 12.37.6.160, port 5060 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Audio is at 208.116.0.218 port 12838 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Adding codec 0x100 (g729) to SDP [Feb 1 20:19:37] VERBOSE[13909] logger.c: Adding codec 0x4 (ulaw) to SDP [Feb 1 20:19:37] VERBOSE[13909] logger.c: Adding codec 0x8 (alaw) to SDP [Feb 1 20:19:37] VERBOSE[13909] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Feb 1 20:19:37] VERBOSE[13909] logger.c: Reliably Transmitting (no NAT) to 12.37.6.160:5060: INVITE sip:2655209225@12.37.6.160 SIP/2.0 Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK232cfe62;rport From: ;tag=as4c0b22b8 To: "Espacio" ;tag=as79712a0e Contact: Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160 CSeq: 102 INVITE User-Agent: Ziggoo Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 308 v=0 o=root 13853 13855 IN IP4 69.28.243.3 s=session c=IN IP4 69.28.243.3 t=0 0 m=audio 6600 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 1 20:19:37] VERBOSE[13909] logger.c: <--- SIP read from 12.37.6.160:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK232cfe62;rport;received=208.116.0.218 From: ;tag=as4c0b22b8 To: "Espacio" ;tag=as79712a0e Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 309 v=0 o=root 2617 2618 IN IP4 12.37.6.160 s=session c=IN IP4 12.37.6.160 t=0 0 m=audio 15580 RTP/AVP 0 18 8 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> [Feb 1 20:19:37] VERBOSE[13909] logger.c: --- (12 headers 14 lines) --- [Feb 1 20:19:37] VERBOSE[13909] logger.c: Found RTP audio format 0 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Found RTP audio format 18 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Found RTP audio format 8 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Found RTP audio format 97 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Found RTP audio format 101 [Feb 1 20:19:37] DEBUG[13909] chan_sip.c: Peer doesn't provide T.38 UDPTL [Feb 1 20:19:37] VERBOSE[13909] logger.c: Peer audio RTP is at port 12.37.6.160:15580 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Found description format PCMU for ID 0 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Found description format G729 for ID 18 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Found description format PCMA for ID 8 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Found description format iLBC for ID 97 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Found description format telephone-event for ID 101 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) [Feb 1 20:19:37] VERBOSE[13909] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 1 20:19:37] VERBOSE[13909] logger.c: Peer audio RTP is at port 12.37.6.160:15580 [Feb 1 20:19:37] VERBOSE[13909] logger.c: list_route: hop: [Feb 1 20:19:37] DEBUG[13909] chan_sip.c: Strict routing enforced for session 6ab3df5d298822913fa87a2f788751d0@12.37.6.160 [Feb 1 20:19:37] VERBOSE[13909] logger.c: set_destination: Parsing for address/port to send to [Feb 1 20:19:37] VERBOSE[13909] logger.c: set_destination: set destination to 12.37.6.160, port 5060 [Feb 1 20:19:37] VERBOSE[13909] logger.c: Transmitting (no NAT) to 12.37.6.160:5060: ACK sip:2655209225@12.37.6.160 SIP/2.0 Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK3a50185d;rport From: ;tag=as4c0b22b8 To: "Espacio" ;tag=as79712a0e Contact: Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160 CSeq: 102 ACK User-Agent: Ziggoo Max-Forwards: 70 Content-Length: 0 ---