<-- SIP read from 192.168.10.86:1024: INVITE sip:2345@192.168.10.86 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.86:1024;rport;branch=z9hG4bKc0a82f01000000b9455b65412d974d2e00000371 Content-Length: 368 Contact: Call-ID: 6D216858-1DD2-11B2-8B19-F4C9A83F7A4C@192.168.47.1 Content-Type: application/sdp CSeq: 1 INVITE From: "unknown";tag=39764310891868888273 Max-Forwards: 70 To: User-Agent: SJphone/1.60.299a/L (SJ Labs) v=0 o=- 3372606401 3372606401 IN IP4 192.168.10.86 s=SJphone c=IN IP4 192.168.10.86 t=0 0 a=direction:active m=audio 49166 RTP/AVP 3 97 98 110 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --- (11 headers 16 lines) --- Using INVITE request as basis request - 6D216858-1DD2-11B2-8B19-F4C9A83F7A4C@192.168.47.1 Sending to 192.168.10.86 : 1024 (non-NAT) Found user '1234' Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 110 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.10.86:49166 Found description format GSM Found description format iLBC Found description format iLBC Found description format speex Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 2345 in default (domain 192.168.10.86) list_route: hop: Transmitting (no NAT) to 192.168.10.86:1024: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.86:1024;rport;branch=z9hG4bKc0a82f01000000b9455b65412d974d2e00000371;received=192.168.10.86 From: "unknown";tag=39764310891868888273 To: Call-ID: 6D216858-1DD2-11B2-8B19-F4C9A83F7A4C@192.168.47.1 CSeq: 1 INVITE User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- -- Executing Dial("SIP/1234-de50", "sip/2345") in new stack Setting up to call extension '' at '2345' We're at 192.168.10.86 port 15824 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (no NAT) to 192.168.10.212:5060: INVITE sip:2345@192.168.10.212:5060;line=fz4zp37j SIP/2.0 Via: SIP/2.0/UDP 192.168.10.86:5060;branch=z9hG4bK1d401fdb;rport From: "unknown" ;tag=as0d9e6784 To: Contact: Call-ID: 1a2d6433731de57f7aa67d41181d3533@192.168.10.86 CSeq: 102 INVITE User-Agent: OpenPBX Max-Forwards: 70 Date: Wed, 15 Nov 2006 19:06:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 263 v=0 o=root 8711 8711 IN IP4 192.168.10.86 s=session c=IN IP4 192.168.10.86 t=0 0 m=audio 15824 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 2345 <-- SIP read from 192.168.10.212:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.86:5060;branch=z9hG4bK1d401fdb;rport=5060 From: "unknown" ;tag=as0d9e6784 To: ;tag=0yx149j2ot Call-ID: 1a2d6433731de57f7aa67d41181d3533@192.168.10.86 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 --- (10 headers 0 lines) --- -- SIP/2345-dbd2 is ringing Transmitting (no NAT) to 192.168.10.86:1024: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.86:1024;rport;branch=z9hG4bKc0a82f01000000b9455b65412d974d2e00000371;received=192.168.10.86 From: "unknown";tag=39764310891868888273 To: ;tag=as365e1742 Call-ID: 6D216858-1DD2-11B2-8B19-F4C9A83F7A4C@192.168.47.1 CSeq: 1 INVITE User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- <-- SIP read from 192.168.10.212:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.86:5060;branch=z9hG4bK1d401fdb;rport=5060 From: "unknown" ;tag=as0d9e6784 To: ;tag=0yx149j2ot Call-ID: 1a2d6433731de57f7aa67d41181d3533@192.168.10.86 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 --- (10 headers 0 lines) --- -- SIP/2345-dbd2 is ringing <-- SIP read from 192.168.10.212:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.86:5060;branch=z9hG4bK1d401fdb;rport=5060 From: "unknown" ;tag=as0d9e6784 To: ;tag=0yx149j2ot Call-ID: 1a2d6433731de57f7aa67d41181d3533@192.168.10.86 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 --- (10 headers 0 lines) --- -- SIP/2345-dbd2 is ringing <-- SIP read from 192.168.10.212:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.86:5060;branch=z9hG4bK1d401fdb;rport=5060 From: "unknown" ;tag=as0d9e6784 To: ;tag=0yx149j2ot Call-ID: 1a2d6433731de57f7aa67d41181d3533@192.168.10.86 CSeq: 102 INVITE Contact: User-Agent: snom200-3.56z Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Content-Type: application/sdp Content-Length: 211 v=0 o=root 1855868769 1855868769 IN IP4 192.168.10.212 s=call c=IN IP4 192.168.10.212 t=0 0 m=audio 10322 RTP/AVP 3 101 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (13 headers 10 lines) --- Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 192.168.10.212:10322 Found description format gsm Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.10.212, port 5060 Transmitting (no NAT) to 192.168.10.212:5060: ACK sip:2345@192.168.10.212:5060;line=fz4zp37j SIP/2.0 Via: SIP/2.0/UDP 192.168.10.86:5060;branch=z9hG4bK7a10771c;rport From: "unknown" ;tag=as0d9e6784 To: ;tag=0yx149j2ot Contact: Call-ID: 1a2d6433731de57f7aa67d41181d3533@192.168.10.86 CSeq: 102 ACK User-Agent: OpenPBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/2345-dbd2 answered SIP/1234-de50 We're at 192.168.10.86 port 12360 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.10.86:1024: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.86:1024;rport;branch=z9hG4bKc0a82f01000000b9455b65412d974d2e00000371;received=192.168.10.86 From: "unknown";tag=39764310891868888273 To: ;tag=as365e1742 Call-ID: 6D216858-1DD2-11B2-8B19-F4C9A83F7A4C@192.168.47.1 CSeq: 1 INVITE User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 8711 8711 IN IP4 192.168.10.86 s=session c=IN IP4 192.168.10.86 t=0 0 m=audio 12360 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/1234-de50 and SIP/2345-dbd2 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.10.212, port 5060 We're at 192.168.10.86 port 15824 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x200 (speex) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 14 lines Reliably Transmitting (no NAT) to 192.168.10.212:5060: INVITE sip:2345@192.168.10.212:5060;line=fz4zp37j SIP/2.0 Via: SIP/2.0/UDP 192.168.10.86:5060;branch=z9hG4bK22713795;rport From: "unknown" ;tag=as0d9e6784 To: ;tag=0yx149j2ot Contact: Call-ID: 1a2d6433731de57f7aa67d41181d3533@192.168.10.86 CSeq: 103 INVITE User-Agent: OpenPBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-openpbx-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 318 v=0 o=root 8711 8712 IN IP4 192.168.10.86 s=session c=IN IP4 192.168.10.86 t=0 0 m=audio 49166 RTP/AVP 3 0 8 110 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-- SIP read from 192.168.10.86:1024: INVITE sip:2345@192.168.10.86 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.86:1024;rport;branch=z9hG4bKc0a82f01000000b9455b65412d974d2e00000371 Content-Length: 368 Contact: Call-ID: 6D216858-1DD2-11B2-8B19-F4C9A83F7A4C@192.168.47.1 Content-Type: application/sdp CSeq: 1 INVITE From: "unknown";tag=39764310891868888273 Max-Forwards: 70 To: User-Agent: SJphone/1.60.299a/L (SJ Labs) v=0 o=- 3372606401 3372606401 IN IP4 192.168.10.86 s=SJphone c=IN IP4 192.168.10.86 t=0 0 a=direction:active m=audio 49166 RTP/AVP 3 97 98 110 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --- (11 headers 16 lines) --- Using INVITE request as basis request - 6D216858-1DD2-11B2-8B19-F4C9A83F7A4C@192.168.47.1 Sending to 192.168.10.86 : 1024 (non-NAT) Found user '1234' Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 110 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.10.86:49166 Found description format GSM Found description format iLBC Found description format iLBC Found description format speex Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 2345 in default (domain 192.168.10.86) list_route: hop: Transmitting (no NAT) to 192.168.10.86:1024: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.86:1024;rport;branch=z9hG4bKc0a82f01000000b9455b65412d974d2e00000371;received=192.168.10.86 From: "unknown";tag=39764310891868888273 To: Call-ID: 6D216858-1DD2-11B2-8B19-F4C9A83F7A4C@192.168.47.1 CSeq: 1 INVITE User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- -- Executing Dial("SIP/1234-de50", "sip/2345") in new stack Setting up to call extension '' at '2345' We're at 192.168.10.86 port 15824 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (no NAT) to 192.168.10.212:5060: INVITE sip:2345@192.168.10.212:5060;line=fz4zp37j SIP/2.0 Via: SIP/2.0/UDP 192.168.10.86:5060;branch=z9hG4bK1d401fdb;rport From: "unknown" ;tag=as0d9e6784 To: Contact: Call-ID: 1a2d6433731de57f7aa67d41181d3533@192.168.10.86 CSeq: 102 INVITE User-Agent: OpenPBX Max-Forwards: 70 Date: Wed, 15 Nov 2006 19:06:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 263 v=0 o=root 8711 8711 IN IP4 192.168.10.86 s=session c=IN IP4 192.168.10.86 t=0 0 m=audio 15824 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 2345 <-- SIP read from 192.168.10.212:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.86:5060;branch=z9hG4bK1d401fdb;rport=5060 From: "unknown" ;tag=as0d9e6784 To: ;tag=0yx149j2ot Call-ID: 1a2d6433731de57f7aa67d41181d3533@192.168.10.86 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 --- (10 headers 0 lines) --- -- SIP/2345-dbd2 is ringing Transmitting (no NAT) to 192.168.10.86:1024: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.86:1024;rport;branch=z9hG4bKc0a82f01000000b9455b65412d974d2e00000371;received=192.168.10.86 From: "unknown";tag=39764310891868888273 To: ;tag=as365e1742 Call-ID: 6D216858-1DD2-11B2-8B19-F4C9A83F7A4C@192.168.47.1 CSeq: 1 INVITE User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- <-- SIP read from 192.168.10.212:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.86:5060;branch=z9hG4bK1d401fdb;rport=5060 From: "unknown" ;tag=as0d9e6784 To: ;tag=0yx149j2ot Call-ID: 1a2d6433731de57f7aa67d41181d3533@192.168.10.86 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 --- (10 headers 0 lines) --- -- SIP/2345-dbd2 is ringing <-- SIP read from 192.168.10.212:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.86:5060;branch=z9hG4bK1d401fdb;rport=5060 From: "unknown" ;tag=as0d9e6784 To: ;tag=0yx149j2ot Call-ID: 1a2d6433731de57f7aa67d41181d3533@192.168.10.86 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 --- (10 headers 0 lines) --- -- SIP/2345-dbd2 is ringing <-- SIP read from 192.168.10.212:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.10.86:5060;branch=z9hG4bK1d401fdb;rport=5060 From: "unknown" ;tag=as0d9e6784 To: ;tag=0yx149j2ot Call-ID: 1a2d6433731de57f7aa67d41181d3533@192.168.10.86 CSeq: 102 INVITE Contact: User-Agent: snom200-3.56z Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Content-Type: application/sdp Content-Length: 211 v=0 o=root 1855868769 1855868769 IN IP4 192.168.10.212 s=call c=IN IP4 192.168.10.212 t=0 0 m=audio 10322 RTP/AVP 3 101 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (13 headers 10 lines) --- Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 192.168.10.212:10322 Found description format gsm Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.10.212, port 5060 Transmitting (no NAT) to 192.168.10.212:5060: ACK sip:2345@192.168.10.212:5060;line=fz4zp37j SIP/2.0 Via: SIP/2.0/UDP 192.168.10.86:5060;branch=z9hG4bK7a10771c;rport From: "unknown" ;tag=as0d9e6784 To: ;tag=0yx149j2ot Contact: Call-ID: 1a2d6433731de57f7aa67d41181d3533@192.168.10.86 CSeq: 102 ACK User-Agent: OpenPBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/2345-dbd2 answered SIP/1234-de50 We're at 192.168.10.86 port 12360 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.10.86:1024: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.86:1024;rport;branch=z9hG4bKc0a82f01000000b9455b65412d974d2e00000371;received=192.168.10.86 From: "unknown";tag=39764310891868888273 To: ;tag=as365e1742 Call-ID: 6D216858-1DD2-11B2-8B19-F4C9A83F7A4C@192.168.47.1 CSeq: 1 INVITE User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 8711 8711 IN IP4 192.168.10.86 s=session c=IN IP4 192.168.10.86 t=0 0 m=audio 12360 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/1234-de50 and SIP/2345-dbd2 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.10.212, port 5060 We're at 192.168.10.86 port 15824 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x200 (speex) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 14 lines Reliably Transmitting (no NAT) to 192.168.10.212:5060: INVITE sip:2345@192.168.10.212:5060;line=fz4zp37j SIP/2.0 Via: SIP/2.0/UDP 192.168.10.86:5060;branch=z9hG4bK22713795;rport From: "unknown" ;tag=as0d9e6784 To: ;tag=0yx149j2ot Contact: Call-ID: 1a2d6433731de57f7aa67d41181d3533@192.168.10.86 CSeq: 103 INVITE User-Agent: OpenPBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-openpbx-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 318 v=0 o=root 8711 8712 IN IP4 192.168.10.86 s=session c=IN IP4 192.168.10.86 t=0 0 m=audio 49166 RTP/AVP 3 0 8 110 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -