<--- SIP read from UDP:129.46.78.62:5060 ---> INVITE sip:6000@129.46.72.197:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 129.46.78.62:5060;branch=z9hG4bK88bcd301137BC796 From: "6003" ;tag=9970AA57-E19033EC To: CSeq: 2 INVITE Call-ID: 5ecc0e98-3f23d7ad-1138f642@129.46.78.62 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.2.0078 Supported: 100rel,replaces Allow-Events: talk,hold,conference Authorization: Digest username="Polycom3", realm="asterisk", nonce="60a71f59", uri="sip:6000@129.46.72.197:5060;user=phone", response="cce0bb0dd9d42691e065352e30e96d48", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 273 v=0 o=- 1245798614 1245798614 IN IP4 129.46.78.62 s=Polycom IP Phone c=IN IP4 129.46.78.62 t=0 0 m=audio 2252 RTP/AVP 9 0 8 18 101 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (15 headers 12 lines) --- Sending to 129.46.78.62 : 5060 (no NAT) Using INVITE request as basis request - 5ecc0e98-3f23d7ad-1138f642@129.46.78.62 Found peer 'Polycom3' for 'Polycom3' from 129.46.78.62:5060 Found RTP audio format 9 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 129.46.78.62:2252 Found audio description format G722 for ID 9 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8100e (gsm|ulaw|alaw|g722|h263), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100c (ulaw|alaw|g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 129.46.78.62:2252 Looking for 6000 in from-sip (domain 129.46.72.197) list_route: hop: <--- Transmitting (no NAT) to 129.46.78.62:5060 ---> SIP/2.0 100 Trying v: SIP/2.0/UDP 129.46.78.62:5060;branch=z9hG4bK88bcd301137BC796;received=129.46.78.62 f: "6003" ;tag=9970AA57-E19033EC t: i: 5ecc0e98-3f23d7ad-1138f642@129.46.78.62 CSeq: 2 INVITE Server: pindrop-mgw Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY k: replaces, timer m: l: 0 <------------> -- Executing [6000@from-sip:1] Dial("SIP/Polycom3-0897efe0", "SIP/Polycom1") in new stack == Using SIP RTP CoS mark 5 Audio is at 129.46.72.175 port 15704 Adding codec 0x1000 (g722) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 129.46.78.161:5060: INVITE sip:Polycom1@129.46.78.161 SIP/2.0 v: SIP/2.0/UDP 129.46.72.175:5060;branch=z9hG4bK7941aefb;rport Max-Forwards: 70 f: "6003" ;tag=as6475383d t: m: i: 15fc83e827d3c3a879f017d94aab47c5@129.46.72.175 CSeq: 102 INVITE User-Agent: pindrop-mgw Date: Tue, 23 Jun 2009 23:00:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY k: replaces, timer c: application/sdp l: 342 v=0 o=root 1672693545 1672693545 IN IP4 129.46.72.175 s=Asterisk PBX 1.6.2.0-beta3 c=IN IP4 129.46.72.175 t=0 0 m=audio 15704 RTP/AVP 9 0 3 8 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/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 --- -- Called Polycom1 <--- SIP read from UDP:129.46.78.161:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 129.46.72.175:5060;branch=z9hG4bK7941aefb;rport From: "6003" ;tag=as6475383d To: ;tag=E5387CFF-698B2454 CSeq: 102 INVITE Call-ID: 15fc83e827d3c3a879f017d94aab47c5@129.46.72.175 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.2.0078 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:129.46.78.161:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 129.46.72.175:5060;branch=z9hG4bK7941aefb;rport From: "6003" ;tag=as6475383d To: ;tag=E5387CFF-698B2454 CSeq: 102 INVITE Call-ID: 15fc83e827d3c3a879f017d94aab47c5@129.46.72.175 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.2.0078 Allow-Events: talk,hold,conference Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/Polycom1-089825f0 is ringing <--- Transmitting (no NAT) to 129.46.78.62:5060 ---> SIP/2.0 180 Ringing v: SIP/2.0/UDP 129.46.78.62:5060;branch=z9hG4bK88bcd301137BC796;received=129.46.78.62 f: "6003" ;tag=9970AA57-E19033EC t: ;tag=as4d982cdf i: 5ecc0e98-3f23d7ad-1138f642@129.46.78.62 CSeq: 2 INVITE Server: pindrop-mgw Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY k: replaces, timer m: l: 0 <------------> <--- SIP read from UDP:129.46.78.161:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 129.46.72.175:5060;branch=z9hG4bK7941aefb;rport From: "6003" ;tag=as6475383d To: ;tag=E5387CFF-698B2454 CSeq: 102 INVITE Call-ID: 15fc83e827d3c3a879f017d94aab47c5@129.46.72.175 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.2.0078 Content-Type: application/sdp Content-Length: 201 v=0 o=- 1245797833 1245797833 IN IP4 129.46.78.161 s=Polycom IP Phone c=IN IP4 129.46.78.161 t=0 0 m=audio 2244 RTP/AVP 9 101 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (11 headers 9 lines) --- Found RTP audio format 9 Found RTP audio format 101 Peer audio RTP is at port 129.46.78.161:2244 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8100e (gsm|ulaw|alaw|g722|h263), 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 129.46.78.161:2244 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 129.46.78.161, port 5060 Transmitting (no NAT) to 129.46.78.161:5060: ACK sip:Polycom1@129.46.78.161 SIP/2.0 v: SIP/2.0/UDP 129.46.72.175:5060;branch=z9hG4bK0958b29a;rport Max-Forwards: 70 f: "6003" ;tag=as6475383d t: ;tag=E5387CFF-698B2454 m: i: 15fc83e827d3c3a879f017d94aab47c5@129.46.72.175 CSeq: 102 ACK User-Agent: pindrop-mgw l: 0 --- -- SIP/Polycom1-089825f0 answered SIP/Polycom3-0897efe0 Audio is at 129.46.72.175 port 14626 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 (no NAT) to 129.46.78.62:5060 ---> SIP/2.0 200 OK v: SIP/2.0/UDP 129.46.78.62:5060;branch=z9hG4bK88bcd301137BC796;received=129.46.78.62 f: "6003" ;tag=9970AA57-E19033EC t: ;tag=as4d982cdf i: 5ecc0e98-3f23d7ad-1138f642@129.46.78.62 CSeq: 2 INVITE Server: pindrop-mgw Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY k: replaces, timer m: c: application/sdp l: 317 v=0 o=root 370147630 370147630 IN IP4 129.46.72.175 s=Asterisk PBX 1.6.2.0-beta3 c=IN IP4 129.46.72.175 t=0 0 m=audio 14626 RTP/AVP 9 0 8 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Native bridging SIP/Polycom3-0897efe0 and SIP/Polycom1-089825f0 set_destination: Parsing for address/port to send to set_destination: set destination to 129.46.78.161, port 5060 Audio is at 129.46.72.175 port 15704 Adding codec 0x1000 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 129.46.78.161:5060: INVITE sip:Polycom1@129.46.78.161 SIP/2.0 v: SIP/2.0/UDP 129.46.72.175:5060;branch=z9hG4bK498b9e3d;rport Max-Forwards: 70 f: "6003" ;tag=as6475383d t: ;tag=E5387CFF-698B2454 m: i: 15fc83e827d3c3a879f017d94aab47c5@129.46.72.175 CSeq: 103 INVITE User-Agent: pindrop-mgw Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY k: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) c: application/sdp l: 268 v=0 o=root 1672693545 1672693546 IN IP4 129.46.78.62 s=Asterisk PBX 1.6.2.0-beta3 c=IN IP4 129.46.78.62 t=0 0 m=audio 2252 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from UDP:129.46.78.62:5060 ---> ACK sip:6000@129.46.72.175 SIP/2.0 Via: SIP/2.0/UDP 129.46.78.62:5060;branch=z9hG4bK58b7f0ea554150FF From: "6003" ;tag=9970AA57-E19033EC To: ;tag=as4d982cdf CSeq: 2 ACK Call-ID: 5ecc0e98-3f23d7ad-1138f642@129.46.78.62 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.2.0078 Authorization: Digest username="Polycom3", realm="asterisk", nonce="60a71f59", uri="sip:6000@129.46.72.197:5060;user=phone", response="cce0bb0dd9d42691e065352e30e96d48", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Segmentation fault (core dumped) [root@localhost wizard]#