May 30 19:58:54 VERBOSE[23174] logger.c: -- Executing Set("Zap/32-1", "CDR(accountcode)=971") in new stack May 30 19:58:54 VERBOSE[23174] logger.c: -- Executing Dial("Zap/32-1", "SIP/190400000002301@siptest1.mydomain.cc") in new stack May 30 19:58:54 VERBOSE[23174] logger.c: -- parse_srv: SRV mapped to host siptest1.mydomain.cc, port 5060 May 30 19:58:54 VERBOSE[23174] logger.c: We're at 2XX.XXX.XXX.79 port 12308 May 30 19:58:54 VERBOSE[23174] logger.c: Adding codec 0x4 (ulaw) to SDP May 30 19:58:54 VERBOSE[23174] logger.c: Adding codec 0x100 (g729) to SDP May 30 19:58:54 VERBOSE[23174] logger.c: Adding codec 0x1 (g723) to SDP May 30 19:58:54 VERBOSE[23174] logger.c: Adding codec 0x2 (gsm) to SDP May 30 19:58:54 VERBOSE[23174] logger.c: Adding codec 0x400 (ilbc) to SDP May 30 19:58:54 VERBOSE[23174] logger.c: Adding codec 0x8 (alaw) to SDP May 30 19:58:54 VERBOSE[23174] logger.c: Adding non-codec 0x1 (telephone-event) to SDP May 30 19:58:54 VERBOSE[23174] logger.c: 13 headers, 16 lines May 30 19:58:54 VERBOSE[23174] logger.c: Reliably Transmitting (no NAT) to 2XX.XXX.XXX.99:5060: INVITE sip:190400000002301@siptest1.mydomain.cc SIP/2.0 Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK0340a86c;rport From: "Unknown" ;tag=as4685a108 To: Contact: Call-ID: 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 CSeq: 102 INVITE User-Agent: MyAsteriskServer1 Max-Forwards: 70 Date: Tue, 30 May 2006 17:58:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 360 v=0 o=root 26177 26177 IN IP4 2XX.XXX.XXX.79 s=session c=IN IP4 2XX.XXX.XXX.79 t=0 0 m=audio 12308 RTP/AVP 0 18 4 3 97 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- May 30 19:58:54 VERBOSE[23174] logger.c: -- Called 190400000002301@siptest1.mydomain.cc May 30 19:58:54 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK0340a86c;rport=5060 From: "Unknown" ;tag=as4685a108 To: Call-ID: 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 CSeq: 102 INVITE Server: OpenSer (1.0.1 (i386/linux)) Content-Length: 0 Warning: 392 2XX.XXX.XXX.99:5060 "Noisy feedback tells: pid=12541 req_src_ip=2XX.XXX.XXX.79 req_src_port=5060 in_uri=sip:190400000002301@siptest1.mydomain.cc out_uri=sip:$ May 30 19:58:54 VERBOSE[26195] logger.c: --- (9 headers 0 lines)May 30 19:58:54 VERBOSE[26195] logger.c: --- (9 headers 0 lines)--- May 30 19:58:55 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK0340a86c;rport=5060 From: "Unknown" ;tag=as4685a108 To: ;tag=5CBB8D43-301D1CEC CSeq: 102 INVITE Call-ID: 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 Contact: Record-Route: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Allow-Events: talk,hold,conference Content-Length: 0 May 30 19:58:55 VERBOSE[26195] logger.c: --- (11 headers 0 lines)May 30 19:58:55 VERBOSE[26195] logger.c: --- (11 headers 0 lines)--- May 30 19:58:55 VERBOSE[23174] logger.c: -- SIP/siptest1.mydomain.cc-2f88 is ringing May 30 19:58:57 VERBOSE[26195] logger.c: 12 headers, 0 lines May 30 19:58:57 VERBOSE[26195] logger.c: Reliably Transmitting (NAT) to 2XX.XXX.XXX.99:5060: OPTIONS sip:2XX.XXX.XXX.99 SIP/2.0 Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK438d88e0;rport From: "asterisk" ;tag=as0f935f53 To: Contact: Call-ID: 4966105a470fbae869883db628dfc76d@2XX.XXX.XXX.79 CSeq: 102 OPTIONS User-Agent: MyAsteriskServer1 Max-Forwards: 70 Date: Tue, 30 May 2006 17:58:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- May 30 19:58:57 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK438d88e0;rport=5060 From: "asterisk" ;tag=as0f935f53 To: ;tag=1dc14ad750ef67299e96c51001f386c4.9692 Call-ID: 4966105a470fbae869883db628dfc76d@2XX.XXX.XXX.79 CSeq: 102 OPTIONS Server: OpenSer (1.0.1 (i386/linux)) Content-Length: 0 Warning: 392 2XX.XXX.XXX.99:5060 "Noisy feedback tells: pid=12543 req_src_ip=2XX.XXX.XXX.79 req_src_port=5060 in_uri=sip:2XX.XXX.XXX.99 out_uri=sip:2XX.XXX.XXX.99 via_cnt=$ May 30 19:58:57 VERBOSE[26195] logger.c: --- (9 headers 0 lines)May 30 19:58:57 VERBOSE[26195] logger.c: --- (9 headers 0 lines)--- May 30 19:58:57 VERBOSE[26195] logger.c: Destroying call '4966105a470fbae869883db628dfc76d@2XX.XXX.XXX.79' May 30 19:58:57 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK0340a86c;rport=5060 From: "Unknown" ;tag=as4685a108 To: ;tag=5CBB8D43-301D1CEC CSeq: 102 INVITE Call-ID: 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 Contact: Record-Route: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Content-Type: application/sdp Content-Length: 233 v=0 o=- 1149011928 1149011928 IN IP4 10.1.99.161 s=Polycom IP Phone c=IN IP4 8X.XXX.XX.196 t=0 0 m=audio 2224 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=direction:active a=oldmediaip:10.1.99.161 May 30 19:58:57 VERBOSE[26195] logger.c: --- (12 headers 10 lines)May 30 19:58:57 VERBOSE[26195] logger.c: --- (12 headers 10 lines)--- May 30 19:58:57 VERBOSE[26195] logger.c: Found RTP audio format 0 May 30 19:58:57 VERBOSE[26195] logger.c: Found RTP audio format 101 May 30 19:58:57 VERBOSE[26195] logger.c: Peer audio RTP is at port 8X.XXX.XX.196:2224 May 30 19:58:57 VERBOSE[26195] logger.c: Found description format PCMU May 30 19:58:57 VERBOSE[26195] logger.c: Found description format telephone-event May 30 19:58:57 VERBOSE[26195] logger.c: Capabilities: us - 0x8050f (g723|gsm|ulaw|alaw|g729|ilbc|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4$ May 30 19:58:57 VERBOSE[26195] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) May 30 19:58:57 VERBOSE[26195] logger.c: Transmitting (no NAT) to 2XX.XXX.XXX.99:5060: ACK sip:190400000002301@8X.XXX.XX.196:5060 SIP/2.0 Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK7be66ad2;rport Route: From: "Unknown" ;tag=as4685a108 To: ;tag=5CBB8D43-301D1CEC Contact: Call-ID: 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 CSeq: 102 ACK User-Agent: MyAsteriskServer1 Max-Forwards: 70 Content-Length: 0 --- May 30 19:58:57 VERBOSE[23174] logger.c: -- SIP/siptest1.mydomain.cc-2f88 answered Zap/32-1 May 30 19:59:01 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: INVITE sip:Unknown@2XX.XXX.XXX.79 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 2XX.XXX.XXX.99;branch=z9hG4bK44f5.fb30ccb6.0 Via: SIP/2.0/UDP 10.1.99.161:5060;rport=5060;received=8X.XXX.XX.196;branch=z9hG4bK8e9356ee37C498DD From: ;tag=5CBB8D43-301D1CEC To: "Unknown" ;tag=as4685a108 CSeq: 1 INVITE Call-ID: 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 69 Content-Type: application/sdp Content-Length: 201 v=0 o=- 1149011929 1149011929 IN IP4 10.1.99.161 s=Polycom IP Phone c=IN IP4 0.0.0.0 t=0 0 m=audio 2224 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=direction:active May 30 19:59:01 VERBOSE[26195] logger.c: --- (16 headers 9 lines)May 30 19:59:01 VERBOSE[26195] logger.c: --- (16 headers 9 lines)--- May 30 19:59:01 VERBOSE[26195] logger.c: Using INVITE request as basis request - 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 May 30 19:59:01 VERBOSE[26195] logger.c: Sending to 2XX.XXX.XXX.99 : 5060 (non-NAT) May 30 19:59:01 VERBOSE[26195] logger.c: Found RTP audio format 0 May 30 19:59:01 VERBOSE[26195] logger.c: Found RTP audio format 101 May 30 19:59:01 VERBOSE[26195] logger.c: Peer audio RTP is at port 0.0.0.0:2224 May 30 19:59:01 VERBOSE[26195] logger.c: Found description format PCMU May 30 19:59:01 VERBOSE[26195] logger.c: Found description format telephone-event May 30 19:59:01 VERBOSE[26195] logger.c: Capabilities: us - 0x8050f (g723|gsm|ulaw|alaw|g729|ilbc|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4$ May 30 19:59:01 VERBOSE[26195] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) May 30 19:59:01 VERBOSE[26195] logger.c: -- Started music on hold, class 'default', on channel 'Zap/32-1' May 30 19:59:01 VERBOSE[26195] logger.c: We're at 2XX.XXX.XXX.79 port 12308 May 30 19:59:01 VERBOSE[26195] logger.c: Adding codec 0x4 (ulaw) to SDP May 30 19:59:01 VERBOSE[26195] logger.c: Adding non-codec 0x1 (telephone-event) to SDP May 30 19:59:01 VERBOSE[26195] logger.c: Reliably Transmitting (no NAT) to 2XX.XXX.XXX.99:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 2XX.XXX.XXX.99;branch=z9hG4bK44f5.fb30ccb6.0;received=2XX.XXX.XXX.99 Via: SIP/2.0/UDP 10.1.99.161:5060;rport=5060;received=8X.XXX.XX.196;branch=z9hG4bK8e9356ee37C498DD Record-Route: From: ;tag=5CBB8D43-301D1CEC To: "Unknown" ;tag=as4685a108 Call-ID: 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 CSeq: 1 INVITE User-Agent: MyAsteriskServer1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 216 v=0 o=root 26177 26178 IN IP4 2XX.XXX.XXX.79 s=session c=IN IP4 2XX.XXX.XXX.79 t=0 0 m=audio 12308 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- May 30 19:59:01 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: ACK sip:Unknown@2XX.XXX.XXX.79 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 2XX.XXX.XXX.99;branch=0 Via: SIP/2.0/UDP 10.1.99.161:5060;rport=5060;received=8X.XXX.XX.196;branch=z9hG4bKaa175940BEECC87 From: ;tag=5CBB8D43-301D1CEC To: "Unknown" ;tag=as4685a108 CSeq: 1 ACK Call-ID: 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Max-Forwards: 69 Content-Length: 0 May 30 19:59:01 VERBOSE[26195] logger.c: --- (13 headers 0 lines)May 30 19:59:01 VERBOSE[26195] logger.c: --- (13 headers 0 lines)--- May 30 19:59:05 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: INVITE sip:972@asterisk1.mydomain.cc:5060;user=phone SIP/2.0 Record-Route: Via: SIP/2.0/UDP 2XX.XXX.XXX.99;branch=z9hG4bK6052.bc186037.0 Via: SIP/2.0/UDP 10.1.99.161:5060;rport=5060;received=8X.XXX.XX.196;branch=z9hG4bKcfe1f9cf4469D30A From: "Test 1" ;tag=EDAB3BA8-B8CC04E5 To: CSeq: 2 INVITE Call-ID: 3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 69 Content-Type: application/sdp Content-Length: 295 v=0 o=- 1149011936 1149011936 IN IP4 10.1.99.161 s=Polycom IP Phone c=IN IP4 8X.XXX.XX.196 t=0 0 a=sendrecv m=audio 2226 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=direction:active a=oldmediaip:10.1.99.161 May 30 19:59:05 VERBOSE[26195] logger.c: --- (16 headers 13 lines)May 30 19:59:05 VERBOSE[26195] logger.c: --- (16 headers 13 lines)--- May 30 19:59:05 VERBOSE[26195] logger.c: Using INVITE request as basis request - 3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161 May 30 19:59:05 VERBOSE[26195] logger.c: Sending to 2XX.XXX.XXX.99 : 5060 (non-NAT) May 30 19:59:05 VERBOSE[26195] logger.c: Found peer '2XX.XXX.XXX.99' May 30 19:59:05 VERBOSE[26195] logger.c: Found RTP audio format 0 May 30 19:59:05 VERBOSE[26195] logger.c: Found RTP audio format 8 May 30 19:59:05 VERBOSE[26195] logger.c: Found RTP audio format 18 May 30 19:59:05 VERBOSE[26195] logger.c: Found RTP audio format 101 May 30 19:59:05 VERBOSE[26195] logger.c: Peer audio RTP is at port 8X.XXX.XX.196:2226 May 30 19:59:05 VERBOSE[26195] logger.c: Found description format PCMU May 30 19:59:05 VERBOSE[26195] logger.c: Found description format PCMA May 30 19:59:05 VERBOSE[26195] logger.c: Found description format G729 May 30 19:59:05 VERBOSE[26195] logger.c: Found description format telephone-event May 30 19:59:05 VERBOSE[26195] logger.c: Capabilities: us - 0x8050f (g723|gsm|ulaw|alaw|g729|ilbc|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), co$ May 30 19:59:05 VERBOSE[26195] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) May 30 19:59:05 VERBOSE[26195] logger.c: Looking for 972 in from-internal (domain asterisk1.mydomain.cc) May 30 19:59:05 VERBOSE[26195] logger.c: list_route: hop: May 30 19:59:05 VERBOSE[26195] logger.c: Transmitting (NAT) to 2XX.XXX.XXX.99:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 2XX.XXX.XXX.99;branch=z9hG4bK6052.bc186037.0;received=2XX.XXX.XXX.99 Via: SIP/2.0/UDP 10.1.99.161:5060;rport=5060;received=8X.XXX.XX.196;branch=z9hG4bKcfe1f9cf4469D30A From: "Test 1" ;tag=EDAB3BA8-B8CC04E5 To: Call-ID: 3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161 CSeq: 2 INVITE User-Agent: MyAsteriskServer1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- May 30 19:59:05 VERBOSE[23209] logger.c: -- Executing Set("SIP/siptest1.mydomain.cc-0a003c10", "CDR(accountcode)=972") in new stack May 30 19:59:05 VERBOSE[23209] logger.c: -- Executing Dial("SIP/siptest1.mydomain.cc-0a003c10", "SIP/190400000002801@siptest1.mydomain.cc") in new stack May 30 19:59:05 VERBOSE[23209] logger.c: -- parse_srv: SRV mapped to host siptest1.mydomain.cc, port 5060 May 30 19:59:05 VERBOSE[23209] logger.c: We're at 2XX.XXX.XXX.79 port 14014 May 30 19:59:05 VERBOSE[23209] logger.c: Adding codec 0x4 (ulaw) to SDP May 30 19:59:05 VERBOSE[23209] logger.c: Adding codec 0x100 (g729) to SDP May 30 19:59:05 VERBOSE[23209] logger.c: Adding codec 0x1 (g723) to SDP May 30 19:59:05 VERBOSE[23209] logger.c: Adding codec 0x2 (gsm) to SDP May 30 19:59:05 VERBOSE[23209] logger.c: Adding codec 0x400 (ilbc) to SDP May 30 19:59:05 VERBOSE[23209] logger.c: Adding codec 0x8 (alaw) to SDP May 30 19:59:05 VERBOSE[23209] logger.c: Adding non-codec 0x1 (telephone-event) to SDP May 30 19:59:05 VERBOSE[23209] logger.c: 13 headers, 16 lines May 30 19:59:05 VERBOSE[23209] logger.c: Reliably Transmitting (no NAT) to 2XX.XXX.XXX.99:5060: INVITE sip:190400000002801@siptest1.mydomain.cc SIP/2.0 Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK750b4118;rport From: "Test 1" ;tag=as6fd59beb To: Contact: Call-ID: 44d0a90b505938b1659b05b22c20a6ba@2XX.XXX.XXX.79 CSeq: 102 INVITE User-Agent: MyAsteriskServer1 Max-Forwards: 70 Date: Tue, 30 May 2006 17:59:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 360 v=0 o=root 26177 26177 IN IP4 2XX.XXX.XXX.79 s=session c=IN IP4 2XX.XXX.XXX.79 t=0 0 m=audio 14014 RTP/AVP 0 18 4 3 97 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- May 30 19:59:05 VERBOSE[23209] logger.c: -- Called 190400000002801@siptest1.mydomain.cc May 30 19:59:05 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK750b4118;rport=5060 From: "Test 1" ;tag=as6fd59beb To: Call-ID: 44d0a90b505938b1659b05b22c20a6ba@2XX.XXX.XXX.79 CSeq: 102 INVITE Server: OpenSer (1.0.1 (i386/linux)) Content-Length: 0 Warning: 392 2XX.XXX.XXX.99:5060 "Noisy feedback tells: pid=12537 req_src_ip=2XX.XXX.XXX.79 req_src_port=5060 in_uri=sip:190400000002801@siptest1.mydomain.cc out_uri=sip:$ May 30 19:59:05 VERBOSE[26195] logger.c: --- (9 headers 0 lines)May 30 19:59:05 VERBOSE[26195] logger.c: --- (9 headers 0 lines)--- May 30 19:59:05 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK750b4118;rport=5060 Record-Route: From: "Test 1" ;tag=as6fd59beb To: ;tag=n6ypitbvai Call-ID: 44d0a90b505938b1659b05b22c20a6ba@2XX.XXX.XXX.79 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 May 30 19:59:05 VERBOSE[26195] logger.c: --- (11 headers 0 lines)May 30 19:59:05 VERBOSE[26195] logger.c: --- (11 headers 0 lines)--- May 30 19:59:05 VERBOSE[23209] logger.c: -- SIP/siptest1.mydomain.cc-9a28 is ringing May 30 19:59:05 VERBOSE[23209] logger.c: Transmitting (NAT) to 2XX.XXX.XXX.99:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2XX.XXX.XXX.99;branch=z9hG4bK6052.bc186037.0;received=2XX.XXX.XXX.99 Via: SIP/2.0/UDP 10.1.99.161:5060;rport=5060;received=8X.XXX.XX.196;branch=z9hG4bKcfe1f9cf4469D30A From: "Test 1" ;tag=EDAB3BA8-B8CC04E5 To: ;tag=as375248fe Call-ID: 3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161 CSeq: 2 INVITE User-Agent: MyAsteriskServer1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- May 30 19:59:05 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK750b4118;rport=5060 From: "Test 1" ;tag=as6fd59beb To: ;tag=88251B10-C8D161ED CSeq: 102 INVITE Call-ID: 44d0a90b505938b1659b05b22c20a6ba@2XX.XXX.XXX.79 Contact: Record-Route: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Allow-Events: talk,hold,conference Content-Length: 0 May 30 19:59:05 VERBOSE[26195] logger.c: --- (11 headers 0 lines)May 30 19:59:05 VERBOSE[26195] logger.c: --- (11 headers 0 lines)--- May 30 19:59:05 VERBOSE[23209] logger.c: -- SIP/siptest1.mydomain.cc-9a28 is ringing May 30 19:59:06 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK750b4118;rport=5060 Record-Route: From: "Test 1" ;tag=as6fd59beb To: ;tag=n6ypitbvai Call-ID: 44d0a90b505938b1659b05b22c20a6ba@2XX.XXX.XXX.79 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 May 30 19:59:06 VERBOSE[26195] logger.c: --- (11 headers 0 lines)May 30 19:59:06 VERBOSE[26195] logger.c: --- (11 headers 0 lines)--- May 30 19:59:06 VERBOSE[23209] logger.c: -- SIP/siptest1.mydomain.cc-9a28 is ringing May 30 19:59:07 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK750b4118;rport=5060 Record-Route: From: "Test 1" ;tag=as6fd59beb To: ;tag=n6ypitbvai Call-ID: 44d0a90b505938b1659b05b22c20a6ba@2XX.XXX.XXX.79 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 May 30 19:59:07 VERBOSE[26195] logger.c: --- (11 headers 0 lines)May 30 19:59:07 VERBOSE[26195] logger.c: --- (11 headers 0 lines)--- May 30 19:59:07 VERBOSE[23209] logger.c: -- SIP/siptest1.mydomain.cc-9a28 is ringing May 30 19:59:09 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK750b4118;rport=5060 Record-Route: From: "Test 1" ;tag=as6fd59beb To: ;tag=n6ypitbvai Call-ID: 44d0a90b505938b1659b05b22c20a6ba@2XX.XXX.XXX.79 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 May 30 19:59:09 VERBOSE[26195] logger.c: --- (11 headers 0 lines)May 30 19:59:09 VERBOSE[26195] logger.c: --- (11 headers 0 lines)--- May 30 19:59:09 VERBOSE[23209] logger.c: -- SIP/siptest1.mydomain.cc-9a28 is ringing May 30 19:59:09 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK750b4118;rport=5060 Record-Route: From: "Test 1" ;tag=as6fd59beb To: ;tag=n6ypitbvai Call-ID: 44d0a90b505938b1659b05b22c20a6ba@2XX.XXX.XXX.79 CSeq: 102 INVITE Contact: User-Agent: snom190/3.60w 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: 254 v=0 o=root 210869429 210869430 IN IP4 10.1.99.169 s=call c=IN IP4 8X.XXX.XX.196 t=0 0 m=audio 54700 RTP/AVP 18 101 a=rtpmap:18 g729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=direction:active a=oldmediaip:10.1.99.169 May 30 19:59:09 VERBOSE[26195] logger.c: --- (14 headers 12 lines)May 30 19:59:09 VERBOSE[26195] logger.c: --- (14 headers 12 lines)--- May 30 19:59:09 VERBOSE[26195] logger.c: Found RTP audio format 18 May 30 19:59:09 VERBOSE[26195] logger.c: Found RTP audio format 101 May 30 19:59:09 VERBOSE[26195] logger.c: Peer audio RTP is at port 8X.XXX.XX.196:54700 May 30 19:59:09 VERBOSE[26195] logger.c: Found description format g729 May 30 19:59:09 VERBOSE[26195] logger.c: Found description format telephone-event May 30 19:59:09 VERBOSE[26195] logger.c: Capabilities: us - 0x8050f (g723|gsm|ulaw|alaw|g729|ilbc|h263), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0$ May 30 19:59:09 VERBOSE[26195] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) May 30 19:59:09 VERBOSE[26195] logger.c: Transmitting (no NAT) to 2XX.XXX.XXX.99:5060: ACK sip:190400000002801@8X.XXX.XX.196:2091;line=pa523as2 SIP/2.0 Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK5665862c;rport Route: From: "Test 1" ;tag=as6fd59beb To: ;tag=n6ypitbvai Contact: Call-ID: 44d0a90b505938b1659b05b22c20a6ba@2XX.XXX.XXX.79 CSeq: 102 ACK User-Agent: MyAsteriskServer1 Max-Forwards: 70 Content-Length: 0 --- May 30 19:59:09 VERBOSE[23209] logger.c: -- SIP/siptest1.mydomain.cc-9a28 answered SIP/siptest1.mydomain.cc-0a003c10 May 30 19:59:09 VERBOSE[23209] logger.c: We're at 2XX.XXX.XXX.79 port 14866 May 30 19:59:09 VERBOSE[23209] logger.c: Adding codec 0x4 (ulaw) to SDP May 30 19:59:09 VERBOSE[23209] logger.c: Adding codec 0x100 (g729) to SDP May 30 19:59:09 VERBOSE[23209] logger.c: Adding codec 0x8 (alaw) to SDP May 30 19:59:09 VERBOSE[23209] logger.c: Adding non-codec 0x1 (telephone-event) to SDP May 30 19:59:09 VERBOSE[23209] logger.c: Reliably Transmitting (NAT) to 2XX.XXX.XXX.99:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 2XX.XXX.XXX.99;branch=z9hG4bK6052.bc186037.0;received=2XX.XXX.XXX.99 Via: SIP/2.0/UDP 10.1.99.161:5060;rport=5060;received=8X.XXX.XX.196;branch=z9hG4bKcfe1f9cf4469D30A Record-Route: From: "Test 1" ;tag=EDAB3BA8-B8CC04E5 To: ;tag=as375248fe Call-ID: 3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161 CSeq: 2 INVITE User-Agent: MyAsteriskServer1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 26177 26177 IN IP4 2XX.XXX.XXX.79 s=session c=IN IP4 2XX.XXX.XXX.79 t=0 0 m=audio 14866 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- May 30 19:59:09 VERBOSE[23209] logger.c: -- Attempting native bridge of SIP/siptest1.mydomain.cc-0a003c10 and SIP/siptest1.mydomain.cc-9a28 May 30 19:59:09 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: ACK sip:972@2XX.XXX.XXX.79:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 2XX.XXX.XXX.99;branch=0 Via: SIP/2.0/UDP 10.1.99.161:5060;rport=5060;received=8X.XXX.XX.196;branch=z9hG4bKbb1cc0174A67FAB2 From: "Test 1" ;tag=EDAB3BA8-B8CC04E5 To: ;tag=as375248fe CSeq: 2 ACK Call-ID: 3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Max-Forwards: 69 Content-Length: 0 May 30 19:59:09 VERBOSE[26195] logger.c: --- (13 headers 0 lines)May 30 19:59:09 VERBOSE[26195] logger.c: --- (13 headers 0 lines)--- May 30 19:59:12 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: REFER sip:Unknown@2XX.XXX.XXX.79 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 2XX.XXX.XXX.99;branch=z9hG4bK14f5.f2e67207.0 Via: SIP/2.0/UDP 10.1.99.161:5060;rport=5060;received=8X.XXX.XX.196;branch=z9hG4bK3771ecd18DF803A4 From: ;tag=5CBB8D43-301D1CEC To: "Unknown" ;tag=as4685a108 CSeq: 2 REFER Call-ID: 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Refer-To: Referred-By: Max-Forwards: 69 Content-Length: 0 May 30 19:59:12 VERBOSE[26195] logger.c: --- (14 headers 0 lines)May 30 19:59:12 VERBOSE[26195] logger.c: --- (14 headers 0 lines)--- May 30 19:59:12 VERBOSE[26195] logger.c: Transfer to 972 in default May 30 19:59:12 VERBOSE[26195] logger.c: Transfer from 190400000002301 in default May 30 19:59:12 VERBOSE[26195] logger.c: -- Stopped music on hold on Zap/32-1 May 30 19:59:12 VERBOSE[26195] logger.c: Transmitting (no NAT) to 2XX.XXX.XXX.99:5060: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 2XX.XXX.XXX.99;branch=z9hG4bK14f5.f2e67207.0;received=2XX.XXX.XXX.99 Via: SIP/2.0/UDP 10.1.99.161:5060;rport=5060;received=8X.XXX.XX.196;branch=z9hG4bK3771ecd18DF803A4 Record-Route: From: ;tag=5CBB8D43-301D1CEC To: "Unknown" ;tag=as4685a108 Call-ID: 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 CSeq: 2 REFER User-Agent: MyAsteriskServer1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- May 30 19:59:12 VERBOSE[26195] logger.c: set_destination: Parsing for address/port to send to May 30 19:59:12 VERBOSE[26195] logger.c: set_destination: set destination to 2XX.XXX.XXX.99, port 5060 May 30 19:59:12 VERBOSE[26195] logger.c: Reliably Transmitting (no NAT) to 2XX.XXX.XXX.99:5060: NOTIFY sip:190400000002301@8X.XXX.XX.196:5060 SIP/2.0 Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK7efa0ab2;rport Route: From: "Unknown" ;tag=as4685a108 To: ;tag=5CBB8D43-301D1CEC Contact: Call-ID: 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 CSeq: 103 NOTIFY User-Agent: MyAsteriskServer1 Max-Forwards: 70 Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- May 30 19:59:12 VERBOSE[23209] logger.c: set_destination: Parsing for address/port to send to May 30 19:59:12 VERBOSE[23209] logger.c: set_destination: set destination to 2XX.XXX.XXX.99, port 5060 May 30 19:59:12 VERBOSE[26195] logger.c: set_destination: Parsing for address/port to send to May 30 19:59:12 VERBOSE[23209] logger.c: Reliably Transmitting (NAT) to 2XX.XXX.XXX.99:5060: BYE sip:190400000002301@8X.XXX.XX.196:5060 SIP/2.0 Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK05f70a67;rport Route: From: ;tag=as375248fe To: "Test 1" ;tag=EDAB3BA8-B8CC04E5 Contact: Call-ID: 3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161 CSeq: 102 BYE User-Agent: MyAsteriskServer1 Max-Forwards: 70 Content-Length: 0 --- May 30 19:59:12 VERBOSE[26195] logger.c: set_destination: set destination to 2XX.XXX.XXX.99, port 5060 May 30 19:59:12 VERBOSE[26195] logger.c: Reliably Transmitting (no NAT) to 2XX.XXX.XXX.99:5060: BYE sip:190400000002301@8X.XXX.XX.196:5060 SIP/2.0 Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK7b440ff8;rport Route: From: "Unknown" ;tag=as4685a108 To: ;tag=5CBB8D43-301D1CEC Contact: Call-ID: 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 CSeq: 104 BYE User-Agent: MyAsteriskServer1 Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing Content-Length: 0 --- May 30 19:59:12 VERBOSE[23174] logger.c: == Spawn extension (asteriskfunctions, 971, 2) exited non-zero on 'SIP/siptest1.mydomain.cc-0a003c10' May 30 19:59:12 VERBOSE[26195] logger.c: Retransmitting #1 (NAT) to 2XX.XXX.XXX.99:5060: BYE sip:190400000002301@8X.XXX.XX.196:5060 SIP/2.0 Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK05f70a67;rport Route: From: ;tag=as375248fe To: "Test 1" ;tag=EDAB3BA8-B8CC04E5 Contact: Call-ID: 3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161 CSeq: 102 BYE User-Agent: MyAsteriskServer1 Max-Forwards: 70 Content-Length: 0 --- May 30 19:59:12 VERBOSE[26195] logger.c: Retransmitting #2 (NAT) to 2XX.XXX.XXX.99:5060: BYE sip:190400000002301@8X.XXX.XX.196:5060 SIP/2.0 Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK05f70a67;rport Route: From: ;tag=as375248fe To: "Test 1" ;tag=EDAB3BA8-B8CC04E5 Contact: Call-ID: 3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161 CSeq: 102 BYEBYE sip:190400000002301@8X.XXX.XX.196:5060 SIP/2.0 Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK05f70a67;rport Route: From: ;tag=as375248fe To: "Test 1" ;tag=EDAB3BA8-B8CC04E5 Contact: Call-ID: 3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161 CSeq: 102 BYE User-Agent: MyAsteriskServer1 Max-Forwards: 70 Content-Length: 0 --- User-Agent: MyAsteriskServer1 Max-Forwards: 70 Content-Length: 0 --- May 30 19:59:12 VERBOSE[26195] logger.c: Retransmitting #3 (NAT) to 2XX.XXX.XXX.99:5060: BYE sip:190400000002301@8X.XXX.XX.196:5060 SIP/2.0 Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK05f70a67;rport Route: From: ;tag=as375248fe To: "Test 1" ;tag=EDAB3BA8-B8CC04E5 Contact: Call-ID: 3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161 CSeq: 102 BYE User-Agent: MyAsteriskServer1 Max-Forwards: 70 Content-Length: 0 --- May 30 19:59:12 VERBOSE[26195] logger.c: Retransmitting #4 (NAT) to 2XX.XXX.XXX.99:5060: BYE sip:190400000002301@8X.XXX.XX.196:5060 SIP/2.0 Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK05f70a67;rport Route: From: ;tag=as375248fe To: "Test 1" ;tag=EDAB3BA8-B8CC04E5 Contact: Call-ID: 3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161 CSeq: 102 BYE User-Agent: MyAsteriskServer1 Max-Forwards: 70 Content-Length: 0 --- May 30 19:59:12 VERBOSE[26195] logger.c: Retransmitting #5 (NAT) to 2XX.XXX.XXX.99:5060: BYE sip:190400000002301@8X.XXX.XX.196:5060 SIP/2.0 Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK05f70a67;rport Route: From: ;tag=as375248fe To: "Test 1" ;tag=EDAB3BA8-B8CC04E5 Contact: Call-ID: 3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161 CSeq: 102 BYE User-Agent: MyAsteriskServer1 Max-Forwards: 70 Content-Length: 0 --- May 30 19:59:12 VERBOSE[26195] logger.c: Retransmitting #6 (NAT) to 2XX.XXX.XXX.99:5060: BYE sip:190400000002301@8X.XXX.XX.196:5060 SIP/2.0 Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK05f70a67;rport Route: From: ;tag=as375248fe To: "Test 1" ;tag=EDAB3BA8-B8CC04E5 Contact: Call-ID: 3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161 CSeq: 102 BYE User-Agent: MyAsteriskServer1 Max-Forwards: 70 Content-Length: 0 --- May 30 19:59:12 VERBOSE[26195] logger.c: Destroying call '3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161' May 30 19:59:12 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK7efa0ab2;rport=5060 From: "Unknown" ;tag=as4685a108 To: ;tag=5CBB8D43-301D1CEC CSeq: 103 NOTIFY Call-ID: 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 Contact: Record-Route: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Content-Length: 0 May 30 19:59:12 VERBOSE[26195] logger.c: --- (11 headers 0 lines)May 30 19:59:12 VERBOSE[26195] logger.c: --- (11 headers 0 lines)--- May 30 19:59:12 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: BYE sip:Unknown@2XX.XXX.XXX.79 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 2XX.XXX.XXX.99;branch=z9hG4bK24f5.76cb2e92.0 Via: SIP/2.0/UDP 10.1.99.161:5060;rport=5060;received=8X.XXX.XX.196;branch=z9hG4bKf674ee9b951C6A66 From: ;tag=5CBB8D43-301D1CEC To: "Unknown" ;tag=as4685a108 CSeq: 3 BYE Call-ID: 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Max-Forwards: 69 Content-Length: 0 May 30 19:59:12 VERBOSE[26195] logger.c: --- (12 headers 0 lines)May 30 19:59:12 VERBOSE[26195] logger.c: --- (12 headers 0 lines)--- May 30 19:59:12 VERBOSE[26195] logger.c: Sending to 2XX.XXX.XXX.99 : 5060 (non-NAT) May 30 19:59:12 VERBOSE[26195] logger.c: Transmitting (no NAT) to 2XX.XXX.XXX.99:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 2XX.XXX.XXX.99;branch=z9hG4bK24f5.76cb2e92.0;received=2XX.XXX.XXX.99 Via: SIP/2.0/UDP 10.1.99.161:5060;rport=5060;received=8X.XXX.XX.196;branch=z9hG4bKf674ee9b951C6A66 Record-Route: From: ;tag=5CBB8D43-301D1CEC To: "Unknown" ;tag=as4685a108 Call-ID: 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 CSeq: 3 BYE User-Agent: MyAsteriskServer1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- May 30 19:59:12 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK05f70a67;rport=5060 From: ;tag=as375248fe To: "Test 1" ;tag=EDAB3BA8-B8CC04E5 CSeq: 102 BYE Call-ID: 3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161 Contact: Record-Route: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Content-Length: 0 May 30 19:59:12 VERBOSE[26195] logger.c: --- (10 headers 0 lines)May 30 19:59:12 VERBOSE[26195] logger.c: --- (10 headers 0 lines)--- May 30 19:59:12 VERBOSE[26195] logger.c: Destroying call '3e78dcd4-ff6c7316-2b7d1a0b@10.1.99.161' May 30 19:59:13 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK7b440ff8;rport=5060 From: "Unknown" ;tag=as4685a108 To: ;tag=5CBB8D43-301D1CEC CSeq: 104 BYE Call-ID: 20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79 Contact: Record-Route: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Content-Length: 0 May 30 19:59:13 VERBOSE[26195] logger.c: --- (10 headers 0 lines)May 30 19:59:13 VERBOSE[26195] logger.c: --- (10 headers 0 lines)--- May 30 19:59:13 VERBOSE[26195] logger.c: -- Incoming call: Got SIP response 500 "Internal Server Error" back from 2XX.XXX.XXX.99 May 30 19:59:13 VERBOSE[26195] logger.c: Destroying call '20cdf1004a7849a936295b4101d27477@2XX.XXX.XXX.79' BYE sip:190400000002801@8X.XXX.XX.196:2091;line=pa523as2 SIP/2.0 Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK7f32c635;rport Route: From: "Test 1" ;tag=as6fd59beb To: ;tag=n6ypitbvai Contact: Call-ID: 44d0a90b505938b1659b05b22c20a6ba@2XX.XXX.XXX.79 CSeq: 103 BYE User-Agent: MyAsteriskServer1 Max-Forwards: 70 Content-Length: 0 --- May 30 19:59:16 VERBOSE[23209] logger.c: == Spawn extension (asteriskfunctions, 972, 2) exited non-zero on 'Zap/32-1' May 30 19:59:16 VERBOSE[23209] logger.c: -- Hungup 'Zap/32-1' May 30 19:59:16 VERBOSE[26195] logger.c: <-- SIP read from 2XX.XXX.XXX.99:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 2XX.XXX.XXX.79:5060;branch=z9hG4bK7f32c635;rport=5060 Record-Route: From: "Test 1" ;tag=as6fd59beb To: ;tag=n6ypitbvai Call-ID: 44d0a90b505938b1659b05b22c20a6ba@2XX.XXX.XXX.79 CSeq: 103 BYE Contact: User-Agent: snom190/3.60w Content-Length: 0 May 30 19:59:16 VERBOSE[26195] logger.c: --- (10 headers 0 lines)May 30 19:59:16 VERBOSE[26195] logger.c: --- (10 headers 0 lines)--- May 30 19:59:16 VERBOSE[26195] logger.c: Destroying call '44d0a90b505938b1659b05b22c20a6ba@2XX.XXX.XXX.79'