Tony Yu Posted March 16, 2012 Report Share Posted March 16, 2012 Hi there, We are having intermittent issues where we drop calls. Sometimes after a dropped call I get an email saying User disconnects call in the subject. Email below. Also it seems that we cannot get past the 11 minute mark while on the phone. The call gets disconnected and the phone has to be rebooted in order to make more calls. One side of the call between sip:15097207844@localhost;user=phone and sip:402@localhost did not receive media for 15.6 s and the other side of the call disconnected the call. The address of the other side was 66.23.129.253 (User-Agent=ENSR3.0.66.0-IS1-RMRG40340-RG2190-CPO3036). You may use this email as hint for a potential problem. The SIP messages are attached. 2012/2/27 14:50:29 Tx: tcp:192.168.1.10:53201 (281 bytes) SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-eoaf8rzrzc70;rport=53201 From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq To: <sip:15097207844@localhost;user=phone>;tag=4f001275af Call-ID: 73de4b4f0797-xa67qm6rn81r CSeq: 1 INVITE Content-Length: 0 2012/2/27 14:50:29 Tx: tcp:192.168.1.10:53201 (495 bytes) SIP/2.0 401 Authentication Required Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-eoaf8rzrzc70;rport=53201 From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq To: <sip:15097207844@localhost;user=phone>;tag=4f001275af Call-ID: 73de4b4f0797-xa67qm6rn81r CSeq: 1 INVITE User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids WWW-Authenticate: Digest realm="localhost",nonce="49a2d4d203befa6ee2e15b672eed4617",domain="sip:15097207844@localhost;user=phone",algorithm=MD5 Content-Length: 0 2012/2/27 14:50:30 Rx: tcp:192.168.1.10:53201 (420 bytes) ACK sip:15097207844@localhost;user=phone SIP/2.0 Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-eoaf8rzrzc70;rport From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq To: <sip:15097207844@localhost;user=phone>;tag=4f001275af Call-ID: 73de4b4f0797-xa67qm6rn81r CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:402@192.168.1.10:53201;transport=tcp;line=uuebwkg5>;reg-id=1 Proxy-Require: buttons Content-Length: 0 2012/2/27 14:50:30 Rx: tcp:192.168.1.10:53201 (1742 bytes) INVITE sip:15097207844@localhost;user=phone SIP/2.0 Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-fslucigg78pa;rport From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq To: <sip:15097207844@localhost;user=phone> Call-ID: 73de4b4f0797-xa67qm6rn81r CSeq: 2 INVITE Max-Forwards: 70 Contact: <sip:402@192.168.1.10:53201;transport=tcp;line=uuebwkg5>;reg-id=1 X-Serialnumber: 00041371061C P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom760/8.7.2.9 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Authorization: Digest username="402",realm="localhost",nonce="49a2d4d203befa6ee2e15b672eed4617",uri="sip:15097207844@localhost;user=phone",response="aaa9202c6872ccdb974a356b26449e79",algorithm=MD5 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 733 v=0 o=root 118993245 118993245 IN IP4 192.168.1.10 s=call c=IN IP4 192.168.1.10 t=0 0 m=audio 59760 RTP/AVP 9 0 8 3 97 98 99 100 106 107 108 109 18 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:8agvdPvzMo+D6DHUsQ9dT9t/rYStIyTEPtrHpab4 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 G726-16/8000 a=rtpmap:98 G726-24/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:100 G726-40/8000 a=rtpmap:106 AAL2-G726-16/8000 a=rtpmap:107 AAL2-G726-24/8000 a=rtpmap:108 AAL2-G726-32/8000 a=rtpmap:109 AAL2-G726-40/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv 2012/2/27 14:50:30 Tx: tcp:192.168.1.10:53201 (281 bytes) SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-fslucigg78pa;rport=53201 From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq To: <sip:15097207844@localhost;user=phone>;tag=4f001275af Call-ID: 73de4b4f0797-xa67qm6rn81r CSeq: 2 INVITE Content-Length: 0 2012/2/27 14:50:30 Tx: tcp:192.168.1.10:53201 (900 bytes) SIP/2.0 183 Session Progress Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-fslucigg78pa;rport=53201 From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq To: <sip:15097207844@localhost;user=phone>;tag=4f001275af Call-ID: 73de4b4f0797-xa67qm6rn81r CSeq: 2 INVITE Contact: <sip:402@192.168.1.26:5060;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 298 v=0 o=- 1271747016 1271747016 IN IP4 192.168.1.26 s=- c=IN IP4 192.168.1.26 t=0 0 m=audio 56720 RTP/AVP 0 18 101 a=rtpmap:0 pcmu/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv 2012/2/27 14:50:30 Rx: tcp:192.168.1.10:53201 (587 bytes) PRACK sip:402@192.168.1.26:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-851e4fsca841;rport From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq To: <sip:15097207844@localhost;user=phone>;tag=4f001275af Call-ID: 73de4b4f0797-xa67qm6rn81r CSeq: 3 PRACK Max-Forwards: 70 Contact: <sip:402@192.168.1.10:53201;transport=tcp;line=uuebwkg5>;reg-id=1 RAck: 1 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Proxy-Require: buttons Content-Length: 0 2012/2/27 14:50:30 Tx: tcp:192.168.1.10:53201 (380 bytes) SIP/2.0 200 Ok Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-851e4fsca841;rport=53201 From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq To: <sip:15097207844@localhost;user=phone>;tag=4f001275af Call-ID: 73de4b4f0797-xa67qm6rn81r CSeq: 3 PRACK Contact: <sip:402@192.168.1.26:5060;transport=tcp> User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids Content-Length: 0 2012/2/27 14:50:42 Tx: tcp:192.168.1.10:53201 (860 bytes) SIP/2.0 200 Ok Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-fslucigg78pa;rport=53201 From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq To: <sip:15097207844@localhost;user=phone>;tag=4f001275af Call-ID: 73de4b4f0797-xa67qm6rn81r CSeq: 2 INVITE Contact: <sip:402@192.168.1.26:5060;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids Content-Type: application/sdp Content-Length: 298 v=0 o=- 1271747016 1271747016 IN IP4 192.168.1.26 s=- c=IN IP4 192.168.1.26 t=0 0 m=audio 56720 RTP/AVP 0 18 101 a=rtpmap:0 pcmu/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv 2012/2/27 14:50:42 Rx: tcp:192.168.1.10:53201 (423 bytes) ACK sip:402@192.168.1.26:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-o52xku6si97r;rport From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq To: <sip:15097207844@localhost;user=phone>;tag=4f001275af Call-ID: 73de4b4f0797-xa67qm6rn81r CSeq: 2 ACK Max-Forwards: 70 Contact: <sip:402@192.168.1.10:53201;transport=tcp;line=uuebwkg5>;reg-id=1 Proxy-Require: buttons Content-Length: 0 2012/2/27 14:51:03 Rx: tcp:192.168.1.10:53201 (597 bytes) BYE sip:402@192.168.1.26:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-fsyo66d8nyme;rport From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq To: <sip:15097207844@localhost;user=phone>;tag=4f001275af Call-ID: 73de4b4f0797-xa67qm6rn81r CSeq: 4 BYE Max-Forwards: 70 Contact: <sip:402@192.168.1.10:53201;transport=tcp;line=uuebwkg5>;reg-id=1 User-Agent: snom760/8.7.2.9 RTP-RxStat: Total_Rx_Pkts=894,Rx-Pkts=891,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=1663,Tx_Pkts=1663,Remote_Tx_Pkts=1663 Proxy-Require: buttons Content-Length: 0 2012/2/27 14:51:03 Tx: tcp:192.168.1.10:53201 (378 bytes) SIP/2.0 200 Ok Via: SIP/2.0/TCP 192.168.1.10:53201;branch=z9hG4bK-fsyo66d8nyme;rport=53201 From: "Tony Yu" <sip:402@localhost>;tag=s3o3cgfajq To: <sip:15097207844@localhost;user=phone>;tag=4f001275af Call-ID: 73de4b4f0797-xa67qm6rn81r CSeq: 4 BYE Contact: <sip:402@192.168.1.26:5060;transport=tcp> User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids Content-Length: 0 ============================== 2012/2/27 14:50:30 Tx: udp:66.23.129.253:5060 (1050 bytes) INVITE sip:5097207844@nexvortex.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK-8331bd4cc3c22de864eacf37e1fa6244;rport From: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872 To: <sip:5097207844@localhost;user=phone> Call-ID: 83be7b59@pbx CSeq: 31301 INVITE Max-Forwards: 70 Contact: <sip:Pe8345WcBd@192.168.1.26:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids P-Asserted-Identity: <sip:8775784777> Privacy: id P-Charging-Vector: icid-value=;icid-generated-at=192.168.1.26;orig-ioi=localhost Content-Type: application/sdp Content-Length: 284 v=0 o=- 986850866 986850866 IN IP4 192.168.1.26 s=- c=IN IP4 192.168.1.26 t=0 0 m=audio 55236 RTP/AVP 18 0 101 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv 2012/2/27 14:50:30 Rx: udp:66.23.129.253:5060 (480 bytes) SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK-8331bd4cc3c22de864eacf37e1fa6244;rport=5060 From: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872 To: <sip:5097207844@localhost;user=phone>;tag=9c6a9fdfd4d16ebaa52f34c4c528cbe5.63ba Call-ID: 83be7b59@pbx CSeq: 31301 INVITE Proxy-Authenticate: Digest realm="nexvortex.com", nonce="4f4bdfa54d8a9eae60d6a580a08a1cf71ac78015" Server: nVSIP 12.02.01 Content-Length: 0 2012/2/27 14:50:30 Tx: udp:66.23.129.253:5060 (1276 bytes) INVITE sip:5097207844@nexvortex.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK-97eadd1d07a98b29848791af7b631541;rport From: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872 To: <sip:5097207844@localhost;user=phone> Call-ID: 83be7b59@pbx CSeq: 31302 INVITE Max-Forwards: 70 Contact: <sip:Pe8345WcBd@192.168.1.26:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids P-Asserted-Identity: <sip:8775784777> Privacy: id P-Charging-Vector: icid-value=;icid-generated-at=192.168.1.26;orig-ioi=localhost Proxy-Authorization: Digest realm="nexvortex.com",nonce="4f4bdfa54d8a9eae60d6a580a08a1cf71ac78015",response="dc5f48a93ac0c225710f334d4d7ff85e",username="Pe8345WcBd",uri="sip:5097207844@nexvortex.com;user=phone",algorithm=MD5 Content-Type: application/sdp Content-Length: 284 v=0 o=- 986850866 986850866 IN IP4 192.168.1.26 s=- c=IN IP4 192.168.1.26 t=0 0 m=audio 55236 RTP/AVP 18 0 101 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv 2012/2/27 14:50:30 Rx: udp:66.23.129.253:5060 (347 bytes) SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK-97eadd1d07a98b29848791af7b631541;rport=5060 From: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872 To: <sip:5097207844@localhost;user=phone> Call-ID: 83be7b59@pbx CSeq: 31302 INVITE Server: nVSIP 12.02.01 Content-Length: 0 2012/2/27 14:50:33 Rx: udp:66.23.129.253:5060 (979 bytes) SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK-97eadd1d07a98b29848791af7b631541;rport=5060 From: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872 To: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001 Call-ID: 83be7b59@pbx CSeq: 31302 INVITE Record-Route: <sip:206.80.67.68;lr=on;ftag=327783872;osp-o=t5713901616956271617_s66.23.129.253:5060_a1330372217_c4> Record-Route: <sip:5097207844@66.23.129.253:5060;nat=yes;ftag=327783872;lr=on> Contact: <sip:15097207844@65.98.234.158:5060;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,INFO Content-Type: application/sdp User-Agent: ENSR3.0.66.0-IS1-RMRG40340-RG2190-CPO3036 Content-Length: 223 v=0 o=- 1016715875 1016715875 IN IP4 65.98.234.158 s=ENSResip c=IN IP4 65.98.234.158 t=0 0 m=audio 54878 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv 2012/2/27 14:50:42 Rx: udp:66.23.129.253:5060 (965 bytes) SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK-97eadd1d07a98b29848791af7b631541;rport=5060 From: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872 To: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001 Call-ID: 83be7b59@pbx CSeq: 31302 INVITE Record-Route: <sip:206.80.67.68;lr=on;ftag=327783872;osp-o=t5713901616956271617_s66.23.129.253:5060_a1330372217_c4> Record-Route: <sip:5097207844@66.23.129.253:5060;nat=yes;ftag=327783872;lr=on> Contact: <sip:15097207844@65.98.234.158:5060;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,INFO Content-Type: application/sdp User-Agent: ENSR3.0.66.0-IS1-RMRG40340-RG2190-CPO3036 Content-Length: 223 v=0 o=- 1016715875 1016715875 IN IP4 65.98.234.158 s=ENSResip c=IN IP4 65.98.234.158 t=0 0 m=audio 54878 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv 2012/2/27 14:50:42 Tx: udp:66.23.129.253:5060 (756 bytes) ACK sip:15097207844@65.98.234.158:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK-64b09f91d60aa72f0f47543dc52c2990;rport Route: <sip:5097207844@66.23.129.253:5060;nat=yes;ftag=327783872;lr=on> Route: <sip:206.80.67.68;lr=on;ftag=327783872;osp-o=t5713901616956271617_s66.23.129.253:5060_a1330372217_c4> From: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872 To: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001 Call-ID: 83be7b59@pbx CSeq: 31302 ACK Max-Forwards: 70 Contact: <sip:Pe8345WcBd@192.168.1.26:5060;transport=udp> P-Asserted-Identity: <sip:8775784777> Privacy: id P-Charging-Vector: icid-value=;icid-generated-at=192.168.1.26;orig-ioi=localhost Content-Length: 0 2012/2/27 14:50:43 Rx: udp:66.23.129.253:5060 (1178 bytes) INVITE sip:Pe8345WcBd@192.168.1.26:5060;transport=udp SIP/2.0 Record-Route: <sip:Pe8345WcBd@66.23.129.253:5060;nat=yes;ftag=SDtj14499-3c99ddd9-co3112-INS001;lr=on> Record-Route: <sip:206.80.67.68;lr=on;ftag=SDtj14499-3c99ddd9-co3112-INS001> Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bK6178.c92882f6.0 Via: SIP/2.0/UDP 206.80.67.68;branch=z9hG4bK6178.75830a7.0 Via: SIP/2.0/UDP 65.98.234.158:5060;branch=z9hG4bK3uscdq30384gi35927g0.1 Max-Forwards: 16 Contact: <sip:15097207844@65.98.234.158:5060;transport=udp> To: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872 From: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001 Call-ID: 83be7b59@pbx CSeq: 311201 INVITE Allow: INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,INFO,UPDATE Content-Type: application/sdp Date: Mon, 27 Feb 2012 19:49:55 GMT User-Agent: ENSR3.0.66.0-IS1-RMRG40340-RG2190-CPO3036 Privacy: id Content-Length: 223 v=0 o=- 1016715875 1016715876 IN IP4 65.98.234.158 s=ENSResip c=IN IP4 65.98.234.158 t=0 0 m=audio 54878 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv 2012/2/27 14:50:43 Tx: udp:66.23.129.253:5060 (1153 bytes) SIP/2.0 200 Ok Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bK6178.c92882f6.0 Via: SIP/2.0/UDP 206.80.67.68;branch=z9hG4bK6178.75830a7.0 Via: SIP/2.0/UDP 65.98.234.158:5060;branch=z9hG4bK3uscdq30384gi35927g0.1 Record-Route: <sip:Pe8345WcBd@66.23.129.253:5060;nat=yes;ftag=SDtj14499-3c99ddd9-co3112-INS001;lr=on> Record-Route: <sip:206.80.67.68;lr=on;ftag=SDtj14499-3c99ddd9-co3112-INS001> From: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001 To: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872 Call-ID: 83be7b59@pbx CSeq: 311201 INVITE Contact: <sip:Pe8345WcBd@192.168.1.26:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids Content-Type: application/sdp Content-Length: 249 v=0 o=- 986850866 986850866 IN IP4 192.168.1.26 s=- c=IN IP4 192.168.1.26 t=0 0 m=audio 55236 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv 2012/2/27 14:50:43 Rx: udp:66.23.129.253:5060 (1178 bytes) INVITE sip:Pe8345WcBd@192.168.1.26:5060;transport=udp SIP/2.0 Record-Route: <sip:Pe8345WcBd@66.23.129.253:5060;nat=yes;ftag=SDtj14499-3c99ddd9-co3112-INS001;lr=on> Record-Route: <sip:206.80.67.68;lr=on;ftag=SDtj14499-3c99ddd9-co3112-INS001> Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bK6178.c92882f6.0 Via: SIP/2.0/UDP 206.80.67.68;branch=z9hG4bK6178.75830a7.0 Via: SIP/2.0/UDP 65.98.234.158:5060;branch=z9hG4bK3uscdq30384gi35927g0.1 Max-Forwards: 16 Contact: <sip:15097207844@65.98.234.158:5060;transport=udp> To: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872 From: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001 Call-ID: 83be7b59@pbx CSeq: 311201 INVITE Allow: INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,INFO,UPDATE Content-Type: application/sdp Date: Mon, 27 Feb 2012 19:49:55 GMT User-Agent: ENSR3.0.66.0-IS1-RMRG40340-RG2190-CPO3036 Privacy: id Content-Length: 223 v=0 o=- 1016715875 1016715876 IN IP4 65.98.234.158 s=ENSResip c=IN IP4 65.98.234.158 t=0 0 m=audio 54878 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv 2012/2/27 14:50:43 Tr: udp:66.23.129.253:5060 (1153 bytes) SIP/2.0 200 Ok Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bK6178.c92882f6.0 Via: SIP/2.0/UDP 206.80.67.68;branch=z9hG4bK6178.75830a7.0 Via: SIP/2.0/UDP 65.98.234.158:5060;branch=z9hG4bK3uscdq30384gi35927g0.1 Record-Route: <sip:Pe8345WcBd@66.23.129.253:5060;nat=yes;ftag=SDtj14499-3c99ddd9-co3112-INS001;lr=on> Record-Route: <sip:206.80.67.68;lr=on;ftag=SDtj14499-3c99ddd9-co3112-INS001> From: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001 To: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872 Call-ID: 83be7b59@pbx CSeq: 311201 INVITE Contact: <sip:Pe8345WcBd@192.168.1.26:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.5.0.1016 Alpha Monocerotids Content-Type: application/sdp Content-Length: 249 v=0 o=- 986850866 986850866 IN IP4 192.168.1.26 s=- c=IN IP4 192.168.1.26 t=0 0 m=audio 55236 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv 2012/2/27 14:50:43 Rx: udp:66.23.129.253:5060 (705 bytes) ACK sip:Pe8345WcBd@192.168.1.26:5060;transport=udp SIP/2.0 Record-Route: <sip:Pe8345WcBd@66.23.129.253:5060;nat=yes;ftag=SDtj14499-3c99ddd9-co3112-INS001;lr=on> Record-Route: <sip:206.80.67.68;lr=on;ftag=SDtj14499-3c99ddd9-co3112-INS001> Via: SIP/2.0/UDP 66.23.129.253:5060;branch=0 Via: SIP/2.0/UDP 206.80.67.68;branch=z9hG4bK6178.75830a7.2 Via: SIP/2.0/UDP 65.98.234.158:5060;branch=z9hG4bK896l8230d08hka59c200.1 Max-Forwards: 16 To: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872 From: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001 Call-ID: 83be7b59@pbx CSeq: 311201 ACK User-Agent: ENSR3.0.66.0-IS1-RMRG40340-RG2190-CPO3036 Content-Length: 0 2012/2/27 14:50:43 Rx: udp:66.23.129.253:5060 (705 bytes) ACK sip:Pe8345WcBd@192.168.1.26:5060;transport=udp SIP/2.0 Record-Route: <sip:Pe8345WcBd@66.23.129.253:5060;nat=yes;ftag=SDtj14499-3c99ddd9-co3112-INS001;lr=on> Record-Route: <sip:206.80.67.68;lr=on;ftag=SDtj14499-3c99ddd9-co3112-INS001> Via: SIP/2.0/UDP 66.23.129.253:5060;branch=0 Via: SIP/2.0/UDP 206.80.67.68;branch=z9hG4bK6178.75830a7.2 Via: SIP/2.0/UDP 65.98.234.158:5060;branch=z9hG4bK896l8230d08hka59c200.1 Max-Forwards: 16 To: "Tony Yu" <sip:8775784777@localhost;user=phone>;tag=327783872 From: <sip:5097207844@localhost;user=phone>;tag=SDtj14499-3c99ddd9-co3112-INS001 Call-ID: 83be7b59@pbx CSeq: 311201 ACK User-Agent: ENSR3.0.66.0-IS1-RMRG40340-RG2190-CPO3036 Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted March 16, 2012 Report Share Posted March 16, 2012 We are having intermittent issues where we drop calls. Sometimes after a dropped call I get an email saying User disconnects call in the subject. Email below. Also it seems that we cannot get past the 11 minute mark while on the phone. The call gets disconnected and the phone has to be rebooted in order to make more calls. That is obviously a problem of the phone. The 7xx series had a couple of software images over the past few months. Not sure which is the latest one, but you need a new image. One side of the call between sip:15097207844@localhost;user=phone and sip:402@localhost did not receive media for 15.6 s and the other side of the call disconnected the call. The address of the other side was 66.23.129.253 (User-Agent=ENSR3.0.66.0-IS1-RMRG40340-RG2190-CPO3036). You may use this email as hint for a potential problem. The SIP messages are attached. That is a sign that there is trouble with one-way audio on hte trunk side. When the user disconnects the call, the PBX checks if the phone had to chance to hear anything, and in this case that seems not to be the case. It seems that your service provider does support the PBX running on a unroutable IP address, otherwise you would never get the INVITE from the SP. At first glance, it seems there is trouble with the firewall. Is this a cable modem router? Quote Link to comment Share on other sites More sharing options...
Tony Yu Posted March 16, 2012 Author Report Share Posted March 16, 2012 That is obviously a problem of the phone. The 7xx series had a couple of software images over the past few months. Not sure which is the latest one, but you need a new image. That is a sign that there is trouble with one-way audio on hte trunk side. When the user disconnects the call, the PBX checks if the phone had to chance to hear anything, and in this case that seems not to be the case. It seems that your service provider does support the PBX running on a unroutable IP address, otherwise you would never get the INVITE from the SP. At first glance, it seems there is trouble with the firewall. Is this a cable modem router? Not sure what you mean by software images? Are we talking about firmware? We have a T1 line running clean into PBX with no firewalls. Any ideas? Thanks, Quote Link to comment Share on other sites More sharing options...
pbx support Posted March 16, 2012 Report Share Posted March 16, 2012 Looking at the log, the call was established only for 21 seconds. PBX sends 200 OK to the phone (760) 2012/2/27 14:50:42 Tx: tcp:192.168.1.10:53201 (860 bytes) SIP/2.0 200 Ok The user at 760 disconnects the call after not getting any media for 15.6 s. 2012/2/27 14:51:03 Rx: tcp:192.168.1.10:53201 (597 bytes) BYE sip:402@192.168.1.26:5060;transport=tcp SIP/2.0 The issue could be anywhere all the way from the remote device/remote network/local network/Gateway/PBX/Phone etc Just curious, is the remote side MagicJack? Quote Link to comment Share on other sites More sharing options...
Tony Yu Posted March 16, 2012 Author Report Share Posted March 16, 2012 Looking at the log, the call was established only for 21 seconds. PBX sends 200 OK to the phone (760) 2012/2/27 14:50:42 Tx: tcp:192.168.1.10:53201 (860 bytes) SIP/2.0 200 Ok The user at 760 disconnects the call after not getting any media for 15.6 s. 2012/2/27 14:51:03 Rx: tcp:192.168.1.10:53201 (597 bytes) BYE sip:402@192.168.1.26:5060;transport=tcp SIP/2.0 The issue could be anywhere all the way from the remote device/remote network/local network/Gateway/PBX/Phone etc Just curious, is the remote side MagicJack? Nope, the remote side is NexVortex in US Any suggestions to help with my problem? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted March 18, 2012 Report Share Posted March 18, 2012 I would make sure that internal calls work fine. Make a couple of calls from the 760 to another phone and check if they are okay. If thats the case the problem is related to the SIP trunk. There we would have to focus on NAT related problems with the firewall, as they are most of the time the problem. Quote Link to comment Share on other sites More sharing options...
Recommended Posts
Join the conversation
You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.