Allstate Computers Posted August 17, 2011 Report Share Posted August 17, 2011 We're using hosted snom ONE ver 4.0.1.3499 (Linux). We're running into an intermittent issue where when you dial a number and hit the check mark it grabs a shared line you'll hear the comfort noise and then the light goes out and the call goes dead. Now the call does still go through to the other side but the callee gets dead air. We've tried 3 different providers, and 2 different Internet connections (Comcast and Windstream T1), as well as 3 different routers (Linksys WRT54GL, PFSense, SonicWall TZ180). The phones are Ssnom 320s and 370s. I've only seen it happen when we dial the number with the receiver on the hook, hit the check mark, and then pick up the phone before it starts to ring. However, I've had clients call with this problem that state they weren't using speakerphone. Sometimes if you hit the redial button, it'll do it again, and sometimes it'll go right through. Here is the call flow in the logfile: x's are the last 4 of callee's phone number [7] 2011/08/17 10:55:36: SIP Rx tls:64.139.77.137:54696: INVITE sip:xxxxxxx@pbx.allstatecomputers.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-5sw0ts03k0wc;rport From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone> Call-ID: 3c4b75252475-v0yf8yotze2n CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:405@192.168.100.120:1402;transport=tls;line=5z91mad1>;reg-id=1 X-Serialnumber: 00041326601A P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom370/8.2.29 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 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 530 [7] 2011/08/17 10:55:36: SIP Tx tls:64.139.77.137:54696: SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-5sw0ts03k0wc;rport=54696;received=64.139.77.137 From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone>;tag=aa112c7956 Call-ID: 3c4b75252475-v0yf8yotze2n CSeq: 1 INVITE Content-Length: 0 [7] 2011/08/17 10:55:36: Set packet length to 20 [6] 2011/08/17 10:55:36: Sending RTP for 3c4b75252475-v0yf8yotze2n#aa112c7956 to 192.168.100.120:59716 [7] 2011/08/17 10:55:36: SIP Tx udp:208.115.60.142:5060: INVITE sip:1561427xxx@208.115.60.142;user=phone SIP/2.0 Via: SIP/2.0/UDP 208.115.60.136:5060;branch=z9hG4bK-3e3e98944e760e7f42026e1128561203;rport From: <sip:5617431521@208.115.60.142>;tag=174310263 To: <sip:1561427xxxx@208.115.60.142;user=phone> Call-ID: a086938b@pbx CSeq: 14029 INVITE Max-Forwards: 70 Contact: <sip:5617431521@208.115.60.136:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: AllstateTelecom-PBX/4.0.1.3499 Content-Type: application/sdp Content-Length: 290 v=0 o=- 1066957623 1066957623 IN IP4 208.115.60.136 s=- c=IN IP4 208.115.60.136 t=0 0 m=audio 61400 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 [7] 2011/08/17 10:55:36: Set packet length to 20 [6] 2011/08/17 10:55:36: Send codec pcmu/8000 [7] 2011/08/17 10:55:36: SIP Tx tls:64.139.77.137:54696: SIP/2.0 183 Session Progress Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-5sw0ts03k0wc;rport=54696;received=64.139.77.137 From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone>;tag=aa112c7956 Call-ID: 3c4b75252475-v0yf8yotze2n CSeq: 1 INVITE Contact: <sip:405@208.115.60.136:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: AllstateTelecom-PBX/4.0.1.3499 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 484 v=0 o=- 1439637230 1439637230 IN IP4 208.115.60.136 s=- c=IN IP4 208.115.60.136 t=0 0 m=audio 54918 RTP/AVP 0 8 9 18 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:kdbRvVLTrWAjCHLIafsjJnz9s8mbK+JTtKFDOscV a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/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 [7] 2011/08/17 10:55:36: SIP Rx udp:208.115.60.142:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 208.115.60.136:5060;branch=z9hG4bK-3e3e98944e760e7f42026e1128561203;rport=5060 From: <sip:5617431521@208.115.60.142>;tag=174310263 To: <sip:1561427xxxx@208.115.60.142;user=phone>;tag=7696ca4181be7f5787517527c78a3092.7f39 Call-ID: a086938b@pbx CSeq: 14029 INVITE Proxy-Authenticate: Digest realm="208.115.60.142", nonce="4e4bd777000004bf7965b8ab80081eb702aeab3b0d6c916b" Server: OpenSIPS (1.6.4-2-tls (i386/linux)) Content-Length: 0 [7] 2011/08/17 10:55:36: SIP Tx udp:208.115.60.142:5060: ACK sip:1561427xxxx@208.115.60.142;user=phone SIP/2.0 Via: SIP/2.0/UDP 208.115.60.136:5060;branch=z9hG4bK-3e3e98944e760e7f42026e1128561203;rport From: <sip:5617431521@208.115.60.142>;tag=174310263 To: <sip:1561427xxxx@208.115.60.142;user=phone>;tag=7696ca4181be7f5787517527c78a3092.7f39 Call-ID: a086938b@pbx CSeq: 14029 ACK Max-Forwards: 70 Content-Length: 0 [7] 2011/08/17 10:55:36: SIP Tx udp:208.115.60.142:5060: INVITE sip:1561427xxxx@208.115.60.142;user=phone SIP/2.0 Via: SIP/2.0/UDP 208.115.60.136:5060;branch=z9hG4bK-5e0d32504503372356c54f08ea6ef395;rport From: <sip:5617431521@208.115.60.142>;tag=174310263 To: <sip:1561427xxxx@208.115.60.142;user=phone> Call-ID: a086938b@pbx CSeq: 14030 INVITE Max-Forwards: 70 Contact: <sip:5617431521@208.115.60.136:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: AllstateTelecom-PBX/4.0.1.3499 Proxy-Authorization: Digest realm="208.115.60.142",nonce="4e4bd777000004bf7965b8ab80081eb702aeab3b0d6c916b",response="643e10a3e747539004c22ef2b9175779",username="5617431521",uri="sip:1561427xxxx@208.115.60.142;user=phone",algorithm=MD5 Content-Type: application/sdp Content-Length: 290 v=0 o=- 1066957623 1066957623 IN IP4 208.115.60.136 s=- c=IN IP4 208.115.60.136 t=0 0 m=audio 61400 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 [7] 2011/08/17 10:55:36: SIP Tr udp:208.115.60.142:5060: INVITE sip:1561427xxxx@208.115.60.142;user=phone SIP/2.0 Via: SIP/2.0/UDP 208.115.60.136:5060;branch=z9hG4bK-5e0d32504503372356c54f08ea6ef395;rport From: <sip:5617431521@208.115.60.142>;tag=174310263 To: <sip:1561427xxxx@208.115.60.142;user=phone> Call-ID: a086938b@pbx CSeq: 14030 INVITE Max-Forwards: 70 Contact: <sip:5617431521@208.115.60.136:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: AllstateTelecom-PBX/4.0.1.3499 Proxy-Authorization: Digest realm="208.115.60.142",nonce="4e4bd777000004bf7965b8ab80081eb702aeab3b0d6c916b",response="643e10a3e747539004c22ef2b9175779",username="5617431521",uri="sip:1561427xxxx@208.115.60.142;user=phone",algorithm=MD5 Content-Type: application/sdp Content-Length: 290 v=0 o=- 1066957623 1066957623 IN IP4 208.115.60.136 s=- c=IN IP4 208.115.60.136 t=0 0 m=audio 61400 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 [7] 2011/08/17 10:55:37: SIP Rx tls:64.139.77.137:54696: PRACK sip:405@208.115.60.136:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-s7rdxl5fra9l;rport From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone>;tag=aa112c7956 Call-ID: 3c4b75252475-v0yf8yotze2n CSeq: 2 PRACK Max-Forwards: 70 Contact: <sip:405@192.168.100.120:1402;transport=tls;line=5z91mad1>;reg-id=1 RAck: 1 1 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 [7] 2011/08/17 10:55:37: SIP Tx tls:64.139.77.137:54696: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-s7rdxl5fra9l;rport=54696;received=64.139.77.137 From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone>;tag=aa112c7956 Call-ID: 3c4b75252475-v0yf8yotze2n CSeq: 2 PRACK Contact: <sip:405@208.115.60.136:5061;transport=tls> User-Agent: AllstateTelecom-PBX/4.0.1.3499 Content-Length: 0 [7] 2011/08/17 10:55:37: Discard SRTCP packet from 64.139.77.137:3725 with wrong MAC [6] 2011/08/17 10:55:37: Sending RTP for 3c4b75252475-v0yf8yotze2n#aa112c7956 to 64.139.77.137:3724 [7] 2011/08/17 10:55:37: SIP Rx udp:208.115.60.142:5060: SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 208.115.60.136:5060;branch=z9hG4bK-5e0d32504503372356c54f08ea6ef395;rport=5060 From: <sip:5617431521@208.115.60.142>;tag=174310263 To: <sip:1561427xxxx@208.115.60.142;user=phone> Call-ID: a086938b@pbx CSeq: 14030 INVITE Server: OpenSIPS (1.6.4-2-tls (i386/linux)) Content-Length: 0 [7] 2011/08/17 10:55:37: SIP Rx udp:208.115.60.142:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 208.115.60.136:5060;branch=z9hG4bK-5e0d32504503372356c54f08ea6ef395;rport=5060 From: <sip:5617431521@208.115.60.142>;tag=174310263 To: <sip:1561427xxxx@208.115.60.142;user=phone>;tag=7696ca4181be7f5787517527c78a3092.0d2d Call-ID: a086938b@pbx CSeq: 14030 INVITE Proxy-Authenticate: Digest realm="208.115.60.142", nonce="4e4bd779000004c0e6af14460e0b7de18ace196d8577917f", stale=true Server: OpenSIPS (1.6.4-2-tls (i386/linux)) Content-Length: 0 [7] 2011/08/17 10:55:37: Call a086938b@pbx#174310263: Clear last INVITE [7] 2011/08/17 10:55:37: SIP Tx udp:208.115.60.142:5060: ACK sip:1561427xxxx@208.115.60.142;user=phone SIP/2.0 Via: SIP/2.0/UDP 208.115.60.136:5060;branch=z9hG4bK-5e0d32504503372356c54f08ea6ef395;rport From: <sip:5617431521@208.115.60.142>;tag=174310263 To: <sip:1561427xxxx@208.115.60.142;user=phone>;tag=7696ca4181be7f5787517527c78a3092.0d2d Call-ID: a086938b@pbx CSeq: 14030 ACK Max-Forwards: 70 Contact: <sip:5617431521@208.115.60.136:5060;transport=udp> Content-Length: 0 [5] 2011/08/17 10:55:37: INVITE Response 407 Proxy Authentication Required: Terminate a086938b@pbx [7] 2011/08/17 10:55:37: SIP Tx tls:64.139.77.137:54696: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-5sw0ts03k0wc;rport=54696;received=64.139.77.137 From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone>;tag=aa112c7956 Call-ID: 3c4b75252475-v0yf8yotze2n CSeq: 1 INVITE Contact: <sip:405@208.115.60.136:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: AllstateTelecom-PBX/4.0.1.3499 Content-Length: 0 [7] 2011/08/17 10:55:38: SIP Rx tls:64.139.77.137:54696: ACK sip:427xxxx@pbx.allstatecomputers.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-5sw0ts03k0wc;rport From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone>;tag=aa112c7956 Call-ID: 3c4b75252475-v0yf8yotze2n CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:405@192.168.100.120:1402;transport=tls;line=5z91mad1>;reg-id=1 Proxy-Require: buttons Content-Length: 0 [7] 2011/08/17 10:55:39: SIP Rx udp:208.115.60.142:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 208.115.60.136:5060;received=208.115.60.136;branch=z9hG4bK-5e0d32504503372356c54f08ea6ef395;rport=5060 Record-Route: <sip:sansay527215264rdb19609@64.136.174.30:5060;lr;transport=udp> Record-Route: <sip:208.115.60.142;lr=on> To: <sip:1561427xxxx@208.115.60.142;user=phone>;tag=sansay527215264rdb19609 From: <sip:5617431521@208.115.60.142>;tag=174310263 Call-ID: a086938b@pbx CSeq: 14030 INVITE Contact: <sip:1561427xxxx@64.136.174.30:5060> Content-Type: application/sdp Content-Length: 214 v=0 o=Sansay-VSXi 188 1 IN IP4 64.136.174.30 s=Session Controller c=IN IP4 74.120.93.70 t=0 0 m=audio 21672 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 [7] 2011/08/17 10:55:41: SIP Rx tls:64.139.77.137:54696: CANCEL sip:427xxxx@pbx.allstatecomputers.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-5sw0ts03k0wc;rport From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone> Call-ID: 3c4b75252475-v0yf8yotze2n CSeq: 1 CANCEL Max-Forwards: 70 Reason: SIP;cause=487;text="Request terminated by user" Proxy-Require: buttons Content-Length: 0 [7] 2011/08/17 10:55:41: SIP Tx tls:64.139.77.137:54696: SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-5sw0ts03k0wc;rport=54696;received=64.139.77.137 From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone> Call-ID: 3c4b75252475-v0yf8yotze2n CSeq: 1 CANCEL Content-Length: 0 Any input would be greatly appreciated. Thanks, Brian Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted August 18, 2011 Report Share Posted August 18, 2011 check the obvious stuff. Has security clobbered the ip addresses ranges Call me crazy, but our crafted Centos and Windows PBX's have public facing IP addresses, Are your trunk provider interfaces separate nics? Are they static Routed to the Carrier? Can you duplicate this with a NON-SNOM, soft - Phone Etc. While PBX traces are Ok, a passive wireshark trace tells a much more complete story. The Goal in any troubleshooting scenario is to ask questions and take small incremental steps so that the only answer to the question being asked is YES or No. Hopefully you can take a SWAG at the first question so isolate the system into two parts. then take action and when looking at the evidence you can determine is this a problem 100% on the left or right side of where you are looking. Is the problem on the trunk or caller side? Does the same issue exist with another NIC? and on, and on, and on, Good Luck. Andy Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted August 19, 2011 Report Share Posted August 19, 2011 Well, from what I see it look like you have the wrong password. The service provider responds with a challenge, then the PBX answers and the servie provider challenges again. This is typically the case then password is not correct or you have chosen the wrong account. Quote Link to comment Share on other sites More sharing options...
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