mybusinessvoice Posted September 13, 2011 Report Share Posted September 13, 2011 Hi, I have a hosted PBX, all of my domains are having the same problem with calls coming into the PBX cut off after about 3-4 rings but 50% of the time the call can ring and go to voicemail Sip logs [5] 2011/09/14 21:50:37: Domain trunk Second trunk@sydneydentalprofessionals sends call to 190 in domain sydneydentalprofessionals [7] 2011/09/14 21:50:37: Call 2403e639@pbx#2049: Clear last request [7] 2011/09/14 21:50:37: Call 0e3adfe2@pbx#9137: Clear last request [8] 2011/09/14 21:50:47: Hangup: Call call-F1835DA7-4AC1-2E10-1D0C-4EF93@203.176.186.12~1o#7189fba86e not found [7] 2011/09/14 21:50:47: Call 2403e639@pbx#2049: Clear last request [7] 2011/09/14 21:50:47: Call 0e3adfe2@pbx#9137: Clear last request [7] 2011/09/14 21:50:47: Call 2403e639@pbx#2049: Clear last INVITE [5] 2011/09/14 21:50:47: INVITE Response 487 Request Terminated: Terminate 2403e639@pbx [7] 2011/09/14 21:50:47: Call 0e3adfe2@pbx#9137: Clear last INVITE [5] 2011/09/14 21:50:47: INVITE Response 487 Request Terminated: Terminate 0e3adfe2@pbx All my extensions and trunk are set to G729 Thanks Leigh Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 14, 2011 Report Share Posted September 14, 2011 The log does not produce anything suspicious... G729 is always suspicious because of the way the codec must be licensed (per call), maybe you just have too many calls and then the PBX drops calls more or less randomly. Quote Link to comment Share on other sites More sharing options...
mybusinessvoice Posted September 14, 2011 Author Report Share Posted September 14, 2011 i have a hosted rental lic with snom plus i tested the system when there was know one on it Quote Link to comment Share on other sites More sharing options...
mybusinessvoice Posted September 14, 2011 Author Report Share Posted September 14, 2011 im running firmware 4.0.1.3499 is this a problem? Quote Link to comment Share on other sites More sharing options...
pbx support Posted September 14, 2011 Report Share Posted September 14, 2011 im running firmware 4.0.1.3499 is this a problem? Nothing that we know of. But it is always possible that the caller hangup or the provider sending a BYE before the call goes to voice mail. Quote Link to comment Share on other sites More sharing options...
mybusinessvoice Posted September 14, 2011 Author Report Share Posted September 14, 2011 Ok thanks, my system is a hosted PBX with multiple domains as the system grows in extensions do i need to increase the global settings in the system to handle more traffic? if so which ones. Because this problem has only just started as ive increase the number of domains and extensions Thanks Quote Link to comment Share on other sites More sharing options...
mybusinessvoice Posted September 17, 2011 Author Report Share Posted September 17, 2011 Can you please help me with this problem i still have incoming calls drop out after 3-4 rings Logfile: Clear or Reload the log. [7] 2011/09/17 10:27:39: SIP Rx udp:203.176.185.10:5060: CANCEL sip:61298993799@119.252.88.194:5060;transport=udp;line=182be0c5 SIP/2.0 Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bK4602.0865205332bb78e8c86dd7016a6df7c9.0 Via: SIP/2.0/UDP 203.176.185.10:5061;rport=5061;branch=z9hG4bK6385b86ed87d03ed097262b49db7069e Max-Forwards: 16 From: <sip:0414352401@203.176.185.10>;tag=b64bc8384a85907ad21d3d20eb61e8e3 To: <sip:61298993799@203.176.185.10> Call-ID: call-F1789020-34C3-2E10-0A0D-278CA@203.176.186.11~1o CSeq: 200 CANCEL Expires: 300 User-Agent: Sippy [7] 2011/09/17 10:27:39: SIP Tx udp:203.176.185.10:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bK4602.0865205332bb78e8c86dd7016a6df7c9.0 Via: SIP/2.0/UDP 203.176.185.10:5061;rport=5061;branch=z9hG4bK6385b86ed87d03ed097262b49db7069e From: <sip:0414352401@203.176.185.10>;tag=b64bc8384a85907ad21d3d20eb61e8e3 To: <sip:61298993799@203.176.185.10>;tag=bd9c12b372 Call-ID: call-F1789020-34C3-2E10-0A0D-278CA@203.176.186.11~1o CSeq: 200 CANCEL Contact: <sip:61298993799@119.252.88.194:5060;transport=udp> User-Agent: Mybusinessvoice/4.0.1.3499 Content-Length: 0 [7] 2011/09/17 10:27:39: SIP Tx udp:203.176.185.10:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bK4602.0865205332bb78e8c86dd7016a6df7c9.0 Via: SIP/2.0/UDP 203.176.185.10:5061;branch=z9hG4bK6385b86ed87d03ed097262b49db7069e;rport=5061 Record-Route: <sip:203.176.185.10;ftag=b64bc8384a85907ad21d3d20eb61e8e3;lr> From: <sip:0414352401@203.176.185.10>;tag=b64bc8384a85907ad21d3d20eb61e8e3 To: <sip:61298993799@203.176.185.10>;tag=bd9c12b372 Call-ID: call-F1789020-34C3-2E10-0A0D-278CA@203.176.186.11~1o CSeq: 200 INVITE Contact: <sip:61298993799@119.252.88.194:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: Mybusinessvoice/4.0.1.3499 Content-Length: 0 [7] 2011/09/17 10:27:39: SIP Tx udp:165.228.88.83:2048: CANCEL sip:100@165.228.88.83:2048;line=uxvspv5f SIP/2.0 Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-b6f5d0a576a3752b8b3510b6cebfcc49;rport From: <sip:0414352401@203.176.185.10;user=phone>;tag=18577 To: <sip:61298993799@203.176.185.10;user=phone> Call-ID: 95295cf1@pbx CSeq: 4223 CANCEL Max-Forwards: 70 Content-Length: 0 [7] 2011/09/17 10:27:39: SIP Tx udp:165.228.88.83:1024: CANCEL sip:101@165.228.88.83:1024;line=cozyryzk SIP/2.0 Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-fbaa22b07c9e5fcda08b0e1dcae35b45;rport From: <sip:0414352401@203.176.185.10;user=phone>;tag=49205 To: <sip:61298993799@203.176.185.10;user=phone> Call-ID: 60769716@pbx CSeq: 10 CANCEL Max-Forwards: 70 Content-Length: 0 [7] 2011/09/17 10:27:39: SIP Tx udp:165.228.88.83:1026: CANCEL sip:102@165.228.88.83:1026;line=ybc60i8q SIP/2.0 Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-070d3ac9608ccb9742d1d9e49b19c814;rport From: <sip:0414352401@203.176.185.10;user=phone>;tag=62968 To: <sip:61298993799@203.176.185.10;user=phone> Call-ID: 9885769c@pbx CSeq: 23188 CANCEL Max-Forwards: 70 Content-Length: 0 [7] 2011/09/17 10:27:39: SIP Tx udp:165.228.88.83:1025: CANCEL sip:103@165.228.88.83:1025;line=bpetirz2 SIP/2.0 Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-fcd6a3e79831ca2c4991cc4f09818cb7;rport From: <sip:0414352401@203.176.185.10;user=phone>;tag=7414 To: <sip:61298993799@203.176.185.10;user=phone> Call-ID: 0262a24e@pbx CSeq: 16007 CANCEL Max-Forwards: 70 Content-Length: 0 [7] 2011/09/17 10:27:39: SIP Rx udp:203.176.185.10:5060: ACK sip:61298993799@119.252.88.194:5060;transport=udp;line=182be0c5 SIP/2.0 Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bK4602.0865205332bb78e8c86dd7016a6df7c9.0 From: <sip:0414352401@203.176.185.10>;tag=b64bc8384a85907ad21d3d20eb61e8e3 Call-ID: call-F1789020-34C3-2E10-0A0D-278CA@203.176.186.11~1o To: <sip:61298993799@203.176.185.10>;tag=bd9c12b372 CSeq: 200 ACK User-Agent: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 [8] 2011/09/17 10:27:39: Hangup: Call call-F1789020-34C3-2E10-0A0D-278CA@203.176.186.11~1o#bd9c12b372 not found [7] 2011/09/17 10:27:39: SIP Rx udp:165.228.88.83:2048: SIP/2.0 200 OK Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-b6f5d0a576a3752b8b3510b6cebfcc49;rport=5060 From: <sip:0414352401@203.176.185.10;user=phone>;tag=18577 To: <sip:61298993799@203.176.185.10;user=phone>;tag=0r2e4lbqfr Call-ID: 95295cf1@pbx CSeq: 4223 CANCEL Content-Length: 0 [7] 2011/09/17 10:27:39: Call 95295cf1@pbx#18577: Clear last request [7] 2011/09/17 10:27:39: SIP Rx udp:165.228.88.83:1024: SIP/2.0 200 OK Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-fbaa22b07c9e5fcda08b0e1dcae35b45;rport=5060 From: <sip:0414352401@203.176.185.10;user=phone>;tag=49205 To: <sip:61298993799@203.176.185.10;user=phone>;tag=l75r3t8le3 Call-ID: 60769716@pbx CSeq: 10 CANCEL Content-Length: 0 [7] 2011/09/17 10:27:39: Call 60769716@pbx#49205: Clear last request [7] 2011/09/17 10:27:39: SIP Rx udp:165.228.88.83:1026: SIP/2.0 200 OK Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-070d3ac9608ccb9742d1d9e49b19c814;rport=5060 From: <sip:0414352401@203.176.185.10;user=phone>;tag=62968 To: <sip:61298993799@203.176.185.10;user=phone>;tag=lvaq3si9p0 Call-ID: 9885769c@pbx CSeq: 23188 CANCEL Content-Length: 0 [7] 2011/09/17 10:27:39: Call 9885769c@pbx#62968: Clear last request [7] 2011/09/17 10:27:39: SIP Rx udp:165.228.88.83:2048: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-b6f5d0a576a3752b8b3510b6cebfcc49;rport=5060 From: <sip:0414352401@203.176.185.10;user=phone>;tag=18577 To: <sip:61298993799@203.176.185.10;user=phone>;tag=0r2e4lbqfr Call-ID: 95295cf1@pbx CSeq: 4223 INVITE Contact: <sip:100@165.228.88.83:2048;line=uxvspv5f>;reg-id=1 Content-Length: 0 [7] 2011/09/17 10:27:39: Call 95295cf1@pbx#18577: Clear last INVITE [7] 2011/09/17 10:27:39: SIP Tx udp:165.228.88.83:2048: ACK sip:100@165.228.88.83:2048;line=uxvspv5f SIP/2.0 Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-b6f5d0a576a3752b8b3510b6cebfcc49;rport From: <sip:0414352401@203.176.185.10;user=phone>;tag=18577 To: <sip:61298993799@203.176.185.10;user=phone>;tag=0r2e4lbqfr Call-ID: 95295cf1@pbx CSeq: 4223 ACK Max-Forwards: 70 Contact: <sip:100@119.252.88.194:5060;transport=udp> Content-Length: 0 [5] 2011/09/17 10:27:39: INVITE Response 487 Request Terminated: Terminate 95295cf1@pbx [7] 2011/09/17 10:27:39: SIP Rx udp:165.228.88.83:1025: SIP/2.0 200 OK Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-fcd6a3e79831ca2c4991cc4f09818cb7;rport=5060 From: <sip:0414352401@203.176.185.10;user=phone>;tag=7414 To: <sip:61298993799@203.176.185.10;user=phone>;tag=aob35ydmc0 Call-ID: 0262a24e@pbx CSeq: 16007 CANCEL Content-Length: 0 [7] 2011/09/17 10:27:39: Call 0262a24e@pbx#7414: Clear last request [7] 2011/09/17 10:27:39: SIP Rx udp:165.228.88.83:1024: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-fbaa22b07c9e5fcda08b0e1dcae35b45;rport=5060 From: <sip:0414352401@203.176.185.10;user=phone>;tag=49205 To: <sip:61298993799@203.176.185.10;user=phone>;tag=l75r3t8le3 Call-ID: 60769716@pbx CSeq: 10 INVITE Contact: <sip:101@165.228.88.83:1024;line=cozyryzk>;reg-id=1 Content-Length: 0 [7] 2011/09/17 10:27:39: Call 60769716@pbx#49205: Clear last INVITE [7] 2011/09/17 10:27:39: SIP Tx udp:165.228.88.83:1024: ACK sip:101@165.228.88.83:1024;line=cozyryzk SIP/2.0 Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-fbaa22b07c9e5fcda08b0e1dcae35b45;rport From: <sip:0414352401@203.176.185.10;user=phone>;tag=49205 To: <sip:61298993799@203.176.185.10;user=phone>;tag=l75r3t8le3 Call-ID: 60769716@pbx CSeq: 10 ACK Max-Forwards: 70 Contact: <sip:101@119.252.88.194:5060;transport=udp> Content-Length: 0 [5] 2011/09/17 10:27:39: INVITE Response 487 Request Terminated: Terminate 60769716@pbx [7] 2011/09/17 10:27:39: SIP Rx udp:165.228.88.83:1026: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-070d3ac9608ccb9742d1d9e49b19c814;rport=5060 From: <sip:0414352401@203.176.185.10;user=phone>;tag=62968 To: <sip:61298993799@203.176.185.10;user=phone>;tag=lvaq3si9p0 Call-ID: 9885769c@pbx CSeq: 23188 INVITE Contact: <sip:102@165.228.88.83:1026;line=ybc60i8q>;reg-id=1 Content-Length: 0 [7] 2011/09/17 10:27:39: Call 9885769c@pbx#62968: Clear last INVITE [7] 2011/09/17 10:27:39: SIP Tx udp:165.228.88.83:1026: ACK sip:102@165.228.88.83:1026;line=ybc60i8q SIP/2.0 Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-070d3ac9608ccb9742d1d9e49b19c814;rport From: <sip:0414352401@203.176.185.10;user=phone>;tag=62968 To: <sip:61298993799@203.176.185.10;user=phone>;tag=lvaq3si9p0 Call-ID: 9885769c@pbx CSeq: 23188 ACK Max-Forwards: 70 Contact: <sip:102@119.252.88.194:5060;transport=udp> Content-Length: 0 [5] 2011/09/17 10:27:39: INVITE Response 487 Request Terminated: Terminate 9885769c@pbx [7] 2011/09/17 10:27:39: SIP Rx udp:165.228.88.83:1025: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-fcd6a3e79831ca2c4991cc4f09818cb7;rport=5060 From: <sip:0414352401@203.176.185.10;user=phone>;tag=7414 To: <sip:61298993799@203.176.185.10;user=phone>;tag=aob35ydmc0 Call-ID: 0262a24e@pbx CSeq: 16007 INVITE Contact: <sip:103@165.228.88.83:1025;line=bpetirz2>;reg-id=1 Content-Length: 0 [7] 2011/09/17 10:27:39: Call 0262a24e@pbx#7414: Clear last INVITE [7] 2011/09/17 10:27:39: SIP Tx udp:165.228.88.83:1025: ACK sip:103@165.228.88.83:1025;line=bpetirz2 SIP/2.0 Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-fcd6a3e79831ca2c4991cc4f09818cb7;rport From: <sip:0414352401@203.176.185.10;user=phone>;tag=7414 To: <sip:61298993799@203.176.185.10;user=phone>;tag=aob35ydmc0 Call-ID: 0262a24e@pbx CSeq: 16007 ACK Max-Forwards: 70 Contact: <sip:103@119.252.88.194:5060;transport=udp> Content-Length: 0 [5] 2011/09/17 10:27:39: INVITE Response 487 Request Terminated: Terminate 0262a24e@pbx Quote 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Vodia PBX Posted September 17, 2011 Report Share Posted September 17, 2011 Well, it seems that the whole CANCEL sequence is triggered by the very first message in the log, coming from the trunk. It seems that the trunk for some reason wants to cancel the call. Possible ideas: The call was never connected for whatever reason, and after 60 seconds the carrier decides to cancel/disconnect the call. OR the carrier has a problem receiving a SDP in the 18x message (you can turn that off in the trunk settings); although I dont remember any SIP device having a problem with that over the last couple of years. It would be interesting to see the INVITE/183 messages that are exchanged between the carrier and PBX; maybe you can filter for the IP address of the trunk (203.176.185.10) and check everything from the INVITE message on. Quote Link to comment Share on other sites More sharing options...
mybusinessvoice Posted September 19, 2011 Author Report Share Posted September 19, 2011 great can you tell me what i need to turn off because i cant find anything called SDP in the trunk settings Thanks Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 19, 2011 Report Share Posted September 19, 2011 There is a settings called "Ringback", change it to "message 180" and givbe it a try. Quote Link to comment Share on other sites More sharing options...
mybusinessvoice Posted September 19, 2011 Author Report Share Posted September 19, 2011 YES it works thank you, with it set to message 180 will this be a problem in the future for any other features? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 19, 2011 Report Share Posted September 19, 2011 It is a old settings, so far we dont see a reason why we should remove it. I thought most SIP trunk providers could "already" deal with media in the ringback phase, but it seems that there is still stuff out there which has a problem with that. Sending media during the ringback phase makes it e.g. possible to play MoH in the ACD while waiting and it does reduce the time for the pickup media establishment. Quote Link to comment Share on other sites More sharing options...
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