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Incoming calls drop out


mybusinessvoice
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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

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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

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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.

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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.

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