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Old July 26th, 2006, 05:51 PM
Junior Member
 
Join Date: Dec 2005
Posts: 29
strobes
Default Registration with Viatalk server unstable

For last few weeks my connection with Viatalk is unstable and most of the problems with incoming calls on the VT trunk.
The calling party has no ring back. And I don’t see the invite message from VT server.
If the caller hangs up and redial – everything works fine.
By the way I think I did not have this problem with my temp DID and it has been discovered when I ported my main number
So here is my advise: do your testing right. This problem drives me nuts. And I hear complains from friends and family, you know what I mean.
Any way VT support is helpless and tickets opened on VT web site disappear?!
So I’m here asking for help. Attached the log file by running sip debug peer and shows that connection dropped ones in a wile and restores.
I think it’s not normal? Can this contribute to my problem? Any recommendations how to fix it?

--- (12 headers 3 lines)---
Transmitting (NAT) to 67.15.199.23:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.15.199.23:5060;branch=z9hG4bK7e6cbc00;received= 67.15.199.23;rport=5060
From: "asterisk" <sip:asterisk@67.15.199.23>;tag=as38a7c9ca
To: <sip:1972612XXXX@71.240.164.245>;tag=as029391cb
Call-ID: 133c4e03076b83146f32a42a12b87704@67.15.199.23
CSeq: 102 NOTIFY
User-Agent: AVTV
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '133c4e03076b83146f32a42a12b87704@67.15.199.23'
Destroying call '4e92c7c264567e705ab1b5d65267b611@127.0.0.1'
12 headers, 0 lines
Reliably Transmitting (NAT) to 67.15.199.23:5060:
OPTIONS sip:mercury.vtnoc.net SIP/2.0
Via: SIP/2.0/UDP 71.240.164.245:5060;branch=z9hG4bK24d150fd;rport
From: "Unknown" <sip:Unknown@71.240.164.245>;tag=as04d02446
To: <sip:mercury.vtnoc.net>
Contact: <sip:Unknown@71.240.164.245>
Call-ID: 60b82b8035a6e14f15b57e427f655b7f@71.240.164.245
CSeq: 102 OPTIONS
User-Agent: AVTV
Max-Forwards: 70
Date: Wed, 26 Jul 2006 16:00:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 67.15.199.23:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 71.240.164.245:5060;branch=z9hG4bK24d150fd;rport;r eceived=71.240.164.245
From: "Unknown" <sip:Unknown@71.240.164.245>;tag=as04d02446
To: <sip:mercury.vtnoc.net>;tag=as7f2afe07
Call-ID: 60b82b8035a6e14f15b57e427f655b7f@71.240.164.245
CSeq: 102 OPTIONS
User-Agent: Viatalk SIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:67.15.199.23>
Accept: application/sdp
Content-Length: 0
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