01-12-2013 03:58 PM - edited 03-21-2019 09:57 AM
Hi fellow members,
Latest firmware installed 1.3.1, all incoming calls are dropped after the first ring and then followed by busy tone, all outgoing calls are fully functional.
There is not much to say except that i followed all the ITSP config requirement and their techs cannot help me because they do not support Cisco products.
Thanks in advance
Martin Roy
01-13-2013 09:18 AM
Can you enable SPA logs and post a capture of problematic call?
Regards.
01-13-2013 10:02 AM
I will setup the log files config in the next few hours and i will keep you posted as soon i have the log file system up and running.
Thanks a lot.
01-13-2013 08:40 PM
01-13-2013 12:27 PM
01-14-2013 03:06 AM
Please, enable also voice debug under VOICE - SYSTEM - MISCELLANEOUS SETTING. Set "Debug Level" to 3 and log priority to "debug".
Thanks.
01-14-2013 07:31 AM
01-14-2013 08:50 AM
Hi,
Here the log file for a "FAILING incoming call" where the SIP Debug option for Line 1 was activated with the "FULL" feature.
n.b. : You know what, the technicians from YAK (ITSP) still insist to say that is a modem port forwarding issue. The technicians from Videotron(ISP) are not at all collaborative.
Could it be possible that the problem is coming from the modem ?
Thanks and regards.
01-14-2013 09:39 AM
In incoming calls, your SPA sends a "180 ringing" to your ITSP. After this phase the ITSP cancels the call with a "Cancel" message.
I think that the 180 message contains some headers that your ITSP doesn't understand. So it rejects the call.
SIP/2.0 180 Ringing
To: <>>100102YY0NJ7BVI@sip.tor.yakdigitalphone.ca>;tag=cc054f2c44a97d0ei0
From: "ROY M" <>>4186592025@sip.tor.yakdigitalphone.ca>;tag=gQeaSS9v835tj
Call-ID: 23857e3d-d90c-1230-dca3-0017a4770048
CSeq: 38754791 INVITE
Via: SIP/2.0/UDP 204.11.120.122;branch=z9hG4bKb555.79c7c1e7.1
Via: SIP/2.0/UDP 204.11.121.216:1216;received=10.150.4.20;rport=5080;branch=z9hG4bKv4c15pp96DX7c
Record-Route: <204.11.120.122>204.11.120.122>
Contact: "Linuxsoft"
Server: Cisco/SPA122-1.3.1(003)
Remote-Party-ID: "Linuxsoft" <>>YY0NJ7BVI@sip.tor.yakdigitalphone.ca>;screen=yes;party=called
Content-Length: 0
CANCEL sip:YY0NJ7BVI@173.177.84.137:5060 SIP/2.0
Via: SIP/2.0/UDP 204.11.120.122;branch=z9hG4bKb555.79c7c1e7.1
From: "ROY M" <>>4186592025@sip.tor.yakdigitalphone.ca>;tag=gQeaSS9v835tj
Call-ID: 23857e3d-d90c-1230-dca3-0017a4770048
To: <>>100102YY0NJ7BVI@sip.tor.yakdigitalphone.ca>
CSeq: 38754791 CANCEL
Max-Forwards: 70
User-Agent: OpenSIPS (1.6.3-tls (x86_64/linux))
Content-Length: 0
Can you ask your ISP to debug a call?
Can your ITSP provide a "180 ringing" message example of a SPA2102?
The provisioning used from your CPE is designed for a SPA2102.
In this way we can compare the behaviours.
Regards.
01-14-2013 12:42 PM
Hi,
Q. Can you ask your ISP to debug a call ?
A. Certainely not since he is promoting his own ITSP service.
Q. Can your ITSP provide a "180 ringing" message example of a SPA2102 ?
A. The tech did not understand of what we are talking about. Therefore he refered the case to his supervisor.
The only think i could get is two provisioning config files.
1. http://provisioning.yakdigital.ca/yakvoip/YAKVoIPGlobal.cfg
2. http://provisioning.yakdigital.ca/yakvoip/YY0NJ7BVI.cfg
The ITSP is in conflict of interests since he is renting and selling LinksysSPA2102.
Probably the best solution is to get their SPA2102 and stop this lost of time for both.
Thanks and regards.
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