Comments on: Asterisk sip.conf configuration

By: Thomas Simon

Thomas Simon — Fri, 01 Apr 2011 07:10:40 +0000

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By: admin

admin — Sat, 13 Feb 2010 17:37:08 +0000

Nalini,

Can you please share your sip.conf and extensions.conf lines for this user/call. That would be better if you little explain your network diagram. In first guess i think it is NAT issue. But would be able to help once you provide more information.

By: admin

admin — Sat, 13 Feb 2010 17:32:09 +0000

Guys, I was alot busy with family and I couldn't answer the question, however i'll try to answer the question that i can.

vishu gaddi, Your question about if you can use variables in Asterisk sip.conf? I don't think you can use variables in sip.conf, if you have 100 accounts, then you would need to create different account parameter for each sip individual account.

Please share with others if you find any solution to use variables in sip.conf

By: Nalini

Nalini — Wed, 10 Feb 2010 09:58:47 +0000

Hi, I am using asterix1.4 server and x-lite as SIP clients. I have made all the changes in sip.conf and extensions.conf. I am able to register the agents as well. But, when I try to make a call, call is getting established, buut I am unable to hear the voice in the call. Don't know what is the reason for this. Somebody, please help.

By: Ceison

Ceison — Fri, 13 Nov 2009 19:55:35 +0000

I installed new Asterisk version 1.6.1.9. the problem is when i dial an extension that setup in dialplan doesn’t work and i see the following on CLI.

Using SIP RTP CoS mark 5

Not sure what does it mean, and how i can fix this issue.

By: Ziv

Ziv — Mon, 20 Jul 2009 06:07:30 +0000

Hi,

I guess I’m having a similar problem. SIP users register at the Asterisk OK, but how can I forward calls that arrive for dynamically registered users to their remote address?

I mean that the incoming request-URI is arriving user@asterisk. I need to change the request-uri to contact-uri-of-the-register-request and to forward the message for that remote URI.

Thanks!

By: admin

admin — Tue, 23 Jun 2009 12:48:16 +0000

Adhya, Can you please explain your question in little more detail? Do you mean you can successfully registered your SIP clients with Asterisk but that client is unable to make call? when does this call fail? when someone dial this ext or when you make outbound call from this ext?

By: Adhya

Adhya — Tue, 23 Jun 2009 06:58:59 +0000

Hi,

How to configure for dynamic users? the registrations succeeds but call fails with a “404 response”.
Kindly assist in this effort.

Thanks

By: Fozia

Fozia — Thu, 19 Mar 2009 11:03:21 +0000

Hi,

My Asterisk PBX technical team is facing some problem. We just upgrade from 1.4 to 1.6 and now we see on CLI that insecure=yes/invite is not supported by Asterisk 1.6 version.
Here is output on CLI:

[Mar 19 16:14:48] WARNING[4841]: chan_sip.c:20115 set_insecure_flags: Unknown insecure mode ‘very’ on line 1605
[Mar 19 16:14:48] WARNING[4841]: chan_sip.c:20115 set_insecure_flags: Unknown insecure mode ‘yes’ on line 1622
[Mar 19 16:14:48] WARNING[4841]: chan_sip.c:20115 set_insecure_flags: Unknown insecure mode ‘yes’ on line 1641
[Mar 19 16:14:48] WARNING[4841]: chan_sip.c:20115 set_insecure_flags: Unknown insecure mode ‘very’ on line 1653

Please suggest what should we use instead?
Thanks in advance

By: admin

admin — Thu, 05 Mar 2009 09:09:30 +0000

Wishu, I don’t think if you can use such variables in sip.conf. However I’d like to suggest go for ARA solution and you have huge sip account list. using ARA you will feel easy to manage sip accounts from database in realtime.