[OpenSIPS-Users] RES: OpenSIPS with Asterisk

AsteriskGuide flavio at asteriskguide.com
Tue Dec 29 14:42:42 CET 2009

Hi Saeed, 


The calls coming from the SIP Proxy will land in the context defined in the
file sip.conf.  If you are using Asterisk Realtime, check the context you
have used in the sip_buddies SQl view. If you are not using Asterisk
realtime, check the general context. 




Flavio E. Goncalves



De: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] Em nome de Saeed Akhtar
Enviada em: Tuesday, December 29, 2009 10:17 AM
Para: OpenSIPS users mailling list
Assunto: [OpenSIPS-Users] OpenSIPS with Asterisk


hi all,

  I'm configuring OpenSIPS with Asterisk. I used
seturi("sip:2001 at ASTERISK_IP:ASTERISK_PORT"); to forward my call to
Asterisk. Now Asterisk receives the call but shows a message that it can't
transfer call to my extension because it says call from ' ' to '2001' cannot
transfer because extension 2001 not found. extension 2001 is already there
in default context. I think problem is that from extension is empty. Can
anyone guide me what exactly is problem. Thanks


Saeed Akhtar

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.opensips.org/pipermail/users/attachments/20091229/91dad1c1/attachment.htm 

More information about the Users mailing list