[asterisk-users] The call is established but without exchanged voice packets

[Salman Zafar]

Choose suitable NAT settings from sip.conf

turn direct media in sip.conf or per peer off


On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed <asabatgirl at hotmail.com>wrote:

> Hello,
>
> I am trying to make my first call on Asterisk to succeed. I have Asterisk
> 1.8.10.1 running on Ubuntu machine.
> The configuration is quite simple just for my first test, Trying to have a
> call between two X-lite sipphone. The subscribers succeeded to register and
> the call is established, but still no voice can be heard, and lead the
> call to be disconnected after! By checking the logs, I can see this
> chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on
> transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2
> (Critical Response)
>
> Here's my  simple sip configuration
> [general]
> context=internal
> allowguest=no
> allowoverlap=no
> bindport=5060
> bindaddr=0.0.0.0
> srvlookup=no
> disallow=all
> allow=ulaw
> alwaysauthreject=yes
> canreinvite=no
> nat=yes
> session-timers=refuse
> externip=<IP>
>
> [7001]
> type=friend
> host=dynamic
> secret=123
> context=internal
>
> [7002]
> type=friend
> host=dynamic
> secret=456
> context=internal
>
> A snoop capture  for my call is uploaded in the following link. I wonder
> if there is any missing configuration or plugin need to be set here!
>
> http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992 <http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992>
> Thanks.
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Regards

**************************
Muhammad Salman
***************************
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130919/c03a7f03/attachment.htm>