[Asterisk-Users] SIP to SIP connection problem

[Martin Joseph]

On Jun 7, 2006, at 6:55 PM, M.Hockings wrote:

> I have a small asterisk setup here with one POTS line, one VOIP SIP 
> connection an FXS connection to the house phones and a bunch of 
> softphones.  Local calls are routed out through the POTS line and long 
> distance through the VOIP line.  This works great for the old house 
> phones but the softphones on the computers can only make local calls. 
> That is any attempt to connect through the VOIP line end in silence as 
> soon as the called party picks up and asterisk attempts to connect the 
> VOIP SIP connection and the softphone SIP connection.  This is using 
> xTen softphones on Linux and Windows.
>
> I was thinking that it might have to do with mismatched codecs or some 
> such?  In the [general] section of the sip.conf I see that freePBX has 
> put
>
> disallow=all
> allow=ulaw
> allow=alaw
>
> and none of the softphone definitions set any different requirements.
>
> If I connect a softphone directly to the VOIP provider it appears to 
> use the g711u codec.
>
> This is all using asterisk 1.2.9.1 and freepbx 2.1.1 running on CentOS 
> 4.3.
>
> Thanks for any suggestions.
>
Sounds more like a port issue to me.  Looking in the asterisk Console 
and setting verbosity up when attempting these calls might give you 
more info.

Also,  you might try using an IAX softphone instead, as these are much 
less of a hassle in my opinion.  There are several available.

Marty