[Asterisk-Users] SIP to SIP connection problem

[M.Hockings]

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.

Mike