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
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