Martin Joseph wrote:
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
Hi Marty, can you expand on the "port issue" a bit. I will admit that
my understanding of sip connection handling is still a bit weak yet.
I can say that the VOIP provider is on the far side of a firewall from
the asterisk box and seems to work OK when talking to an old phone on a
Digium connection. Also the softphones are on the same side of the
firewall as the asterisk box.
Is this a case that asterisk is trying to directly connect the VOIP sip
connection and the softphone sip connection to each other or do both
connect to asterisk and it manages the data flow between the two?
So, I'm not sure how an IAX softphone would help other than forcing
asterisk to be between the voip sip and the softphone iax connection?
Again, thanks for any thoughts or suggestions as to how to get this to
work right.
Mike
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