Thanks for the reply. All of the SIP phones and the Asterisk server are
on the local network (192.168.1.x) on the same side of the router (and
yes the router does have a NAT firewall). I would think the XLite to
XLite would work but it doesn't (yet). I am seeing errors on the
console when the SIP phone goes out the IAXTel line - it is "Unable to
find a path from GSM to G723". Would it do anything to disallow G723 in
the iax.conf file (and allow GSM)?
On Sunday, May 16, 2004, at 02:34 PM, John Todd wrote:
At 9:11 PM -0500 on 5/15/04, Ben Witso wrote:
Hi- let me start off by saying I'm a newbie to Asterisk and this list
and I'll also apologize up front for stupid questions.
I have Asterisk running and 2 SIP phones (X-Lite) plus an iaxtel
gateway set up. I used the configurations from the O'Reilly article
and I haven't even set up voice mail (the only change was to add the
iaxtel entry). My problem is the audio out from my SIP phone isn't
reaching the destination phone (whether it is the other SIP phone or
a PSTN/POTS phone that I am calling through the iaxtel gateway). The
call makes it through OK and audio comes back from the remote end.
The audio looks like it is leaving the SIP phone - the level meter is
reacting to the speech - but it doesn't reach the other end. Also - I
have tried turning off some of the codecs in the X-Lite phone to see
if it was a codec compatibility problem with no success. Any ideas?
Is this likely a SIP phone setup problem? Or a config setting?
(something in the echo canceling maybe??)
TIA for any help.
Ben
(Once this is set up I might have to re-record all the prompts in the
en-us-mn dialect as mentioned earlier) :)
This looks like a NAT problem. Are all of your phones and Asterisk
servers on the same subnet? If not, I'd try removing NAT from the
equation until you get success, and then fiddling with the "nat=1"
settings in Asterisk for those SIP devices behind NAT. Hint: X-Ten is
a real pain to get working be hind NAT with Asterisk, I've found.
JT
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