asterisk users - May 2004 - Newbie question-no outgoing audio

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) :)

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
>