Re: [asterisk-users] No Audio for Extension to Extension

2007-01-23 Thread Tim Panton


On 22 Jan 2007, at 07:28, Troy - Purple Oranges wrote:


I am at a loss, I can terminate and receive calls via any of my
providers with both IAX and SIP.  I use GSM, G729a, and ulaw for those
carriers.

If I make an extension to extension call - there is no audio at all in
either direction.

All my extensions are set to use G729a (I have tried changing that
though to see if it would fix it).  I am fairly sure it is not a
transcoding issue - as the server transcodes for the inbound/outbound
calls.



You really need to tell us more!
At a pure guess however I'd say you have SIP extensions with canreinvite
set to true. Your internal network however does not permit rtp  
traffic between

the handsets.

Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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Re: [asterisk-users] No Audio for Extension to Extension

2007-01-23 Thread Marco Mouta

enable rtp debug in your asterisk CLI and check if there's traffic passing.
Would be a first approach I think.

On 1/23/07, Tim Panton [EMAIL PROTECTED] wrote:



On 22 Jan 2007, at 07:28, Troy - Purple Oranges wrote:

 I am at a loss, I can terminate and receive calls via any of my
 providers with both IAX and SIP.  I use GSM, G729a, and ulaw for those
 carriers.

 If I make an extension to extension call - there is no audio at all in
 either direction.

 All my extensions are set to use G729a (I have tried changing that
 though to see if it would fix it).  I am fairly sure it is not a
 transcoding issue - as the server transcodes for the inbound/outbound
 calls.


You really need to tell us more!
At a pure guess however I'd say you have SIP extensions with canreinvite
set to true. Your internal network however does not permit rtp
traffic between
the handsets.

Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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[asterisk-users] No Audio for Extension to Extension

2007-01-22 Thread Troy - Purple Oranges

I am at a loss, I can terminate and receive calls via any of my
providers with both IAX and SIP.  I use GSM, G729a, and ulaw for those
carriers.

If I make an extension to extension call - there is no audio at all in
either direction.

All my extensions are set to use G729a (I have tried changing that
though to see if it would fix it).  I am fairly sure it is not a
transcoding issue - as the server transcodes for the inbound/outbound
calls.

Has anybody come across this before?

Regards. Troy


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