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
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
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
