On Tue, Oct 04, 2005 at 09:59:56PM +0200, Olle E. Johansson wrote:
I think this is a bug. Please open a report in the bug tracker,
attaching all the requested information. If a re-invite fails, we should
not cancel the call. I am afraid that is exactly what is happening here
and would like to
On Thu, Sep 29, 2005 at 08:54:42PM -0500, Kevin P. Fleming wrote:
Ray Van Dolson wrote:
Our SIP/PSTN gateway provider seems to think that Asterisk should initiate
a
renegotiation to G711 when it sends the 488 message rejecting T38.
This is not correct. The 488 response 'cancels' the
Ray Van Dolson wrote:
On Thu, Sep 29, 2005 at 08:54:42PM -0500, Kevin P. Fleming wrote:
Ray Van Dolson wrote:
Our SIP/PSTN gateway provider seems to think that Asterisk should initiate
a
renegotiation to G711 when it sends the 488 message rejecting T38.
This is not correct. The 488 response
On Tue, Oct 04, 2005 at 09:59:56PM +0200, Olle E. Johansson wrote:
1. Asterisk sends the initial INVITE (requesting G711u)
2. SIP/PSTN gateway says it's trying (100) and its media server begins
sending
G711U RTP traffic.
3. SIP/PSTN gateway sends a 183 session progress message with
Opened bug #5384.
http://bugs.digium.com/view.php?id=5384
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Disclaimer: Yes, I know faxing over G711 is unreliable. :-)
We're running Asterisk 1.0.9 which talks to a Audiocodes SIP Gateway. We're
running Sipura SPA-2002's as ATA's and faxing within our own voice network is
working. If we try and fax out to the world however, we're running into a
Ray Van Dolson wrote:
Our SIP/PSTN gateway provider seems to think that Asterisk should initiate a
renegotiation to G711 when it sends the 488 message rejecting T38.
This is not correct. The 488 response 'cancels' the INVITE, so no codec
change was ever actually involved. The gateway should
