11:23 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] T.38 detection/redirect in OpenSIPS
right, that is exactly what the b2b is up to do - to be able (at
signalling level) to manipulate the call legs
Regards,
Bogdan
Brett Nemeroff wrote:
Bogdan
Iancu
Sent: Wednesday, March 17, 2010 11:23 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] T.38 detection/redirect in OpenSIPS
right, that is exactly what the b2b is up to do - to be able (at
signalling level) to manipulate the call legs
Regards,
Bogdan
Brett Nemeroff
Can OpenSIPS make routing decisions based on the SDP information in an INVITE?
Lets say I have the following config
PSTN - t.38 Gateway - OpenSIPS - UserAgent
I have a TN from the PSTN routed to the UserAgent, I'd like to provide a
service so the user can use the TN for both voice faxing.
If this is possible at all, I would suspect it might require a rather creative
home-grown scenario in the B2BUA module. Since the call is already
established, a simple proxy isn't going to be able to tear down the one side
and reestablish it with another endpoint.
- Jeff
On Mar 17, 2010,
Matt,
I am for sure probably wrong, but I think you would need Asterisk or
Variant to Determine that it is a Fax Call,
I dont think UAC's send T38 information without negotiating with the
other side who request that it is capable, then it brings you to Jeff's
answer.
See above.
Matthew S.
I don't think there is any way to do this without an RTP capable device in
the mix.
What you may be able to do is have asterisk detect that it's a fax, then
reject it if it is.. I don't know if you can do all that without answering
the call.
Then you can forward it back to the proxy if it is a
Hi Matthew,
you do not need to look into the media part, as you can spot the FAX
presence via the re-INVITE with T38 codec in SDP (you can detect it from
opensips cfg).
So, maybe using the b2b module for something like:
- allow the voice call to be setup via the b2b in a transparent way
Bogdan,
But at this point, you are now playing with a dialg that is already
connected to an endpoint. You'd need to drop the first call to
establish a new call with the reinvite. Right?
-Brett
On Mar 17, 2010, at 11:50 AM, Bogdan-Andrei Iancu bog...@voice-system.ro
wrote:
Hi Brett,
Brett
Question: Can you change RTP source/dest ip/port during a REINVITE?
-Matt
- Original Message -
From: Bogdan-Andrei Iancu bog...@voice-system.ro
To: OpenSIPS users mailling list users@lists.opensips.org
Sent: Wednesday, March 17, 2010 12:39:56 PM
Subject: Re: [OpenSIPS-Users] T.38 detection
right, that is exactly what the b2b is up to do - to be able (at
signalling level) to manipulate the call legs
Regards,
Bogdan
Brett Nemeroff wrote:
Bogdan,
But at this point, you are now playing with a dialg that is already
connected to an endpoint. You'd need to drop the first call to
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] T.38 detection/redirect in OpenSIPS
right, that is exactly what the b2b is up to do - to be able (at
signalling level) to manipulate the call legs
Regards,
Bogdan
Brett Nemeroff wrote:
Bogdan,
But at this point, you are now playing
On 17/03/10 17:15, Matthew S. Crocker wrote:
UA - PROXY 200 Ok (SDP/G711)
PROXY - PSTN 200 Ok (SDP/G711)
** RTP Established between UA PSTN (mediaproxy/rtpproxy ??) **
** Gateway detects fax tone and attempts to REINVITE to T.38 **
PSTN - PROXY INVITE (SDP/T38)
PROXY - PSTN 180 Trying
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