[OpenSIPS-Users] T.38 detection/redirect in OpenSIPS

[Bogdan-Andrei Iancu]

Hi Brett,

Brett Nemeroff wrote:
> I don't think there is any way to do this without an RTP capable 
> device in the mix.
you do not need to look into RTP as the FAX is advertised in the 
re-INVITE (in SDP) - so you can detect it from opensips script by 
inspecting the SDP of reINVITES
>
> 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.
no, you cannnot, as first the call is established (from sip point of 
view) as a simple audio call and after that re-negotiated (via 
re-INVITE) for FAX
>
> Then you can forward it back to the proxy if it is a fax with maybe a 
> prefix. 
>
> A lot of assumptions in there. Would like to hear if you find 
> something that works. Not sure if you can SIP Spiral yet in asterisk 
> anyway. ;)
I do not see the need of Asterisk - maybe with some changes, the b2b 
module will be able to handle this - see my prev email.

Regards,
Bogdan

> -Brett
>
>
> On Wed, Mar 17, 2010 at 10:51 AM, David J. <david at styleflare.com 
> <mailto:david at styleflare.com>> wrote:
>
>     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. Crocker wrote:
>     > 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.
>     >
>     > Voice call goes through normally (g.711 g.729 codec)
>     >
>     > Fax call starts off as a normal voice call (INVITE, 180, 183,
>     200).  Once the call is answered the originating end (PSTN) starts
>     sending fax tones. The Gateway hears the fax tones and attempts to
>     RE-INVITE with T.38 in the SDP.  I'd like OpenSIPS to see the T.38
>     capability in the SDP and redirect the call to a fax->e-mail
>     gateway.  So,  the 2nd INVITE comes in, OpenSIPS sends the INVITE
>     to the fax gateway and a BYE to the user.  The fax gateway does a
>     200 and negotiates T.38 with the PSTN gateway.
>     >
>     > I know I can route the call through Asterisk and have it do a
>     quiet answer and listen for the modem sounds.  I'd like to avoid
>     using Asterisk for all RTP traffic and only use it for the fax
>     gateway traffic (i.e. once it has been determined to be a fax
>     Asterisk steps in and handled the T38 -> E-mail)
>     >
>     > -Matt
>     >
>     >
>
>
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-- 
Bogdan-Andrei Iancu
www.voice-system.ro