[asterisk-users] Fax throughput - Asterisk 1.6.1.9

[Alex Balashov]

Set 'canreinvite=no' on all applicable peers?

Cyprus VoIP wrote:

> Hello,
> 
> We are trying to send faxes by T.38 protocol to a remote SIP proxy from 
> a local extension. The local extension sends the INVITE, Asterisk sends 
> the call to the Proxy the call is connected with a regular audio codec. 
> After a few seconds the remote proxy sends an INVITE with UDPTL and the 
> Asterisk sends it to the local extension and it's accepted, but (here 
> the problem starts) just after sending the OK with the proper SDP to the 
> remote Proxy, the Asterisk initiates a new INVITE to the local extension 
> and remote Proxy, with the normal audio codecs again.
> 
> We set "t38pt_udptl=yes" in sip.conf and allowed all the codecs to the 
> local extension and remote Proxy, but it still forces the call to go 
> back to a voice call.
> 
> Any idea why it happens and how to debug it? We set verbose and debug to 
> 20, but no "internal" info is provided to get a clear understanding on 
> Asterisk's "thoughts" during that process.
> 
> Thank you in advance for your assistance,
> 
> Andreas
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
Alex Balashov - Principal
Evariste Systems
Web     : http://www.evaristesys.com/
Tel     : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671