[asterisk-users] problem transferring calls some of the times

[Ian]

Hi Raul

Raúl Gómez C. said the following on 05-Mar-08 07:40 PM:
> Ian,
>
> I'm unable to transfer calls using *2, I'm not sure why. Here's my 
> configs:
<snip>
>
> In the phones the "/Send DTMF:"/ is set to "in-audio" and "via SIP INFO"
It should only be set to SIP INFO, or else the audio comes out too 
choppy (in my case though) for asterisk to recognize it.

It took me about half a days experimenting with sip debug enabled on the 
channel to find that out, btw if you take longer than the 500ms feature 
timeout, asterisk still forwards  the DTMF to the other channel.

Regards
Ian
>
> What I'm missing here?
> -- 
> Raul
> Linux Counter #156439
> ------------------------------------------------------------------------
>
> _______________________________________________
> -- 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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080306/d1b2b0ec/attachment.htm