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

Ian asterisk at iancoetzee.za.net
Tue Mar 4 23:55:06 CST 2008


Hi Raul

Thank you, that would be apreciated, btw I have been talking to the 
users, and they say the "*2" transfers are easier than the built in 
grandstream transfers. I also forgot to state that I had to set the 
Grandstream to send DTMF via sip info, or else it will only work some of 
the time.

Ian

Raúl Gómez C. said the following on 04-Mar-08 03:33 PM:
> Hi Ian,
>
> I will try this workaround, I'll be trying to get this to work with 
> the "chan_local" solution, if I have success I'll let you know, thanks...
>
> On Wed, Mar 5, 2008 at 2:37 AM, Ian <asterisk at iancoetzee.za.net 
> <mailto:asterisk at iancoetzee.za.net>> wrote:
>
>     Hi Raul
>
>     I have bypassed my Grandstream's transfer function, by enabling
>     "*2" transfers in features.conf, and setting "canreinvite=no" in
>     sip.conf
>
>     Hope this helps you
>
>     Ian
>
>
> -- 
> 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/20080305/c40e9835/attachment.htm 


More information about the asterisk-users mailing list