[Asterisk-Users] Problem transfering incoming calls

Anton Krall akrall-lists at intruder.com.mx
Sun Mar 20 15:23:18 MST 2005


My ata uses dtmf=info and my sip.conf uses dtmfmode=rfc2833.

Do they have to match? Weird thing is, when making calls, transfer prompt
works, but no for incoming. 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of C F
Sent: Domingo, 20 de Marzo de 2005 03:56 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Problem transfering incoming calls

looks like an dtmf mode setting problem, make sure you have it set to
dtmfmode=rfc2833 or dtfmmode=info in sip.conf, the same goes for your ata.


On Sun, 20 Mar 2005 15:29:18 -0600, Anton Krall
<akrall-lists at intruder.com.mx> wrote:
> Guys.
> 
> Im having a big problem transfering incoming calls thru zap channels 
> to some other extension. If the call is made by me to the outside via 
> zap channels, no problem, hitting # gets me the transfer prompt, but 
> if the call comes in thru zap and eventhough I am sending the call 
> from the zap channel to my sip ata (GS ata 286) using Dial with wtWT 
> as parameters, when hitting # I don't hear the prompt.
> 
> Any ideas what might be wrong?
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users





More information about the asterisk-users mailing list