[asterisk-users] Transfering not working - how to debug?
Rizwan Hisham
rizwanhasham at gmail.com
Thu Mar 29 07:27:45 MST 2007
Both end devices should be using same codecs. set dtmf = rfc2833 and set
canreinvite = no in sip.conf for both endpoints. This should solve the
problem. you should also check which codecs support rfc2833 for dtmf and
use that codec.
On 3/29/07, Gordon Henderson <gordon+asterisk at drogon.net> wrote:
>
> On Wed, 28 Mar 2007, Alan Chandler wrote:
>
> > I cannot seem to get any transfers to work at all. The console show I
> > have #1 amd #2 set up for Blind and Attended Transfer, but when I hit
> > these buttons on my handset nothing happens (other than I hear the dtmf
> > tones on the other end of the line).
> >
> > roo*CLI> show features
> > Builtin Feature Default Current
> > --------------- ------- -------
> > Pickup *8 *8
> > Blind Transfer # #1
> > Attended Transfer #2
> > One Touch Monitor *1
> > Disconnect Call * *0
> >
> >
> > I am using the tT options in my dial calls (via a macro)
> >
> > [macro-extension]
> > exten => s,1,Dial(${ARG1},20,tT)
>
> I had to fiddle with other things to make this work (needed for the
> Siemens CP4600 SIP/DECT phone)
>
> I found that the default timeouts were a bit tight for my likings (and the
> people who I was testing this with!)
>
> So in features.conf I have:
>
> transferdigittimeout = 8 ; Number of seconds to wait between digits
> when transfering a call
> featuredigittimeout = 999 ; Max time (ms) between digits for
> ; feature activation. Default is 500
>
> [featuremap]
> blindxfer => #1 ; Blind transfer
> atxfer => ## ; Attended transfer
> disconnect => #0 ; Disconnect
>
> If it's still not working, are you sure the DTMF is being picked
> up/transmitted correctly? If it's in-band, is it a codec other than G711?
> (which might give you problems)
>
> Gordon
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--
Regards
Rizwan Hisham
Software Engineer
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