[asterisk-users] DTMF suddenly stopped working on SIP channel

[Darrick Hartman (lists)]

David Nedved wrote:
> Hi All,
> 
> Anyone have any idea what could cause incoming calls on a SIP channel
> to no longer be able to use DTMF?  DTMF on incoming calls on zaptel and
> on local SIP softphones and ATAs all work fine.  Nothing gets
> registered in the CDR or on the console in verbose level 10, it just
> times out.  I haven't changed anything on my part and can't get through
> to Viatalk tech support to ask them what they changed (fat load of luck
> getting that question answered anyway).  Everything was working fine
> with dtmfmode=inband and relaxdtmf at the default, now I've tried all 6
> valid combos of those two settings with no change.  This is on asterisk
> 1.2.27 that's been working fine in production for about 3 months now.
> 
> Here's the section from sip.conf (the way it had been working all
> along):
> 
> [viatalk]
> type=peer
> secret=(yep it's right)
> username=(yep it's right)
> host=newyork-1.vtnoc.net
> canreinvite=no
> insecure=very
> qualify=yes
> context=incoming-viatalk
> dtmfmode=inband         ; Choices are inband, rfc2833, or info
> ;relaxdtmf=yes                  ; Relax dtmf handling
> 
> Thanks in advance for any help.  I've got all incoming calls on Viatalk
> shunted to an extension in the meantime, not an elegant solution.
> 

Do yourself a favor and upgrade a Asterisk 1.4 which has a proper 
implementation of DTMF.  It's likely your SIP provider upgraded to 
something which does not recognize the DTMF tones from Asterisk 1.2.

Darrick
-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com