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

[David Nedved]

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.

Best regards,

David

david_nedved at yahoo.com


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