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david_nedved at yahoo.com
Guest
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 Posted: Wed Mar 26, 2008 5:06 pm Post subject: [asterisk-users] DTMF suddenly stopped working on SIP channe |
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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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eric at fnords.org
Guest
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| Posted: Wed Mar 26, 2008 5:15 pm Post subject: [asterisk-users] DTMF suddenly stopped working on SIP channe |
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Inband only works with the ulaw and alaw codecs.
David Nedved wrote:
| Quote: | 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
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dhartman at djhsolutio...
Guest
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| Posted: Wed Mar 26, 2008 5:24 pm Post subject: [asterisk-users] DTMF suddenly stopped working on SIP channe |
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David Nedved wrote:
| Quote: | 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 |
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david_nedved at yahoo.com
Guest
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| Posted: Thu Mar 27, 2008 10:13 am Post subject: [asterisk-users] DTMF suddenly stopped working on SIP channe |
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--- Eric Wieling <eric at fnords.org> wrote:
| Quote: | Inband only works with the ulaw and alaw codecs.
|
I think you might be onto something here. I don't have any explicit
allow or disallow lines, just taking the defaults. I've got plenty of
bandwidth and CPU, I'm much more concerned about calls going through.
Without knowing what codecs my provider uses and not seeing anything
specific in the logs, is there a setting that would be better than
default for reliability?
I had originally set to inband for outgoing calls because the default
wasn't working for dialing into voicemail systems, etc. Switching to
inband fixed the outgoing DTMF issue and incoming worked fine for
months until earlier this week.
Thanks for any suggestions.
Best regards,
David
david_nedved at yahoo.com
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