[asterisk-users] DTMF suddenly stopped working on SIP channel
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 [EMAIL PROTECTED] Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF suddenly stopped working on SIP channel
Inband only works with the ulaw and alaw codecs. 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 > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF suddenly stopped working on SIP channel
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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF suddenly stopped working on SIP channel
--- Eric Wieling <[EMAIL PROTECTED]> wrote: > 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 [EMAIL PROTECTED] Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
