Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike
try the new "dtmfmode" parameters on the user or peer. Note they are not currently valid in the "[general]" section. you can set dtmfmode=inband or dtmfmode=rfc2833 Mark On Sun, 9 Mar 2003, Mikael Andersson wrote:> > Hi.. > > I just wondering why DTMF are not recognized by aterisk on incoming calls > from my SIP provider ... > > ANy suggesteions ?` > > /Mike > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users at lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
At 14:39 2003-03-09 -0600, Mark Spencer wrote:>try the new "dtmfmode" parameters on the user or peer. Note they are not >currently valid in the "[general]" section. you can set dtmfmode=inband or >dtmfmode=rfc2833 > >Mark > >On Sun, 9 Mar 2003, Mikael Andersson wrote:Exactly where shoud I enter that value ? /regards Mike
Look into sip.conf.sample [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls ;tos=lowdelay ;tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registratio ; ;register => 1234@mysipprovider.com ; Register with a SIP provider ;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider as 1234 ; ;[snomsip] ;type=friend ;secret=blah ;host=dynamic ;dtmfmode=inband <<here is the answer ; Choices are inband, rfc2833, or info ;defaultip=192.168.0.59 ----- Original Message ----- From: "Mikael Andersson" <micke@party.pp.se> To: <asterisk-users@lists.digium.com> Sent: Monday, March 10, 2003 12:29 AM Subject: Re: [Asterisk-Users] DTMF detection on SIP provider ?> At 14:39 2003-03-09 -0600, Mark Spencer wrote: > >try the new "dtmfmode" parameters on the user or peer. Note they are not > >currently valid in the "[general]" section. you can set dtmfmode=inbandor> >dtmfmode=rfc2833 > > > >Mark > > > >On Sun, 9 Mar 2003, Mikael Andersson wrote: > > Exactly where shoud I enter that value ? > > > > /regards Mike > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
Hmm... haven't been able to get this to work on my Cisco ATA-186. Perhaps I'm trying the incorrect knobs? I'm making outbound calls ATA-186->*->iconnecthere->PSTN. I've set my ATA-186 to these various settings: AudioMode: 0x00150015 AudioMode: 0x00250025 AudioMode: 0x00050005 (per the settings for "negotiated", "out-of-band" and "in-band" found on Cisco's notes http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/ataadmn/ata88sip/sip88ape.htm#1027044 I've set my peer with iconnect to dtmfode=inband, rfc2833, and info, and tested each with each of the three AudioMode settings on the ATA listed above. No changes across all of them, except when I switch to "inband only" on the ATA, my Asterisk server no longer shows the DTMF pending and DTMF sending messages on the console debug (obviously.) I sometimes hear slight snatches of DTMF tones as I press keys, but something is muting or snagging them from the audio stream and not allowing them to go all the way through. My CVS updated code is as of one hour ago. Has anyone made this work with the ATA-186 systems? Did I just miss the magic combination somewhere in there? JT>try the new "dtmfmode" parameters on the user or peer. Note they are not >currently valid in the "[general]" section. you can set dtmfmode=inband or >dtmfmode=rfc2833 > >Mark > >On Sun, 9 Mar 2003, Mikael Andersson wrote: > >> >> Hi.. >> >> I just wondering why DTMF are not recognized by aterisk on incoming calls >> from my SIP provider ... >> >> ANy suggesteions ?` >> > > /Mike
At 00:50 2003-03-10 +0100, Andre Bierwirth wrote:>Look into sip.conf.sample > >[general] >port = 5060 ; Port to bind to >bindaddr = 0.0.0.0 ; Address to bind to >context = default ; Default for incoming calls >;tos=lowdelay >;tos=184 >;maxexpirey=3600 ; Max length of incoming registration we >allow >;defaultexpirey=120 ; Default length of incoming/outoing >registratio >; >;register => 1234@mysipprovider.com ; Register with a SIP provider >;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider as >1234 >; >;[snomsip] >;type=friend >;secret=blah >;host=dynamic >;dtmfmode=inband <<here is the answer ; Choices are inband, >rfc2833, or info >;defaultip=192.168.0.59Well.. But I need it on : the [general] part where I do the register ? or ? The "clients" in my case all my ATAs work fine.. But incoming calls doesnt.. /Mike
At 00:50 2003-03-10 +0100, Andre Bierwirth wrote:>;[snomsip] >;type=friend >;secret=blah >;host=dynamic >;dtmfmode=inband <<here is the answer ; Choices are inband, >rfc2833, or info >;defaultip=192.168.0.59Hm.. I think I need more help. My ATAs are working fine, asterisk detects all DTMF input. but on incomming call from my SIP provider, it doesnt seem to work. Any suggestions ? /Mike