[asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy

[Ed Nuñez]

To configure the Cisco for RFC 2833 add the following line to the desired
dial-peer

dtmf-relay rtp-nte

Hope this helps.

Ed Nuñez



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric
"ManxPower" Wieling
Sent: Tuesday, June 26, 2007 11:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF doesn't work between Asterisk and Cisco
SIP Proxy

This is usually a Cisco issue.

You need to set the Cisco to use RFC2833 DTMF.  Check the Cisco docs.

tracinet wrote:
> Jason,
> I am at least having similar issues with rfc2833 DTMF:
> 
> http://bugs.digium.com/view.php?id=10058
> 
> 
> On 6/20/07, Jason Ma <realmj at gmail.com> wrote:
>>
>> Hi buddies,
>> I encountered DTMF issue when I tried to place call from x-lite to a
>> sip conference serice,here is the diagram.
>> X-lite---->Asterisk--->Cisco SIP proxy---->SIP Conference service
>>
>> The Call can be established,and I can hear from x-lite the prompt of
>> the conference,but when I input any digits,nothing happened,the
>> conference service did not recognize my input.At the same time,in the
>> log of asterisk,I can find that asterisk recognized all the
>> digits....I tried "rfc2833","inband","info" in the "dtmfmode"
>> parameter,but did not work ,I'm not sure whether asterisk send the
>> right dtmf to cisco proxy,how can I track that?
>>
>> I made another test,dialing from x-lite registered with Cisco proxy to
>> voicemail service of Asterisk.
>> x-lite---->Cisco SIP proxy---->Asterisk--->Voicemail service
>>
>> Both the call and dtmf worked fine,I can input my mailbox number and
>> password and listen my  voicemail.both "rfc2933" and "inband" worked
>> in this situation,but not "info".
>>
>> My Asterisk is 1.4.4 with asterisk now,I did not configure dtmfmode in
>> the section of  xlite and the trunk to cisco proxy,just configure the
>> dtmfmode in sip.conf.
>>
>> When I used "rfc2833",I can see the log in asterisk as :
>>
>> [2007-06-19 16:01:40] DTMF[8925] channel.c: DTMF begin '2' received on
>> SIP/9999-08269470
>> [2007-06-19 16:01:41] DTMF[8925] channel.c: DTMF end '2' received on
>> SIP/9999-08269470, duration 160 ms
>> [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF begin '1' received on
>> SIP/9999-08269470
>> [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF end '1' received on
>> SIP/9999-08269470, duration 140 ms
>>
>> and when I used "inband",I can see :
>>
>> [2007-06-19 15:55:21] DTMF[8852] channel.c: DTMF end '2' received on
>> SIP/9999-09d916c0, duration 0 ms
>> [2007-06-19 15:55:22] DTMF[8852] channel.c: DTMF end '1' received on
>> SIP/9999-09d916c0, duration 0 ms
>>
>> Is that right?Can I check what digits that asterisk sent out ?
>>
>> How can I track where is wrong with the dtmf?Did asterisk send dtmf to
>> Cisco proxy correctly?
>> I really have no idea about that.Please advise.Thank you very much!!!!
>>
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> 
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