# DTMF isn't working **URL:** \ **Category:** Asterisk Support\ **Created:** [February 11, 2011, 9:56pm UTC](https://community.asterisk.org/t/dtmf-isnt-working/34128 "2011-02-11T21:56:59Z")\ **Posts on this page:** 7\ **Page:** 1
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [February 11, 2011, 9:56pm UTC](https://community.asterisk.org/t/dtmf-isnt-working/34128/1 "2011-02-11T21:56:59Z")
Hi, I am running asterisk 1.8.2 on CentOS 5. When I make a call from Voxeo to the cell phone, both DTMF and speech recognition are working. But when I make a call from the cell phone to Voxeo, I can hear prompts, but DTMF isn’t working. What am I missing here? Here’s my environment. No additional telephony hardware is involved. a cell phone \<–\>SIP Provider\<–\>Asterisk(NATed)\<–\>Voxeo (as a soft-phone) Here’s my sip.conf: [code]pedantic=no register =\> [1231231234@sip.provider.com](mailto:1231231234@sip.provider.com):pass:1231231234@sip.provider.com localnet=192.168.10.0/255.255.255.0 externaddr=123.123.123.111:5060 nat = comedia [[sip.provider.com](http://sip.provider.com)] type=peer context=inbound [host=sip.provider.com](http://host=sip.provider.com) [fromdomain=sip.provider.com](http://fromdomain=sip.provider.com) fromuser=1231231234 secret=pass username=1231231234 insecure=invite authname=1231231234 dtmfmode=rfc2833 dtmf=inband canreinvite=yes directmedia=yes [voxeo] type=friend host=dynamic secret=pass2 context=outbound dtmfmode=info dtmf=info deny=0.0.0.0/0 permit=192.168.10.0/255.255.255.0[/code] Here’s my extensions.conf: [code][inbound] exten =\> \_NXXNXXXXXX, 1, dial(SIP/voxeo,30) [outbound] exten =\> \_1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.provider.com, 30) [/code] ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [February 11, 2011, 11:49pm UTC](https://community.asterisk.org/t/dtmf-isnt-working/34128/2 "2011-02-11T23:49:19Z")
I made a few changes to sip.conf, made a few calls, and reviewed RTP traffic generated in the following cases. It looks like DTMF events with payload type being telephone-event (101) are working whereas those with payload type being DynamicRTP-Type-101 (101) are not. The payload type seems to have something to do with dtmfmode and dtmf property in sip.conf. How can I fix this problem? Thanks. - case 1 (I made a call from Voxeo to my cell phone with the same sip.conf shown in my first post. DTMF doesn’t work here.) - case 2 (I made a call from a cell phone to Voxeo.) With the following changes to sip.conf, DTMF works. ```auto [sip.provider.com] type=peer context=inbound host=sip.provider.com fromdomain=sip.provider.com fromuser=1231231234 secret=pass username=1231231234 insecure=invite authname=1231231234 dtmfmode=rfc2833 dtmf=inband canreinvite=yes directmedia=yes [voxeo] type=friend host=dynamic secret=pass2 context=outbound dtmfmode=rfc2833 dtmf=inband deny=0.0.0.0/0 permit=192.168.10.0/255.255.255.0 ``` ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [February 14, 2011, 12:04pm UTC](https://community.asterisk.org/t/dtmf-isnt-working/34128/3 "2011-02-14T12:04:48Z")
Which codec? In band DTMF will only work reliably for mulaw and alaw. ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [February 14, 2011, 4:05pm UTC](https://community.asterisk.org/t/dtmf-isnt-working/34128/4 "2011-02-14T16:05:20Z")
In my previous tests, I didn’t specify allow or disallow properties. So by default, allow =all was used. I tried different combinations of dtmfmode and dtmf values, and setting dtmfmode=auto and dtmf=auto for both voxeo and [sip.provider.com](http://sip.provider.com) gives me the best results, in which dtmf won’t work only if I initiate an outbound call from voxeo to my cell phone. DTMF is supposed to work regardless of which endpoint initiates a call, right? But why am I running into this problem? ---
**Author:** ![Helix26404](https://avatars.discourse-cdn.com/v4/letter/h/3d9bf3/32.png) [@Helix26404](https://community.asterisk.org/u/Helix26404)\ **Post date:** [February 15, 2011, 5:26am UTC](https://community.asterisk.org/t/dtmf-isnt-working/34128/5 "2011-02-15T05:26:11Z")
What codec(s) do you want to be able to use in your system? Once we know that, we can better recommend optimal settings/troubleshoot those settings if they aren’t functioning correctly. ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [February 15, 2011, 5:00pm UTC](https://community.asterisk.org/t/dtmf-isnt-working/34128/6 "2011-02-15T17:00:19Z")
for now I would like to use g711 PCMU. ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [March 13, 2011, 12:52am UTC](https://community.asterisk.org/t/dtmf-isnt-working/34128/7 "2011-03-13T00:52:33Z")
Hello, I have problem with dtmf on SIP [freecall.com](http://freecall.com) Asterisk 1.4.21.2~dfsg-3+lenny2 in SIP.config i set: `dtmfmode=rfc2833 dtmf=auto rfc2833compensate=yes` in extensions.conf i use SendDTMF(12345,250) and Asterisk system call to my phone, and I not hear DTMF code. in debug sip i have: ```auto ``` What can I do to make this work?