Resolved: No DTMF

[ksalter]

I am able to dial voicemail fine and navigate the menus, but when dialing another extension, or when dialing outbound using a SIP trunk, no DTMFs are being sent or received. I have verified the extension and trunk is setup to use dtmfmode=rfc2833. I have also tried two different phones - a PolyCom 301 and a Grandstream 2000.

Any ideas?

 

[jroper]

Hi

Can I try and give you reassurance that these symptoms in nearly every case relate to DTMF or Codecs, either between the phone and PBX - which you have checked via voicemail menus, or DTMF/Codecs between you and the carrier.

Sorry I cannot be much more help to you, but, I would suggest that you keep looking in that area.

Yours

Joe

 

[ksalter]

Thanks for the info.

However, I suspect the problem is deeper, because if I call extension to extension, I cannot not hear DTMFs from the phones. When I press DTMF on one phone, I can hear the audio cut out on the other phone, but no DTMF is played.

Both extensions are set to use RFC2833.

 

[ksalter]

Here is the debug I'm seeing:

[Dec 11 09:47:25] DEBUG[9526]: rtp.c:727 process_rfc2833: - RTP 2833 Event: 00000001 (len = 4)
[Dec 11 09:47:25] DEBUG[9526]: rtp.c:626 send_dtmf: Sending dtmf: 49 (1), at 10.52.19.110
[Dec 11 09:47:25] DTMF[9526]: channel.c:2419 __ast_read: DTMF begin '1' received on SIP/1008-082a8e50
[Dec 11 09:47:25] DTMF[9526]: channel.c:2429 __ast_read: DTMF begin passthrough '1' on SIP/1008-082a8e50
[Dec 11 09:47:25] DEBUG[9526]: channel.c:4103 ast_generic_bridge: Got DTMF begin on channel (SIP/1008-082a8e50)
[Dec 11 09:47:25] DEBUG[9526]: channel.c:4382 ast_channel_bridge: Bridge stops bridging channels SIP/1000-08263f68 and SIP/1008-082a8e50
[Dec 11 09:47:25] DEBUG[9526]: rtp.c:727 process_rfc2833: - RTP 2833 Event: 00000001 (len = 4)
[Dec 11 09:47:25] DEBUG[9526]: rtp.c:727 process_rfc2833: - RTP 2833 Event: 00000001 (len = 4)
[Dec 11 09:47:25] DEBUG[9526]: rtp.c:727 process_rfc2833: - RTP 2833 Event: 00000001 (len = 4)
[Dec 11 09:47:25] DEBUG[9526]: rtp.c:727 process_rfc2833: - RTP 2833 Event: 00000001 (len = 4)
[Dec 11 09:47:25] DEBUG[9526]: rtp.c:727 process_rfc2833: - RTP 2833 Event: 00000001 (len = 4)
[Dec 11 09:47:25] DEBUG[9526]: rtp.c:727 process_rfc2833: - RTP 2833 Event: 00000001 (len = 4)
[Dec 11 09:47:25] DEBUG[9526]: rtp.c:727 process_rfc2833: - RTP 2833 Event: 00000001 (len = 4)
[Dec 11 09:47:25] DEBUG[9526]: rtp.c:626 send_dtmf: Sending dtmf: 49 (1), at 10.52.19.110
[Dec 11 09:47:25] DTMF[9526]: channel.c:2356 __ast_read: DTMF end '1' received on SIP/1008-082a8e50, duration 100 ms
[Dec 11 09:47:25] DTMF[9526]: channel.c:2397 __ast_read: DTMF end accepted with begin '1' on SIP/1008-082a8e50
[Dec 11 09:47:25] DTMF[9526]: channel.c:2413 __ast_read: DTMF end passthrough '1' on SIP/1008-082a8e50
[Dec 11 09:47:25] DEBUG[9526]: channel.c:4103 ast_generic_bridge: Got DTMF end on channel (SIP/1008-082a8e50)
[Dec 11 09:47:25] DEBUG[9526]: channel.c:4382 ast_channel_bridge: Bridge stops bridging channels SIP/1000-08263f68 and SIP/1008-082a8e50

So it looks like the DTMF is detected from the source phone, but I don't hear it on the destination phone. And I know these phones work on my production PBX.

 

[merlyn]

I had the same problem with my cisco phone.

Try what worked for me.

Make sure the phone is up and working .. rerun genzaptelconf and make sure zaptel is setup ok. (its ok to run a second time it will not hurt anything)

reboot the asterisk pbx and reboot the phone.

wait a while and suddenly it should start working at least it did for me.

Merlyn

 

[ksalter]

Thanks, but didn't work.

However I figured this out.

I am using Vitelity for my oubound calls. If I am using PCMU for the codec, the DTMFs work. If I switch to G729, they don't. In order to get them to work with G729, I had to set dtmfmode to info in my trunk configuration. Now they work.

 

[ednunnemaker]

Similar DTMF problem

Can you elaborate on your fix for the people that aren't as experienced as you?

My problem: call comes in caller presses option 2 from the IVR the call then rings my desk, laptop and cell. If I answer the call on my cell I get a message that says please press 1 to accept this call, I press 1 and get nothing, the message repeats. This call is going out over a Les.net VOIP trunk. If the call goes out over a zap trunk it works fine. Any suggestions would be great. Thank you, Ed

 

[troycarpenter]

Ed,

I was just having the same problem with VoicePulse. When I used their configuration tool for FreePBX, it set up the trunk with the dtmfmode=info parameter. I had to change that to dtmfmode=rfc2833.

Check your trunk for that parameter and either set it to rfc2833 or delete it (the default is also rfc2833).

I use les.net for incoming calls and the trunk does not have the dtmfmode parameter. Incoming DTMF detection works just fine. Also check your configuration on les.net to see if you are set up correctly.

 

[darmock]

Can you elaborate on your fix for the people that aren't as experienced as you?

My problem: call comes in caller presses option 2 from the IVR the call then rings my desk, laptop and cell. If I answer the call on my cell I get a message that says please press 1 to accept this call, I press 1 and get nothing, the message repeats. This call is going out over a Les.net VOIP trunk. If the call goes out over a zap trunk it works fine. Any suggestions would be great. Thank you, Ed

This problem can also be related to the cell phone carrier, type of phone, what tower you are using, and what band is in use. I have run across this a number fo times with clients experiencing similar problems and unfortunately there is no easy solution.

For example I currently use a motorola razor phone which works just fine when pressing one to accept the call. I am using a pots line to do the outbound calls to the cell phone in this instance. However when I use my stanaphone line as the outbound trunk to my phone the one does not work. When I use a les.net line as outbound trunk it also works. Now when I switch over to my wifes cell a nokia on a different provider it does not work at all! I suspect that the regenerated dtmf tones are of insufficient duration.

The most stable config that seems to work the best is the use of a pots line as the outbound trunk to the cell phone. Using this method you can play with the relaxdtmf option that the zap cards use which *may* increase your recognition rate.

BTW on the wifes phone when it does not work using the phones DTMF signals I tried using a pure DTMF tones from a pocket dialer (aka radioshack ... yah I am an old guy) and it worked fine....
I had to hold the one key down for about 2 secs for it to be detected.

I know no solution sorry

Tom

 

[januskey]

I am having a similar issue in that I can access voicemail from grandstream phones. I have tried with both a gxp2000 and a bt100 with no luck. My extensions are set to dtmfmode=rfc2833 and I've tried all combinations of the the three options on the phone. If I use anything other than rfc2833, the phone doesn't even get a dial tone and on the gxp2000, it's repoting a 403 error. I'm using a new vmware image of piaf iwth all updates.


* Running Asterisk Version : Asterisk 1.4.21.2
* Asterisk Source Version : 1.4.21.2
* Zaptel Source Version : 1.4.12.1
* Libpri Source Version : 1.4.7
* Addons Source Version : 1.4.7
********************************************************************
CentOS release 5.2 (Final) - 32 Bit ** Kernel: 2.6.18-92.1.6.el5

Example extension...

[102]
type=friend
secret=SECRET
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=102@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/102
context=from-internal
canreinvite=no
callgroup=
callerid=device <102>
accountcode=
call-limit=50

Any suggestions would be much appreciated.

Also, the phones both work fine on an older version of piaf (I think 1.2.3).