Summary:ASTERISK-02110: DTMF stops working in Voicemail
Reporter:bfranks (bfranks)Labels:
Date Opened:2004-07-26 11:02:09Date Closed:2011-06-07 14:10:13
Versions:Frequency of
Description:Occasionally DTMF will stop functioning, making the user end the current voicemail session and having to log back in.  I can not trigger it by repeating any processes, but it just seems to break.  The Voicemenu's will continue or it will wait for your input.


This occurs using DTMF rfc2833.

I also posted to the users list to make sure it was not a Polycom IP 500 issue, and others with Cisco SIP phones report same symptoms.
Comments:By: qo027 (qo027) 2004-07-26 11:17:52

I had this same problem with Pingtel xpressa phones as well. I changed the Pingtel to work Inband only (Pingtel send both RFC2833 and Inband by default) and Voicemail works. While looking into the problem I used Ethereal to capture the SIP/RTP packets and I can see that DTMF was being passed to the Asterisk box via RFC2833 but Asterisk wasn't handling the packets. Sometimes the Voicemail session would accept some DTMF and then stop working and others it wouldn't work at all. In addtion when I configured the Pingtel for RFC2833 only (i.e. no Inband) all DTMF stopped working and I couldn't make a call or access the voicemail application.

By: Mark Spencer (markster) 2004-07-26 16:38:21

I think this is going to take some SSH time during a call when I can attach gdb and try to poke around and see what's going on.

By: Mark Spencer (markster) 2004-07-26 16:38:40

Does it have to do with putting it on hold and then taking it back off hold?

By: bfranks (bfranks) 2004-07-26 17:00:12

Nope to the hold.  It might cause the same symptoms, but I know this will happen even if teh call isn't placed on hold.  Typically the messages are pretty long for our setup (sometimes greather than 1.5 mins), so they aren't short messages.

- Brent

By: qo027 (qo027) 2004-07-26 19:32:22

Hold didn't make a difference for me either. To be honest I haven't tried this again since HEAD 07-16-2004, so I'm going to try again in the morning. I has already planned on trying again in the morning since one of my technicians informed me this afternoon that they corrected a network negotiation problem with our VLAN router and this could be the cause for delayed packets. (Most likely not since the phone and the Asterisk PBX are on the same VLAN - but I figured due diligence required another look.)

Excuse my ignorance... when you say SSH time, I assume you mean with Brent's or my Asterisk system, is that so? If so, our Asterisk server is internal and is hidden behind about 4 layers of firewalls BUT... I can give you what is called a webtop access to the server ... it is actually a terminal connection but through a web page and displayed in an independent windows... unfortunately fonts can get a little screwed. Would this be OK?

By: Mark Spencer (markster) 2004-07-26 23:02:26

Yes, it means I will have to be able to ssh into the system and watch the problem in progress, attach gdb, and generally make a muck of your system for a few minutes while I try to figure out how this is happening.

By: Brian West (bkw918) 2004-07-27 09:16:49

I can report that this isn't a problem with my sipura or my 7960's with the latest cvs.

By: bfranks (bfranks) 2004-07-27 09:24:55

I would also report that this only happens 1 out of evey 100 times...

Doesn't happen very often.  Seems to happen to my users about once or twice a month.  15 users on the system.

- Brent

By: jesses (jesses) 2004-07-27 14:24:47

Just for kicks can you paste the section of extensions.conf that relates to the extension that users dial to retrieve voicemail... I have about 30 users using SIP polycom IP 500 rfc2833 and I've never had this problem

By: bfranks (bfranks) 2004-07-27 22:53:31

Sure, But I don't think this is it.  It would be more likely a sip.conf issue... So I will paste a SIP Peer as well..  If anything, I would have thought it was the Polycom IP 500 XML config files, however with users on the list responding to my thread Subject: DTMF Stops working in voicemail, (using Cisco and Grandstream), I doubt it's an extensions issue..

Anyways here goes:

; VoiceMail Declarations
; 8000 - Huntingdon Voicemail   (Extension)
; 8500 - Huntingdon Voicemail   (Extension + PIN)
; 8888 - Duncansville Voicemail (Extension + PIN)
exten => 8000,1,Wait(1)
exten => 8000,2,VoicemailMain(${CALLERIDNUM})
exten => 8000,3,Hangup
exten => 8500,1,Answer
exten => 8500,2,Playback(beep)
exten => 8500,3,Wait(3)
exten => 8500,4,VoicemailMain
exten => 8500,5,Hangup
exten => 8888,1,Dial(IAX2/hunasterisk:1111@
exten => 8888,2,Hangup

sip.conf peer definition:

callerid="Charlie" <2132>

By: jesses (jesses) 2004-07-28 00:30:33

Is 8000 the extension people are calling to check their messages? My SIP phones have a problem if I don't have an answer() as the first step

By: bfranks (bfranks) 2004-08-01 22:45:17

Added answer to the statement.  Only time will tell if this makes a difference.  Unfortuantely it may take a month or two to find out whether or not it is fixed as my users complain this only occurs occasionally.

- B

By: bfranks (bfranks) 2004-08-03 07:23:23

I added the Answer statement, and a user said it happened to them.

I also found out (*I think*) that this may deal with when users forward & prepend messages.  After they prepend, then *sometime* DTMF input stops working.

- Brent

By: Olle Johansson (oej) 2004-08-14 15:41:19

Any more information? Does this still happen with latest CVS?
Have you confirmed the relation to forward and prepend?


By: Brian West (bkw918) 2004-08-15 19:49:33

Well Answer has NOTHING to do with it.

because it has:

       if (chan->_state != AST_STATE_UP)

Thus will answer no matter what.


By: bfranks (bfranks) 2004-08-15 23:45:56

Can also now confirm that prepending / forwarding message has nothing to do w/ it.  It will stop working if just listening to messages.

I will try to duplicate and see what is happening by running sip debug and strace.

- Brent

By: Brian West (bkw918) 2004-08-22 22:58:55

If this is still something you can duplicate please find me on IRC and we will reopen this...  We are nearing 1.0 release and need faster response times from bug reporters.

bkw_ on #asterisk @ irc.freenode.net