Summary:ASTERISK-04858: Called party of outbound calls cannot enter DTMF
Reporter:Justin Hawkins (tardisx)Labels:
Date Opened:2005-08-19 01:43:05Date Closed:2011-06-07 14:10:20
Versions:Frequency of
Description:If a call is initiated by asterisk (via placing of a call file in the outbound spool directory) and the user placed into a context where DTMF is required (say via authenticate), asterisk cannot 'hear' the DTMF, and thus the user cannoy proceed.

If the user dials in and navigates to the same context - it does work.


I conversed privately with Kevin Fleming after seeing this post: http://lists.digium.com/pipermail/asterisk-users/2005-August/119661.html and he indicated the bug was fixed, but possibly not backported to the 1.0.x stream.

I duly checked out CVS head via the instructions at http://www.asterisk.org/index.php?menu=download and tried again - same result.

Comments:By: Mark Spencer (markster) 2005-08-19 01:51:28

I'm guessing this is specifically with a SIP channel and does not affect any other channel?

By: Justin Hawkins (tardisx) 2005-08-19 02:06:06

Yes sorry.

I had a much more detailed report which my web browser threw away when I didn't fill out a field and hit back :-|

Do you want me to try against a Zap channel?

By: Mark Spencer (markster) 2005-08-19 02:14:44

Yes, please do.  I believe this is really a SIP configuration issue of some sort.

By: Justin Hawkins (tardisx) 2005-08-19 02:20:56

Tested - outbound via Zap channel was successful; called party could send DTMF.

By: Justin Hawkins (tardisx) 2005-08-19 02:24:44

Just in case it's relevant (I didn't believe so given Kevin's message), my platform is FreeBSD 5.4-RELEASE, with both asterisk 1.0.9 stable and CVS HEAD (as mentioned).

By: Michael Jerris (mikej) 2005-08-19 08:55:44

can you get with a bug marshall on irc #asterisk-bugs to discuss this.  I do think this is likely a config issue on the sip side.  If you can't get to IRC, we need to have details of your sip phone, a full sip debug and verbose, as well as see your sip config for the peer.

By: Justin Hawkins (tardisx) 2005-08-23 01:33:40

I'm interested that you believe it's a configuration issue. I'm certainly
open to the idea, I just didn't think it was originally given the post
on the mailing list.

Since I work for the voip provider in question (:-) I will investigate from
that point of view over the next day or so and see if I can fix it.

If no go I will provide the debugging info.


By: Justin Hawkins (tardisx) 2005-08-23 22:50:12

Hello folks,

Sorry to lead you up the garden path, it was indeed a configuration issue.

Gory details - the voice peer on the cisco router used for outbound calls was configured with the wrong dtmf-relay type.

Sorry to have wasted your time.