Summary:	ASTERISK-04858: Called party of outbound calls cannot enter DTMF
Reporter:	Justin Hawkins (tardisx)	Labels:
Date Opened:	2005-08-19 01:43:05	Date Closed:	2011-06-07 14:10:20
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/Interoperability
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
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. ****** ADDITIONAL INFORMATION ****** 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. Cheers. 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.