|Summary:||ASTERISK-03969: No DTMF detection from PSTN endpoint on outbound calls|
|Reporter:||rod bacon (rod bacon)||Labels:|
|Date Opened:||2005-04-21 20:55:36||Date Closed:||2011-06-07 14:10:33|
|Description:||On digital connections (using TE405P), DTMF is not detected from the PSTN endpoint for calls placed FROM Asterisk. On inbound calls it is fine. I tested on analogue FXO channels, and problem does NOT exist.|
|Comments:||By: Kevin P. Fleming (kpfleming) 2005-04-21 21:26:01|
Note that we originally suspected that this problem was related to creating calls via pbx_spool, but we are now getting reports that it is much more pervasive.
By: petersv (petersv) 2005-04-22 01:38:56
We are seeing this problem. We tried to instrument Asterisk with debugging code a while back. We got as far as seeing that ast_dsp_process in zt_read in chan_zap did not detect the dtmf tones. Unfortunatly we ran out of service time.
In our case the problem is present for all outgoing call legs on all our isdn E1 links. Dtmf detection works on outbound legs that are sip (we do not use inband signalling on sip). The problem exists regardless of whether Asterisk is cpe or net.
edited on: 04-22-05 01:41
By: Matthew Fredrickson (mattf) 2005-04-22 01:44:48
Does this occur in stable as well?
By: petersv (petersv) 2005-04-22 02:02:00
I do not know about stable, hopefully someone else will.
The last version we used that works is HEAD as of 2004-12-12. The next version we tried was 2005-03-18 which did exhibit the problem.
By: Kevin P. Fleming (kpfleming) 2005-04-22 08:16:42
Were you able to isolate whether the problem is caused by changes in Asterisk or Zaptel?
By: petersv (petersv) 2005-04-22 08:39:32
No, we ran out of time to test and had to revert to a working version.
We did not have time to backtrack and try to find the specific version where dtmf broke.
Does dtmf detection on outbound isdn links work for anyone on head?
By: Michael Jerris (mikej) 2005-04-22 11:09:17
If somone can detail specific steps to reproduce, I can do some detective work and isolate when it broke.
By: petersv (petersv) 2005-04-22 14:53:21
We see the problem on the following setup:
ISDN PRI over E1 to a pbx (asterisk is net end, the pbx is cpe).
The problem manifests itself with the following dialplan:
exten => 205, 1, Dial(Zap/g2/205,20,grt)
exten => 205, 1, Dial(Zap/g2/205,20,grtT)
In both cases the called user (exten 205) cannot transfer the call. In the latter case the caller _can_ transfer the call. The following line _does_ allow the called user to transfer the call:
exten => 305, 1, Dial(SIP/petersv,20,grt)
You can contact me at "psvasterisk at psv dot nu" if you want more information on our setup.
edited on: 04-22-05 14:54
By: rod bacon (rod bacon) 2005-04-26 18:47:00
We are using a .call file to call a PSTN (via ISDN) party, then connecting them to an extension that plays a sound file and gives simple menu options (see below).
exten => 1005,1,Answer
exten => 1005,2,DigitTimeout,2
exten => 1005,3,ResponseTimeout,5
exten => 1005,4,Playback(test/test_intro)
exten => 1005,5,Background(test/test_options)
exten => 1,1,Playback(test/test_newYork)
exten => 1,n,Wait(1)
exten => 1,n,Goto(1005,5)
exten => 2,1,Playback(test/test_london)
exten => 2,n,Wait(1)
exten => 2,n,Goto(1005,5)
If we dial IN to 1005 (ORIGINATE the call from PSTN) things work OK. The problem only manifests itself when Asterisk originates the call. Debugging at the console shows no DTMF detection whatsoever.
By: Mark Spencer (markster) 2005-04-26 23:44:16
Please contact Digium technical support for hardware related technical support requests such as this. Thanks.
By: rod bacon (rod bacon) 2005-05-02 19:56:20
The problem does NOT appear in CVS Stable (Zaptel, Libpri, Asterisk), only CVS HEAD.
By: petersv (petersv) 2005-05-03 00:47:33
I have not heared back from Digium on this issue, except for an automated response.
By: Mark Spencer (markster) 2005-05-04 14:44:54
What is your digium support ticket number?
By: Mark Spencer (markster) 2005-05-04 14:58:39
Just e-mail me off the bug tracker with your support ticket id <firstname.lastname@example.org> and I'll find out what's going on.