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Summary:ASTERISK-07320: Dial with option D does not appear to work in 1.2.9.1
Reporter:geisj (geisj)Labels:
Date Opened:2006-07-10 14:57:14Date Closed:2011-06-07 14:07:22
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Applications/app_dial
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) dd.txt
( 1) debug.txt
Description:I have two SIP phones. I come offhook and
dial 57 which is defined as

exten => 57,1,Dial(SIP/222,,tTD(101))

SIP/222 rings, when I answer I do no hear the DTMF of 101.
I see on the screen the message of :
Sending DTMF '101' to called party. But I do not hear it.

This came up as my original purpose was to use this to select
intercom zones after dialing the intercom system and it answered.
That stopped working.

This is the easyest example I can find.

****** ADDITIONAL INFORMATION ******

I tried

exten => 57,1,Dial(SIP/101,,tTD(101:101))

and I hear the DTMF back on the originating phone.
I still do not hear it on the phone I answer.

Comments:By: Serge Vecher (serge-v) 2006-07-10 15:24:49

Ok, will need to see the console log with SIP debug of what's happening.

1) Prepare test environment (reduce the ammount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
  console => notice,warning,error,debug
3) restart Asterik.
4) Enable SIP transaction logging with the following CLI commands:
set debug 4
set verbose 4
sip debug
5) Save complete console log to file and _attach_ said file to the bug.
By: geisj (geisj) 2006-07-12 07:55:15

Was the debug.txt file what you needed? How can I get this to work?
By: Serge Vecher (serge-v) 2006-07-12 09:01:37

Yes, the log looks good. I do see "Sending DTMF '101' to called party"
what's the dtfmode setting for these peers in sip.conf?
By: geisj (geisj) 2006-07-12 09:12:43

For my SIP/402 it is a grandstream 101.
[402]
type=friend
username=402
secret=402
dtmfmode=RFC2833
host=dynamic
context=smvoice-sip
callerid="A 402 A 402" <402>
qualify=no
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm

I verified the setup on the phone and it is rfc2833.

For my 404 phone it is a linksys SPA922

[404]
type=friend
username=404
secret=404
dtmfmode=RFC2833
host=dynamic
context=smvoice-sip
callerid="A 404 A 404" <404>
qualify=no
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm

I have verified it is also rfc2833.


Both phones work just fine under normal operation.
DTMF on menues etc are accepted.

Jerry
By: Serge Vecher (serge-v) 2006-07-12 09:23:35

1. Just to rule out that this is not a device specific problem, can you please reverse the caller-callee roles in your test and see if it is still the callee that doesn't hear dtmf?
2. If above tests passes, let's turn on RTP debug in addition to SIP debug to see if RTP packets with DTMF are going out or not.
By: geisj (geisj) 2006-07-12 09:56:01

Very confused at this point.

When I reversed the call 402 dial 59 below (calls 404) I heard the DTMF 101

exten => 57,1,Dial(IAX2/devcentos64_to_unifiedpaging/528,,tTD(wwww101))
exten => 58,1,Dial(SIP/402,20,tTD(wwww101))
exten => 59,1,Dial(SIP/404,20,tTD(101))

I then tried 404 calling calling 58 and I do not hear the DTMF digits.

The whole question arose when I do 57 above and did not select the appropiate
zone on the intercom system. I then changed 57 to call a uniden phone
to see if I can hear the digits. I can not. I then added the wwww to the
string. I definitely hear the WAIT silence on the channel but no DTMF.

I only hear the DTMF tones with calling the linksys phone 404.

The other devices are not getting the DTMF tones.
The grandstream phone, the uniden phone and the HT488 connected to the intercom system.

Any ideas?
By: Serge Vecher (serge-v) 2006-07-12 10:19:02

enable the RTP debug, let's see if DTMF tones are going out.
By: geisj (geisj) 2006-07-12 10:47:59

I have done further "playing" and it seems that adding wwww
in front of my DTMF digits helps with sending ot the intercom.
I have not had to do that before for what thats worth.

I still dont hear the digits on the phone but it seems to be working now
with the intercom.

DO you want me to do the RTP thing or just close the issue.

THanks
By: Serge Vecher (serge-v) 2006-07-12 10:57:32

do the RTP thing
By: geisj (geisj) 2006-07-12 11:53:12

I added the file with
sip debug
rtp debug
set verbose 4
set debug 4

and ran the call from SIP/404 to SIP/402. I still did not hear the
DTMF on SIP/402.
By: Serge Vecher (serge-v) 2006-08-08 12:28:23

hmmm, interesting. We have the message

  -- Sending DTMF 'wwww101' to the called party.

but rtp packets, prior to the native bridge, are going to/from 192.168.1.162, which is a caller.

Have you done any modification to the source code?
By: geisj (geisj) 2006-08-08 13:03:24

No I did no modification to the source.

jerry
By: Joshua C. Colp (jcolp) 2006-09-06 11:12:46

I don't believe there's anything else we can really do with this, the rtp debug does indeed show that DTMF is going to the device. Now it may have an issue with the way we send RFC2833, but that is not something we can solve in 1.2 easily.
By: Joshua C. Colp (jcolp) 2006-09-27 16:33:25

BOOM! I've suspended this since I don't exactly know where we can go with it, your own debug showed that DTMF was being sent out. To that extent though you may want to give the 1.4 beta a try as it handles DTMF differently. Peace!