Re: Why SendDTMF is not working?

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Here is another debug log:

 == Using SIP RTP CoS mark 5
    -- Executing [123@test2:1] Dial("SIP/test2-00000008", "SIP/+44776XXXXXXXX@voipms,,D(wwwwwwww1ww2ww3ww4)") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/+44776XXXXXXXXXX@voipms
    -- SIP/voipms-00000009 is making progress passing it to SIP/test2-00000008
    -- SIP/voipms-00000009 answered SIP/test2-00000008
    -- Sending DTMF 'wwwwwwww1ww2ww3ww4' to the called party.
    -- Locally bridging SIP/test2-00000008 and SIP/voipms-00000009

When DTMF is finish then "Locally bridging" is executed...

On the softphone it say "State: Early Media" while it sending DTMF even though I cant hear DTMF sound.. after 10 seconds State changed to "Up" (I can hear talking to myself).



On Sun, May 6, 2012 at 4:18 PM, Shahid H <shahidh@xxxxxxxxx> wrote:
When I changed back to dtmfmode=rfc2833 and I cant hear the DTMF sound.. completely silent.

Indeed I have put disallow=all before the allow=ulaw  allow=alaw 

"sip show channels" in the CLI  show during a call:

78.129.xxx.xx   +4477xxxxxxxx    15d909406db14d2  0x4 (ulaw)       No       Tx: ACK
94.192.xxx.xx   test                      MTNlNGNkYjlhODA  0x4 (ulaw)       No       Rx: ACK

Still no luck to get DTMF to work :(

Thanks
Shahid


On Sun, May 6, 2012 at 2:54 PM, Eric Wieling <EWieling@xxxxxxxxx> wrote:
Now you have a totally different issue.  8-)

While the call is up do a "sip show channels" in the CLI.  This will show you the ACTUAL codec for the call.  Likely the call was still using GSM.  Did you remember to put a disallow=all before the allow= lines?

I recommend dtmfmode=rfc2833 with whatever codec you want to use.   Inband DTMF will sound broken and distorted if it is sent over most codecs.


-----Original Message-----
From: asterisk-users-bounces@xxxxxxxxxxxxxxxx [mailto:asterisk-users-bounces@xxxxxxxxxxxxxxxx] On Behalf Of Shahid H
Sent: Sunday, May 06, 2012 9:16 AM
To: Markus
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Why SendDTMF is not working?

Thanks for the suggestion Markus. Here what I did:

In the logger.config I have added 'dtmf':

console => notice,warning,error,dtmf

and then in sip.conf:

allow=ulaw
allow=alaw
; allow=gsm
dtmfmode=inband

I've added a test to call my mobile:

exten => 123,1,Dial(SIP/+4477XXXXXXX@voipms,,D(wwwwwwww1ww2ww3ww4))
exten => 123,n,Hangup()

then restarted asterisk and logged into console (asterisk -r)

I've call my mobile using softphone, I did not see 1,2,3,4 digits being sent on the console but I can hear broken/unclear DTMF on the mobile...

however when I press digits on the softphone I can hear DTMF clear how it should be on my mobile and on the console it is showing DTMF:

astrisk*CLI> [May  6 14:13:06] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '4' received on SIP/test-0000001c [May  6 14:13:06] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '4' on SIP/test-0000001c [May  6 14:13:06] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '4' received on SIP/test-0000001c, duration 120 ms [May  6 14:13:06] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '4' on SIP/test-0000001c [May  6 14:13:06] DTMF[28559]: channel.c:3066 __ast_read: DTMF end passthrough '4' on SIP/test-0000001c [May  6 14:13:07] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '5' received on SIP/test-0000001c [May  6 14:13:07] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '5' on SIP/test-0000001c [May  6 14:13:07] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '5' received on SIP/test-0000001c, duration 120 ms [May  6 14:13:07] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '5' on SIP/test-0000001c [May  6 14:13:07] DTMF[28559]: channel.c:3066 __ast_read: DTMF end passthrough '5' on SIP/test-0000001c [May  6 14:13:08] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '6' received on SIP/test-0000001c [May  6 14:13:08] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '6' on SIP/test-0000001c [May  6 14:13:08] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '6' received on SIP/test-0000001c, duration 120 ms [May  6 14:13:08] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '6' on SIP/test-0000001c [May  6 14:13:08] DTMF[28559]: channel.c:3066 __ast_read: DTMF end passthrough '6' on SIP/test-0000001c

Thanks!

On Sun, May 6, 2012 at 1:03 PM, Markus <universe@xxxxxxxxxxxxx> wrote:


       Am 06.05.2012 13:46, schrieb Shahid H:


               Hello,

               I am having a problem with SendDTMF - it is not sending the numbers
               properly during the phone call.. I want the numbers key to to be
               pressed/sent automatically after 3 seconds during a phone call.



       Log the actual DTMF to your console, set in logger.conf:

       console => something,something,dtmf
                                     ^^^^

       Then try again and check if you see the actual DTMF. If you do and it still doesn't work, try

       dtmfmode=inband

       for your voipms peer.

       rfc2833 has been working always unreliable for me.

       Also, I'm doing DTMF like this:

       exten => 5000,n,Dial(SIP/123456@provider,,D(wwwwww1ww2ww3ww4))

       Just use more w's to generate your 3 seconds pause. No need for SendDTMF.

       For more debugging just call yourself on your UK mobile from a softphone and press digits and watch the console and listen on your mobile if you hear the DTMF.






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