[asterisk-users] Why SendDTMF is not working?

Sun May 6 11:08:49 CDT 2012

Here is another debug log:

 == Using SIP RTP CoS mark 5
    -- Executing [123 at test2:1] Dial("SIP/test2-00000008",
"SIP/+44776XXXXXXXX at voipms,,D(wwwwwwww1ww2ww3ww4)") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/+44776XXXXXXXXXX at 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 at gmail.com> 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 at nyigc.com> 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 at lists.digium.com [mailto:
>> asterisk-users-bounces at lists.digium.com] 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: [asterisk-users] 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 at 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 at truemetal.org> 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 at 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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>
>
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