[asterisk-users] Why SendDTMF is not working?

Sun May 6 10:18:08 CDT 2012

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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