[asterisk-users] Why SendDTMF is not working?
Shahid H
shahidh at gmail.com
Sun May 6 12:35:05 CDT 2012
Hey guys,
I have managed to get to work!!!! Thanks for the help..
I just registered a new account at sipgate.co.uk and test it on asterisk...
and DTMF worked well :)
It seem voip.ms dont work well when sending DTMF to UK.
Do anyone know UK/Europe voip provider to allow you change any callerID as
you like without validation?
I know voip.ms does it and sipgate don't allow it.
Thanks!
On Sun, May 6, 2012 at 5:08 PM, Shahid H <shahidh at gmail.com> wrote:
> 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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