[asterisk-users] DTMF recognized after call establishment

[Gopalakrishnan N]

Let me try with dtmfmode as auto...
On 28 May 2013 19:32, "Asghar Mohammad" <asghar144 at gmail.com> wrote:

> work around was block dtmf.
> set wrong type of dtmf in incoming trunk.
>
>
> On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N <
> gopalakrishnan.an at gmail.com> wrote:
>
>> So any resolution for this?
>>
>> I suspect it could be related to RE INVITE
>>
>>
>> On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad <asghar144 at gmail.com>wrote:
>>
>>> i had this in past there was an ATA configured to send 9 at the end of
>>> dialing in my case.
>>>
>>>
>>> On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N <
>>> gopalakrishnan.an at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> I am receiving DTMF without any reason after call establishment.
>>>>
>>>> The log as follows, and I suspect something related to directmedia,
>>>> [May 17 00:33:35] VERBOSE[4238] app_dial.c:     -- SIP/MyTrunk-000a4b49
>>>> is making progress passing it to SIP/MAN-000a4b48
>>>> [May 17 00:33:35] VERBOSE[4238] app_dial.c:     -- SIP/MyTrunk-000a4b49
>>>> answered SIP/MAN-000a4b48
>>>> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on
>>>> SIP/MyTrunk-000a4b49, duration 0 ms
>>>> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin
>>>> '*' on SIP/MyTrunk-000a4b49
>>>> [May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on
>>>> SIP/MyTrunk-000a4b49
>>>> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on
>>>> SIP/MyTrunk-000a4b49, duration 0 ms
>>>> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin
>>>> '8' on SIP/MyTrunk-000a4b49
>>>> [May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on
>>>> SIP/MyTrunk-000a4b49
>>>> [May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on
>>>> SIP/MAN-000a4af0, duration 100 ms
>>>> [May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8'
>>>> with duration 100 queued on SIP/MAN-000a4af0
>>>> [May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8'
>>>> queued on SIP/MAN-000a4af0
>>>> [May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on
>>>> SIP/MAN-000a4b41, duration 100 ms
>>>> [May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1'
>>>> with duration 100 queued on SIP/MAN-000a4b41
>>>> [May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1'
>>>> queued on SIP/MAN-000a4b41
>>>> [May 17 00:33:55] VERBOSE[4106] pbx.c:   == Spawn extension
>>>> (sip-trunk-inbound, 2127773456, 1) exited non-zero on
>>>> 'SIP/MyTrunk-000a4af3'
>>>> [May 17 00:33:56] VERBOSE[4136] pbx.c:     -- Executing
>>>> [h at trunk-outbound:1] NoOp("SIP/MAN-000a4b09", "16") in new stack
>>>> [May 17 00:33:56] VERBOSE[4136] pbx.c:   == Spawn extension
>>>> (trunk-outbound, 777787457712, 2) exited non-zero on 'SIP/MAN-000a4b09'
>>>>
>>>> Is this some thing related to SIP RE-INVITE?
>>>>
>>>> Thanks.
>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
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>>>
>>>
>>> --
>>> _____________________________________________________________________
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>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>                http://www.asterisk.org/hello
>>
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>>    http://lists.digium.com/mailman/listinfo/asterisk-users
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>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
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