[asterisk-users] DTMF recognized after call establishment

[Gopalakrishnan N]

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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>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
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