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...@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@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.
>
>
> --
> _____________________________________________________________________
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_____________________________________________________________________
-- 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
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
