[asterisk-users] DTMF recognized after call establishment

[Gopalakrishnan N]

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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