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...@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 <asghar...@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...@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 >> > > > -- > _____________________________________________________________________ > -- 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