Let me try with dtmfmode as auto... On 28 May 2013 19:32, "Asghar Mohammad" <asghar...@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...@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 >
-- _____________________________________________________________________ -- 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