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 >
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