Gopalakrishnan N
2013-May-28 06:21 UTC
[asterisk-users] DTMF recognized after call establishment
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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130528/9c6d2670/attachment.htm>
Asghar Mohammad
2013-May-28 08:39 UTC
[asterisk-users] DTMF recognized after call establishment
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.an at 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 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. > > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130528/9510d073/attachment.htm>