When I changed back to dtmfmode=rfc2833 and I cant hear the DTMF sound.. completely silent.
Indeed I have put disallow=all before the allow=ulaw allow=alaw "sip show channels" in the CLI show during a call: 78.129.xxx.xx +4477xxxxxxxx 15d909406db14d2 0x4 (ulaw) No Tx: ACK 94.192.xxx.xx test MTNlNGNkYjlhODA 0x4 (ulaw) No Rx: ACK Still no luck to get DTMF to work :( Thanks Shahid On Sun, May 6, 2012 at 2:54 PM, Eric Wieling <[email protected]> wrote: > Now you have a totally different issue. 8-) > > While the call is up do a "sip show channels" in the CLI. This will show > you the ACTUAL codec for the call. Likely the call was still using GSM. > Did you remember to put a disallow=all before the allow= lines? > > I recommend dtmfmode=rfc2833 with whatever codec you want to use. Inband > DTMF will sound broken and distorted if it is sent over most codecs. > > > -----Original Message----- > From: [email protected] [mailto: > [email protected]] On Behalf Of Shahid H > Sent: Sunday, May 06, 2012 9:16 AM > To: Markus > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Why SendDTMF is not working? > > Thanks for the suggestion Markus. Here what I did: > > In the logger.config I have added 'dtmf': > > console => notice,warning,error,dtmf > > and then in sip.conf: > > allow=ulaw > allow=alaw > ; allow=gsm > dtmfmode=inband > > I've added a test to call my mobile: > > exten => 123,1,Dial(SIP/+4477XXXXXXX@voipms,,D(wwwwwwww1ww2ww3ww4)) > exten => 123,n,Hangup() > > then restarted asterisk and logged into console (asterisk -r) > > I've call my mobile using softphone, I did not see 1,2,3,4 digits being > sent on the console but I can hear broken/unclear DTMF on the mobile... > > however when I press digits on the softphone I can hear DTMF clear how it > should be on my mobile and on the console it is showing DTMF: > > astrisk*CLI> [May 6 14:13:06] DTMF[28559]: channel.c:3082 __ast_read: > DTMF begin '4' received on SIP/test-0000001c [May 6 14:13:06] DTMF[28559]: > channel.c:3092 __ast_read: DTMF begin passthrough '4' on SIP/test-0000001c > [May 6 14:13:06] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '4' > received on SIP/test-0000001c, duration 120 ms [May 6 14:13:06] > DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '4' on > SIP/test-0000001c [May 6 14:13:06] DTMF[28559]: channel.c:3066 __ast_read: > DTMF end passthrough '4' on SIP/test-0000001c [May 6 14:13:07] > DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '5' received on > SIP/test-0000001c [May 6 14:13:07] DTMF[28559]: channel.c:3092 __ast_read: > DTMF begin passthrough '5' on SIP/test-0000001c [May 6 14:13:07] > DTMF[28559]: channel.c:2997 __ast_read: DTMF end '5' received on > SIP/test-0000001c, duration 120 ms [May 6 14:13:07] DTMF[28559]: > channel.c:3037 __ast_read: DTMF end accepted with begin '5' on > SIP/test-0000001c [May 6 14:13:07] DTMF[28559]: channel.c:3066 __ast_read: > DTMF end passthrough '5' on SIP/test-0000001c [May 6 14:13:08] > DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '6' received on > SIP/test-0000001c [May 6 14:13:08] DTMF[28559]: channel.c:3092 __ast_read: > DTMF begin passthrough '6' on SIP/test-0000001c [May 6 14:13:08] > DTMF[28559]: channel.c:2997 __ast_read: DTMF end '6' received on > SIP/test-0000001c, duration 120 ms [May 6 14:13:08] DTMF[28559]: > channel.c:3037 __ast_read: DTMF end accepted with begin '6' on > SIP/test-0000001c [May 6 14:13:08] DTMF[28559]: channel.c:3066 __ast_read: > DTMF end passthrough '6' on SIP/test-0000001c > > Thanks! > > On Sun, May 6, 2012 at 1:03 PM, Markus <[email protected]> wrote: > > > Am 06.05.2012 13:46, schrieb Shahid H: > > > Hello, > > I am having a problem with SendDTMF - it is not sending the > numbers > properly during the phone call.. I want the numbers key to > to be > pressed/sent automatically after 3 seconds during a phone > call. > > > > Log the actual DTMF to your console, set in logger.conf: > > console => something,something,dtmf > ^^^^ > > Then try again and check if you see the actual DTMF. If you do and > it still doesn't work, try > > dtmfmode=inband > > for your voipms peer. > > rfc2833 has been working always unreliable for me. > > Also, I'm doing DTMF like this: > > exten => 5000,n,Dial(SIP/123456@provider,,D(wwwwww1ww2ww3ww4)) > > Just use more w's to generate your 3 seconds pause. No need for > SendDTMF. > > For more debugging just call yourself on your UK mobile from a > softphone and press digits and watch the console and listen on your mobile > if you hear the DTMF. > > > > > > > -- > _____________________________________________________________________ > -- 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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