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
