Hey guys, I have managed to get to work!!!! Thanks for the help..
I just registered a new account at sipgate.co.uk and test it on asterisk... and DTMF worked well :) It seem voip.ms dont work well when sending DTMF to UK. Do anyone know UK/Europe voip provider to allow you change any callerID as you like without validation? I know voip.ms does it and sipgate don't allow it. Thanks! On Sun, May 6, 2012 at 5:08 PM, Shahid H <shah...@gmail.com> wrote: > Here is another debug log: > > == Using SIP RTP CoS mark 5 > -- Executing [123@test2:1] Dial("SIP/test2-00000008", > "SIP/+44776XXXXXXXX@voipms,,D(wwwwwwww1ww2ww3ww4)") in new stack > == Using SIP RTP CoS mark 5 > -- Called SIP/+44776XXXXXXXXXX@voipms > -- SIP/voipms-00000009 is making progress passing it to > SIP/test2-00000008 > -- SIP/voipms-00000009 answered SIP/test2-00000008 > -- Sending DTMF 'wwwwwwww1ww2ww3ww4' to the called party. > -- Locally bridging SIP/test2-00000008 and SIP/voipms-00000009 > > When DTMF is finish then "Locally bridging" is executed... > > On the softphone it say "State: Early Media" while it sending DTMF even > though I cant hear DTMF sound.. after 10 seconds State changed to "Up" (I > can hear talking to myself). > > > > On Sun, May 6, 2012 at 4:18 PM, Shahid H <shah...@gmail.com> wrote: > >> 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 <ewiel...@nyigc.com> 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: asterisk-users-boun...@lists.digium.com [mailto: >>> asterisk-users-boun...@lists.digium.com] 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 <unive...@truemetal.org> 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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