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-00000009When 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 <shahidh@xxxxxxxxx> 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: ACK94.192.xxx.xx test MTNlNGNkYjlhODA 0x4 (ulaw) No Rx: ACKStill no luck to get DTMF to work :(ThanksShahidOn Sun, May 6, 2012 at 2:54 PM, Eric Wieling <EWieling@xxxxxxxxx> 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-bounces@xxxxxxxxxxxxxxxx [mailto:asterisk-users-bounces@xxxxxxxxxxxxxxxx] 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: 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 <universe@xxxxxxxxxxxxx> 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.
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