Hey guys,<div><br></div><div>I have managed to get to work!!!! Thanks for the help..</div><div><br></div><div>I just registered a new account at <a href="http://sipgate.co.uk">sipgate.co.uk</a> and test it on asterisk... and DTMF worked well :)</div>
<div><br></div><div>It seem <a href="http://voip.ms">voip.ms</a> dont work well when sending DTMF to UK. </div><div><br></div><div>Do anyone know UK/Europe voip provider to allow you change any callerID as you like without validation?</div>
<div><br></div><div>I know <a href="http://voip.ms">voip.ms</a> does it and sipgate don't allow it.</div><div><br></div><div>Thanks!</div><div><br><div class="gmail_quote">On Sun, May 6, 2012 at 5:08 PM, Shahid H <span dir="ltr"><<a href="mailto:shahidh@gmail.com" target="_blank">shahidh@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Here is another debug log:<div><br></div><div><div class="im"><div> == Using SIP RTP CoS mark 5</div></div><div> -- Executing [123@test2:1] Dial("SIP/test2-00000008", "SIP/+44776XXXXXXXX@voipms,,D(wwwwwwww1ww2ww3ww4)") in new stack</div>
<div class="im">
<div> == Using SIP RTP CoS mark 5</div></div><div> -- Called SIP/+44776XXXXXXXXXX@voipms</div><div> -- SIP/voipms-00000009 is making progress passing it to SIP/test2-00000008</div><div> -- SIP/voipms-00000009 answered SIP/test2-00000008</div>
<div> -- Sending DTMF 'wwwwwwww1ww2ww3ww4' to the called party.</div><div> -- Locally bridging SIP/test2-00000008 and SIP/voipms-00000009</div><div><br></div><div>When DTMF is finish then "Locally bridging" is executed...</div>
<div><br></div><div>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).</div>
<div><div class="h5"><div>
<br></div><div><br></div><br><div class="gmail_quote">On Sun, May 6, 2012 at 4:18 PM, Shahid H <span dir="ltr"><<a href="mailto:shahidh@gmail.com" target="_blank">shahidh@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
When I changed back to dtmfmode=rfc2833 and I cant hear the DTMF sound.. completely silent.<div><br></div><div>Indeed I have put disallow=all before the allow=ulaw allow=alaw </div><div><br></div><div>"sip show channels" in the CLI show during a call:</div>
<div><br></div><div><div>78.129.xxx.xx +4477xxxxxxxx 15d909406db14d2 0x4 (ulaw) No Tx: ACK</div><div>94.192.xxx.xx test MTNlNGNkYjlhODA 0x4 (ulaw) No Rx: ACK</div></div>
<div><br></div><div>Still no luck to get DTMF to work :(</div><div><br></div><div>Thanks</div><div><span><font color="#888888">Shahid</font></span><div><div><br><br><div class="gmail_quote">On Sun, May 6, 2012 at 2:54 PM, Eric Wieling <span dir="ltr"><<a href="mailto:EWieling@nyigc.com" target="_blank">EWieling@nyigc.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Now you have a totally different issue. 8-)<br>
<br>
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?<br>
<br>
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.<br>
<div><div><br>
<br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Shahid H<br>
Sent: Sunday, May 06, 2012 9:16 AM<br>
To: Markus<br>
Cc: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Subject: Re: [asterisk-users] Why SendDTMF is not working?<br>
<br>
Thanks for the suggestion Markus. Here what I did:<br>
<br>
In the logger.config I have added 'dtmf':<br>
<br>
console => notice,warning,error,dtmf<br>
<br>
and then in sip.conf:<br>
<br>
allow=ulaw<br>
allow=alaw<br>
; allow=gsm<br>
dtmfmode=inband<br>
<br>
I've added a test to call my mobile:<br>
<br>
exten => 123,1,Dial(SIP/+4477XXXXXXX@voipms,,D(wwwwwwww1ww2ww3ww4))<br>
exten => 123,n,Hangup()<br>
<br>
then restarted asterisk and logged into console (asterisk -r)<br>
<br>
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...<br>
<br>
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:<br>
<br>
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<br>
<br>
Thanks!<br>
<br>
On Sun, May 6, 2012 at 1:03 PM, Markus <<a href="mailto:universe@truemetal.org" target="_blank">universe@truemetal.org</a>> wrote:<br>
<br>
<br>
Am 06.05.2012 13:46, schrieb Shahid H:<br>
<br>
<br>
Hello,<br>
<br>
I am having a problem with SendDTMF - it is not sending the numbers<br>
properly during the phone call.. I want the numbers key to to be<br>
pressed/sent automatically after 3 seconds during a phone call.<br>
<br>
<br>
<br>
Log the actual DTMF to your console, set in logger.conf:<br>
<br>
console => something,something,dtmf<br>
^^^^<br>
<br>
Then try again and check if you see the actual DTMF. If you do and it still doesn't work, try<br>
<br>
dtmfmode=inband<br>
<br>
for your voipms peer.<br>
<br>
rfc2833 has been working always unreliable for me.<br>
<br>
Also, I'm doing DTMF like this:<br>
<br>
exten => 5000,n,Dial(SIP/123456@provider,,D(wwwwww1ww2ww3ww4))<br>
<br>
Just use more w's to generate your 3 seconds pause. No need for SendDTMF.<br>
<br>
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.<br>
<br>
<br>
<br>
<br>
<br>
<br>
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</blockquote></div><br></div></div></div>
</blockquote></div><br></div>