Following : <br><br>The Line is SIP, <br>Asterisk is 1.6.2.5-0ubuntu1.4 <br><br>Also I tried konference volume [konfernecename] up but it did not increase<br>
<br><b>See the extensions.conf where i applied increase volume . <br></b><br>---------<br><b>SetGlobalVar(Set(VOLUME(TX)=10))<br>SetGlobalVar(Set(VOLUME(RX)=10)) </b><br>[bigbluebutton]<br>exten => _.,1,Goto(start-dialplan,s,1)<br>
exten => _.,n,Hangup<br>[start-dialplan]<br>exten => s,1,Set(TRIES=1)<br>exten => s,n,Wait(2)<br>exten => s,n,Answer<br>exten => s,n,Goto(prompt,s,1)<br>[prompt]<br>exten => s,1,Read(CONF_NUM,conf-getconfno,6,,3,10)<br>
exten => s,n,Goto(bbb-conference,${CONF_NUM},1)<br>; No need to check if conference is valid as they won't be able to login<br>; if the conference is invalid.<br>;<br>[bbb-voip]<br>exten => _XXXX.,1,Playback(conf-placeintoconf)<br>
; exten => _XXXX.,n,MeetMe(${EXTEN},cdMsT)<br>exten => _XXXX.,n,Konference(${EXTEN})<br>[bbb-conference]<br>include => echo-test<br>exten => _XXXX.,1,Agi(agi://localhost/findConference?conference=${EXTEN})<br>
exten => _XXXX.,n,GotoIf($[${EXTEN} = ${CONFERENCE_FOUND}]?valid:invalid)<br>exten => _XXXX.,n(valid),Playback(conf-placeintoconf)<br>; exten => _XXXX.,n,MeetMe(${CONFERENCE_FOUND},cdMsT)<br>exten => _XXXX.,n,Konference(${CONFERENCE_FOUND})<br>
exten => _XXXX.,n(invalid),Goto(handle-invalid-conference,s,1)<br>[handle-invalid-conference]<br>exten => s,1,Playback(conf-invalid)<br>exten => s,n,GotoIf($[${TRIES} < 3]?try-again:do-not-try-again)<br>exten => s,n(try-again),Set(TRIES=$[${TRIES} + 1])<br>
exten => s,n,Goto(prompt,s,1)<br>exten => s,n(do-not-try-again),Hangup<br>[echo-test]<br>;<br>; Create an extension, 600, for evaluating echo latency.<br>;<br>exten => 600,1,Answer ; Do the echo test<br>
exten => 600,n,Playback(demo-echotest) ; Let them know what's going on<br>exten => 600,n,Echo ; Do the echo test<br>exten => 600,n,Playback(demo-echodone) ; Let them know it's over<br>
exten => 600,n,Goto(s,6) ; Start over<br><br>---------<br><br><br><br>Zeeshan<br><br><br><div class="gmail_quote">On Wed, Aug 3, 2011 at 4:16 PM, Danny Nicholas <span dir="ltr"><<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"><div link="blue" vlink="purple" lang="EN-US"><div><p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">You need to provide more information – is line in SIP or DAHDI, what release of Asterisk, etc.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"><u></u> <u></u></span></p><p class="MsoNormal"><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;"> <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>] <b>On Behalf Of </b>Zeeshan Ali Shah<br>
<b>Sent:</b> Wednesday, August 03, 2011 9:13 AM<br><b>To:</b> <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br><b>Subject:</b> [asterisk-users] Increasing volume ?<u></u><u></u></span></p>
<div><div></div><div class="h5"><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">Hi, I am running asterisk with konference . tried to increase the conference voice but not success <br><br>i tried to add in diaplain <br>
SetGlobalVar(Set(VOLUME(TX)=10))<br>SetGlobalVar(Set(VOLUME(RX)=10))<br><br>but it does not effect..<br><br><br>any hint ?<br><br>Zeeshan<u></u><u></u></p></div></div></div></div><br>--<br>
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