Vivian Alan
2004-May-25 01:50 UTC
[Asterisk-Users] Question IAX and SIP bound to different IP's on the same * box
-----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, May 25, 2004 5:30 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users digest, Vol 1 #3891 - 8 msgs Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-request@lists.digium.com You can reach the person managing the list at asterisk-users-admin@lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. RE: Newbie extensions.conf I need to include [SMS] context. (Gary Ruddock) 2. Re: Document - contains malware (Trevor Peirce) 3. RE: Newbie extensions.conf I need to include [SMS] context. (Jay Milk) 4. Re: Sip Registration Problem (Olle E. Johansson) 5. Using Ser and Asterisk together (=?iso-8859-1?q?Aiden=20Chew?=) 6. RE: 100 analog phones?? HOWTO? (tan@yointernet.com) 7. SipTone II and Choppy/Stuttering Audio (Nick Grindley) 8. RE: Meetme Options (new one) (Ben Merrills) --__--__-- Message: 1 From: "Gary Ruddock" <garyruddock@hotmail.com> To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Newbie extensions.conf I need to include [SMS] context. Date: Tue, 25 May 2004 07:22:29 +0100 Reply-To: asterisk-users@lists.digium.com I have been up all night and I gotta go to bed. If there's anyone out there using asterisk to send SMS text messages in the UK with BT please gis a clue. Do I need to get the latest asterisk CVS?>Could anyone be so kind as to tell me how to modify this dialplan toaccept>and send SMS text messages. Do I need to update my basic Asterisk to >include SMS functionality? In the example contexts a reference is madeto>/usr/lib/asterisk/smsin and I can't find that file. > > >I know that [local] is executed first and it includes other contexts. I>need to add these two contexts > >[smsdial] ; create and send a text message, expectsnumber+message>and >connect to 17094009 >exten = _X.,1,SMS(${CALLERIDNUM},,${EXTEN},${CALLERIDNAME}) >exten = _X.,2,SMS(${CALLERIDNUM}) >exten = _X.,3,Hangup > >and > >[incoming] >exten = _XXXXXX/_8005875290,1,SMS(${EXTEN:3},a) >exten = _XXXXXX/_8005875290,2,System(/usr/lib/asterisk/smsin${EXTEN:3})>exten = _XXXXXX/_80058752[0-8]0,1,SMS(${EXTEN:3}${CALLERIDNUM:8:1},a) >exten = _XXXXXX/_80058752[0-8]0,2,System(/usr/lib/asterisk/smsin >${EXTEN:3}${CALLERIDNUM:8:1}) >exten = _XXXXXX/_80058752X0,3,Hangup > > >*********************** my extensions.conf *************************** >[general] >static=yes >writeprotect=no > >[globals] >TRUNK=Zap/g1 ; Trunk interface >TRUNKMSD=1 ; MSD digits to strip >(usually 1 or 0) > >[trunkint] >;exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) >;exten => _9011.,2,Congestion > >[trunkld] >exten => _90XXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) >exten => _90XXXNXXXXXX,2,Congestion > >[trunklocal] >exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) >exten => _9NXXXXXX,2,Congestion > >exten => _907NXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) >exten => _907NXXXXXXXX,2,Congestion > >[trunktollfree] >exten => _90800NXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) >exten => _90800NXXXXX,2,Congestion > >[international] >ignorepat => 9 >include => longdistance >include => trunkint > >[longdistance] >;ignorepat => 9 >;include => local >include => trunkld > >[local] >ignorepat => 9 >;include => default >include => parkedcalls >include => trunklocal >include => trunktollfree >include => trunkld > >exten => 6001,1,Dial(SIP/6001,20,tr) >exten => 6002,1,Dial(SIP/6002,20,tr) > >exten => 077777,1,Answer >exten => 077777,2,wait(2) >exten => 077777,3,playback(welcome) >exten => 077777,4,GotoIf($[foo${CALLERIDNUM} = foo]?7:5) >exten => >077777,5,Queue(salesq,,https://swiftdrinks.com/admin/callerid.php?callerid=${CALLERIDNUM})>exten => 077777,6,Hangup >exten => 077777,7,Wait(2) >exten => 077777,8,Playback(privacy-unident) >exten => 077777,9,Hangup > >exten => 2500,1,Dial(Zap/32,40) >exten => 2500,2,VoiceMail2(u2500) >exten => 2500,3,Hangup >exten => 2500,102,VoiceMail2(b2500) >exten => 2500,103,Hangup > >exten => 2501,1,Dial(Zap/33,40) >exten => 2501,2,VoiceMail2(u2500) >exten => 2501,3,Hangup >exten => 2501,102,VoiceMail2(b2501) >exten => 2501,103,Hangup > >exten => 81,1,AddQueueMember(salesq|Zap/32) >exten => 81,2,wait(1) >exten => 81,3,Playback(agent-loginok) >exten => 81,4,wait(1) >exten => 81,5,Hangup > >exten => 82,1,RemoveQueueMember(salesq|Zap/32) >exten => 82,2,wait(1) >exten => 82,3,Playback(agent-loggedoff) >exten => 82,4,wait(1) >exten => 82,5,Hangup > >exten => 95,3,Playback(agent-loginok) >exten => 95,4,wait(1) >exten => 95,5,Hangup > >exten => 96,1,RemoveQueueMember(salesq|SIP/6001) >exten => 96,2,wait(1) >exten => 96,3,Playback(agent-loggedoff) >exten => 96,4,wait(1) >exten => 96,5,Hangup > >exten => 97,1,AddQueueMember(salesq|SIP/6002) >exten => 97,2,wait(1) >exten => 97,3,Playback(agent-loginok) >exten => 97,4,wait(1) >exten => 97,5,Hangup > >exten => 98,1,RemoveQueueMember(salesq|SIP/6002) >exten => 98,2,wait(1) >exten => 98,3,Playback(agent-loggedoff) >exten => 98,4,wait(1) >exten => 98,5,Hangup > >[macro-stdexten] >exten => s,1,Dial(${ARG2},20) ; Ringthe>interface, 20 seconds maximum >exten => s,2,Voicemail(u${ARG1}) ; If >unavailable, send to voicemail w/ unavail announce >exten => s,3,Goto(default,s,1) ; Ifthey>press #, return to start >exten => s,102,Voicemail(b${ARG1}) ; Ifbusy,>send to voicemail w/ busy announce >exten => s,103,Goto(default,s,1) ; Ifthey>press #, return to start > >;[mainmenu] >; >; Example "main menu" context with submenu >; >;exten => s,1,Answer >;exten => s,2,Background(thanks) ; "Thanks for callingpress>1 for sales, 2 for support, ..." >;exten => 1,1,Goto(submenu,s,1) >;exten => 2,1,Hangup >;include => default >; >;[submenu] >;exten => s,1,Ringing ; Make them >comfortable with 2 seconds of ringback >;exten => s,2,Wait,2 >;exten => s,3,Background(submenuopts) ; "Thanks for calling the sales>department. Press 1 for steve, 2 for..." >;exten => 1,1,Goto(default,steve,1) >;exten => 2,1,Goto(default,mark,2) > >[default] >;empty > > >>I want to include a new context in my exensions.conf >> >>I have read this page >>http://www.voip-info.org/wiki-Asterisk+howto+dial+plan and I can sortof>>follow it?! >> >>I have a context [local] that I know zapata.conf points to, I haveedited>>extensions.conf and put in my phone, sip and iax extensions. I want toadd>>an sms context. >> >>I understand that all calls go through my [local] context and I haveother>>contexts that get included into [local] for long distance and freefone>>numbers. >> >>At a guess would I put the code below in extensions.conf and include >>[smsdial] into the [local] context? I have read a page onextensions.conf>>parsing, would I include [smsdial] at the end of [local]? >> >>Please help, cos I have to do the same for [fax]. >> >>[smsdial] ; create and send a text message, expectsnumber+message>>and >>connect to 17094009 >>exten = _X.,1,SMS(${CALLERIDNUM},,${EXTEN},${CALLERIDNAME}) >>exten = _X.,2,SMS(${CALLERIDNUM}) >>exten = _X.,3,Hangup >> >>_________________________________________________________________ >>Use MSN Messenger to send music and pics to your friends >>http://www.msn.co.uk/messenger >> >>_______________________________________________ >>Asterisk-Users mailing list >>Asterisk-Users@lists.digium.com >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > >_________________________________________________________________ >It's fast, it's easy and it's free. Get MSN Messenger today! >http://www.msn.co.uk/messenger > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users_________________________________________________________________ Stay in touch with absent friends - get MSN Messenger http://www.msn.co.uk/messenger --__--__-- Message: 2 Date: Mon, 24 May 2004 23:44:30 -0700 From: Trevor Peirce <tpeirce@digitalcon.ca> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Document - contains malware Reply-To: asterisk-users@lists.digium.com hank wrote:>is this a virus? >Yes. --__--__-- Message: 3 From: "Jay Milk" <jay@skimmilk.net> To: <asterisk-users@lists.digium.com> Subject: RE: [Asterisk-Users] Newbie extensions.conf I need to include [SMS] context. Date: Tue, 25 May 2004 02:08:28 -0500 Reply-To: asterisk-users@lists.digium.com Google on "asterisk sms" -- the first result links to a working example. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Gary Ruddock Sent: Tuesday, May 25, 2004 1:22 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Newbie extensions.conf I need to include [SMS] context. I have been up all night and I gotta go to bed. If there's anyone out there using asterisk to send SMS text messages in the UK with BT please gis a clue. Do I need to get the latest asterisk CVS?>Could anyone be so kind as to tell me how to modify this dialplan to >accept >and send SMS text messages. Do I need to update my basic Asterisk to >include SMS functionality? In the example contexts a reference is madeto>/usr/lib/asterisk/smsin and I can't find that file. > > >I know that [local] is executed first and it includes other contexts. I >need to add these two contexts > >[smsdial] ; create and send a text message, expectsnumber+message>and >connect to 17094009 >exten = _X.,1,SMS(${CALLERIDNUM},,${EXTEN},${CALLERIDNAME}) >exten = _X.,2,SMS(${CALLERIDNUM}) >exten = _X.,3,Hangup > >and > >[incoming] >exten = _XXXXXX/_8005875290,1,SMS(${EXTEN:3},a) >exten = _XXXXXX/_8005875290,2,System(/usr/lib/asterisk/smsin >${EXTEN:3}) exten = >_XXXXXX/_80058752[0-8]0,1,SMS(${EXTEN:3}${CALLERIDNUM:8:1},a) >exten = _XXXXXX/_80058752[0-8]0,2,System(/usr/lib/asterisk/smsin >${EXTEN:3}${CALLERIDNUM:8:1}) >exten = _XXXXXX/_80058752X0,3,Hangup > > >*********************** my extensions.conf ***************************>[general] static=yes >writeprotect=no > >[globals] >TRUNK=Zap/g1 ; Trunk interface >TRUNKMSD=1 ; MSD digits to strip >(usually 1 or 0) > >[trunkint] >;exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) >;exten => _9011.,2,Congestion > >[trunkld] >exten => _90XXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) >exten => _90XXXNXXXXXX,2,Congestion > >[trunklocal] >exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) >exten => _9NXXXXXX,2,Congestion > >exten => _907NXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) >exten => _907NXXXXXXXX,2,Congestion > >[trunktollfree] >exten => _90800NXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) >exten => _90800NXXXXX,2,Congestion > >[international] >ignorepat => 9 >include => longdistance >include => trunkint > >[longdistance] >;ignorepat => 9 >;include => local >include => trunkld > >[local] >ignorepat => 9 >;include => default >include => parkedcalls >include => trunklocal >include => trunktollfree >include => trunkld > >exten => 6001,1,Dial(SIP/6001,20,tr) >exten => 6002,1,Dial(SIP/6002,20,tr) > >exten => 077777,1,Answer >exten => 077777,2,wait(2) >exten => 077777,3,playback(welcome) >exten => 077777,4,GotoIf($[foo${CALLERIDNUM} = foo]?7:5) >exten => >077777,5,Queue(salesq,,https://swiftdrinks.com/admin/callerid.php?callerid=${CALLERIDNUM})>exten => 077777,6,Hangup >exten => 077777,7,Wait(2) >exten => 077777,8,Playback(privacy-unident) >exten => 077777,9,Hangup > >exten => 2500,1,Dial(Zap/32,40) >exten => 2500,2,VoiceMail2(u2500) >exten => 2500,3,Hangup >exten => 2500,102,VoiceMail2(b2500) >exten => 2500,103,Hangup > >exten => 2501,1,Dial(Zap/33,40) >exten => 2501,2,VoiceMail2(u2500) >exten => 2501,3,Hangup >exten => 2501,102,VoiceMail2(b2501) >exten => 2501,103,Hangup > >exten => 81,1,AddQueueMember(salesq|Zap/32) >exten => 81,2,wait(1) >exten => 81,3,Playback(agent-loginok) >exten => 81,4,wait(1) >exten => 81,5,Hangup > >exten => 82,1,RemoveQueueMember(salesq|Zap/32) >exten => 82,2,wait(1) >exten => 82,3,Playback(agent-loggedoff) >exten => 82,4,wait(1) >exten => 82,5,Hangup > >exten => 95,3,Playback(agent-loginok) >exten => 95,4,wait(1) >exten => 95,5,Hangup > >exten => 96,1,RemoveQueueMember(salesq|SIP/6001) >exten => 96,2,wait(1) >exten => 96,3,Playback(agent-loggedoff) >exten => 96,4,wait(1) >exten => 96,5,Hangup > >exten => 97,1,AddQueueMember(salesq|SIP/6002) >exten => 97,2,wait(1) >exten => 97,3,Playback(agent-loginok) >exten => 97,4,wait(1) >exten => 97,5,Hangup > >exten => 98,1,RemoveQueueMember(salesq|SIP/6002) >exten => 98,2,wait(1) >exten => 98,3,Playback(agent-loggedoff) >exten => 98,4,wait(1) >exten => 98,5,Hangup > >[macro-stdexten] >exten => s,1,Dial(${ARG2},20) ; Ringthe>interface, 20 seconds maximum >exten => s,2,Voicemail(u${ARG1}) ; If >unavailable, send to voicemail w/ unavail announce >exten => s,3,Goto(default,s,1) ; Ifthey>press #, return to start >exten => s,102,Voicemail(b${ARG1}) ; Ifbusy,>send to voicemail w/ busy announce >exten => s,103,Goto(default,s,1) ; Ifthey>press #, return to start > >;[mainmenu] >; >; Example "main menu" context with submenu >; >;exten => s,1,Answer >;exten => s,2,Background(thanks) ; "Thanks for callingpress>1 for sales, 2 for support, ..." >;exten => 1,1,Goto(submenu,s,1) >;exten => 2,1,Hangup >;include => default >; >;[submenu] >;exten => s,1,Ringing ; Make them >comfortable with 2 seconds of ringback >;exten => s,2,Wait,2 >;exten => s,3,Background(submenuopts) ; "Thanks for calling the sales>department. Press 1 for steve, 2 for..." >;exten => 1,1,Goto(default,steve,1) >;exten => 2,1,Goto(default,mark,2) > >[default] >;empty > > >>I want to include a new context in my exensions.conf >> >>I have read this page >>http://www.voip-info.org/wiki-Asterisk+howto+dial+plan and I can sortof>>follow it?! >> >>I have a context [local] that I know zapata.conf points to, I have >>edited >>extensions.conf and put in my phone, sip and iax extensions. I want toadd>>an sms context. >> >>I understand that all calls go through my [local] context and I have >>other >>contexts that get included into [local] for long distance and freefone>>numbers. >> >>At a guess would I put the code below in extensions.conf and include >>[smsdial] into the [local] context? I have read a page onextensions.conf>>parsing, would I include [smsdial] at the end of [local]? >> >>Please help, cos I have to do the same for [fax]. >> >>[smsdial] ; create and send a text message, expectsnumber+message>>and >>connect to 17094009 >>exten = _X.,1,SMS(${CALLERIDNUM},,${EXTEN},${CALLERIDNAME}) >>exten = _X.,2,SMS(${CALLERIDNUM}) >>exten = _X.,3,Hangup >> >>_________________________________________________________________ >>Use MSN Messenger to send music and pics to your friends >>http://www.msn.co.uk/messenger >> >>_______________________________________________ >>Asterisk-Users mailing list >>Asterisk-Users@lists.digium.com >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > >_________________________________________________________________ >It's fast, it's easy and it's free. Get MSN Messenger today! >http://www.msn.co.uk/messenger > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users_________________________________________________________________ Stay in touch with absent friends - get MSN Messenger http://www.msn.co.uk/messenger _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- Message: 4 Date: Tue, 25 May 2004 09:12:37 +0200 From: "Olle E. Johansson" <oej@edvina.net> Organization: Edvina AB To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sip Registration Problem Reply-To: asterisk-users@lists.digium.com Karl Brose wrote:> Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or > not, Asterisk doesn't do it correctly either. > The host should respond with 200/OK if the call >could< succeed > theoretically if it were an INVITE or else it should send a > 404 or maybe a 487(? hmm, have to look) see the RFC for details.Interesting, didn't know that. Where in the RFC?>> I removed the qualify lines and sip reload [ed]. The extension still >> showed up as "UNREACHABLE" instead of "UNMONITORED". I had to do a >> full restart to get it to stop sending the OPTIONS messages. >> >> What did I do wrong here? How can I make a change to qualify without >> restarting?If a peer is registred at reload/sip reload, it will not change. You have to unload the sip module and reload it or restart asterisk to change the configuration of a registred, i.e. active, peer. /O --__--__-- Message: 5 Date: Tue, 25 May 2004 15:20:43 +0800 (CST) From: =?iso-8859-1?q?Aiden=20Chew?= <ceyi2r@yahoo.com.sg> To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Using Ser and Asterisk together Reply-To: asterisk-users@lists.digium.com Hi all, I would like to know if it is possible to use asterisk and ser together in a single computer system using ser as a sip proxy and forwarding any voice call request to asterisk for calling into the pstn gateway. (or any other alternative that is possible is also welcomed for suggestions). If it is possible can someone kindly show me the necessary configuration files or refer me to any page that can show me how to do it ? Thanks a lot in advance. Kevin __________________________________________________ Do You Yahoo!? Log on to Messenger with your mobile phone! http://sg.messenger.yahoo.com --__--__-- Message: 6 From: <tan@yointernet.com> To: <asterisk-users@lists.digium.com> Subject: RE: [Asterisk-Users] 100 analog phones?? HOWTO? Date: Tue, 25 May 2004 09:05:13 +0100 Organization: TelAppliant Ltd Reply-To: asterisk-users@lists.digium.com 4 x Mediatrix 1124 VoIP Gateways? http://www.voiptalk.org/products/product_info.php?cPath=31&products_id=7 2 Tan -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Paul Mahler Sent: 25 May 2004 03:33 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] 100 analog phones?? HOWTO? I have had good experiences with Adit. Their customer service and documentation are excellent. Paul Paul Mahler pmahler@signate.com Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training & Consulting> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Jeff Gustafson > Sent: Monday, May 24, 2004 4:21 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] 100 analog phones?? HOWTO? > > Does anyone know the best approach to take for handling > 100 analog phones? It seems to me that a chassis like > Carrier Access or Adtran would work. The chassis would do > much of the hard work of converting the analog sound to data. > Any recommendations on hardware for the chassis? > > ...Jeff > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- Message: 7 From: "Nick Grindley" <npg@itvv.co.uk> To: <asterisk-users@lists.digium.com> Date: Tue, 25 May 2004 09:21:02 +0100 Subject: [Asterisk-Users] SipTone II and Choppy/Stuttering Audio Reply-To: asterisk-users@lists.digium.com Hi All, * is running a dream now, however we have an odd problem that I am sure some guru will be able to sort out for me in no time!! When receiving or making a call about 60 seconds or so into the call we develop choppy/stutter audio problems. It then seems to clear itself only to return again, and so the pattern carries on! This has got me stumped! Our equipment is SipTone II handsets, AVM C2 ISDN Card, Suse Linux 9 and we are in the UK. The SipTone II Firmware version is SipTone 1.2.0 rc Z_11 I have tried all codecs on the handset, i.e. g729, g711 ulaw and g711 alaw (should I have altered something in * as well?) In sip.conf we have: - disallow=all allow=alaw allow=ulaw I think that * is unbelievable value and if I could only sort this out I would be a happy bunny!! Once again many thanks to the whole community for "holding my hand" whilst installing this great software. Kind regards to all Nick From: Nick Grindley Position: Managing Director / CEO Company: Intelligent Television and Video Limited Country: United Kingdom --__--__-- Message: 8 Subject: RE: [Asterisk-Users] Meetme Options (new one) Date: Tue, 25 May 2004 09:26:00 +0100 From: "Ben Merrills" <ben@griffin.com> To: <asterisk-users@lists.digium.com> Reply-To: asterisk-users@lists.digium.com This is a multi-part message in MIME format. ------_=_NextPart_001_01C44231.E955B1E0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Seems like it would be a simple modification? =20 Where would I post a feature request like this? :-) =20 Cheers, Ben =20 ________________________________ From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Chris Sullivan Sent: 24 May 2004 17:16 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Meetme Options (new one) =20 =20 On May 24, 2004, at 8:21 AM, Ben Merrills wrote:=20 =20 Is it possible to select the audio stream that's played as a user enters a meetme conference?=20 =20 I was just now doing an RTFS trying to figure that out.=20 =20 At the moment, the sound played on entering is hard-coded. Time for a feature request?=20 ------_=_NextPart_001_01C44231.E955B1E0 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable <html xmlns:v=3D"urn:schemas-microsoft-com:vml" xmlns:o=3D"urn:schemas-microsoft-com:office:office" xmlns:w=3D"urn:schemas-microsoft-com:office:word" xmlns=3D"http://www.w3.org/TR/REC-html40"> <head> <META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; charset=3Dus-ascii"> <meta name=3DGenerator content=3D"Microsoft Word 11 (filtered medium)"> <!--[if !mso]> <style> v\:* {behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);} w\:* {behavior:url(#default#VML);} .shape {behavior:url(#default#VML);} </style> <![endif]--> <style> <!-- /* Font Definitions */ @font-face {font-family:Wingdings; panose-1:5 0 0 0 0 0 0 0 0 0;} @font-face {font-family:Tahoma; panose-1:2 11 6 4 3 5 4 4 2 4;} /* Style Definitions */ p.MsoNormal, li.MsoNormal, div.MsoNormal {margin:0cm; margin-bottom:.0001pt; font-size:12.0pt; font-family:"Times New Roman";} a:link, span.MsoHyperlink {color:blue; text-decoration:underline;} a:visited, span.MsoHyperlinkFollowed {color:purple; text-decoration:underline;} span.EmailStyle17 {mso-style-type:personal-reply; font-family:Arial; color:navy;} @page Section1 {size:612.0pt 792.0pt; margin:72.0pt 90.0pt 72.0pt 90.0pt;} div.Section1 {page:Section1;} --> </style> </head> <body lang=3DEN-US link=3Dblue vlink=3Dpurple> <div class=3DSection1> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size: 10.0pt;font-family:Arial;color:navy'>Seems like it would be a simple modification?<o:p></o:p></span></font></p> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size: 10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size: 10.0pt;font-family:Arial;color:navy'>Where would I post a feature request like this? </span></font><font size=3D2 color=3Dnavy face=3DWingdings><span style=3D'font-size:10.0pt;font-family:Wingdings;color:navy'>J</span></fo nt><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial; color:navy'><o:p></o:p></span></font></p> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size: 10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size: 10.0pt;font-family:Arial;color:navy'>Cheers,<o:p></o:p></span></font></p><p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size: 10.0pt;font-family:Arial;color:navy'><br> Ben<o:p></o:p></span></font></p> <p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=3D'font-size: 10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p> <div> <div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'> <hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1> </span></font></div> <p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt; font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2 face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] <b><span style=3D'font-weight: bold'>On Behalf Of </span></b>Chris Sullivan<br> <b><span style=3D'font-weight:bold'>Sent:</span></b> 24 May 2004 17:16<br> <b><span style=3D'font-weight:bold'>To:</span></b> asterisk-users@lists.digium.com<br> <b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [Asterisk-Users] Meetme Options (new one)</span></font><o:p></o:p></p> </div> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size: 12.0pt'><o:p> </o:p></span></font></p> <p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'><o:p> </o:p></span></font></p> <div> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size: 12.0pt'>On May 24, 2004, at 8:21 AM, Ben Merrills wrote: <o:p></o:p></span></font></p> </div> <blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size: 12.0pt'><o:p> </o:p></span></font></p> <div> <p class=3DMsoNormal><font size=3D3 face=3DArial><span style=3D'font-size:12.0pt; font-family:Arial'>Is it possible to select the audio stream that’s played as a user enters a meetme conference?</span></font> <o:p></o:p></p> </div> </blockquote> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size: 12.0pt'><o:p> </o:p></span></font></p> <div> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size: 12.0pt'>I was just now doing an RTFS trying to figure that out. <o:p></o:p></span></font></p> </div> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size: 12.0pt'><o:p> </o:p></span></font></p> <div> <p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size: 12.0pt'>At the moment, the sound played on entering is hard-coded. Time for a feature request? <o:p></o:p></span></font></p> </div> </div> </body> </html> ------_=_NextPart_001_01C44231.E955B1E0-- --__--__-- _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest
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