[Asterisk-Users] Question IAX and SIP bound to different IP's on the same * box
Vivian Alan
vivian at inttel.net
Tue May 25 01:50:58 MST 2004
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of
asterisk-users-request at lists.digium.com
Sent: Tuesday, May 25, 2004 5:30 AM
To: asterisk-users at lists.digium.com
Subject: Asterisk-Users digest, Vol 1 #3891 - 8 msgs
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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 at 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 at hotmail.com>
To: asterisk-users at 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 at 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 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 made
to
>/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, expects
number+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?calle
rid=${CALLERIDNUM})
>exten => 077777,6,Hangup
>exten => 077777,7,Wait(2)
>exten => 077777,8,Playback(privacy-unident)
>exten => 077777,9,Hangup
>
>exten => 2500,1,Dial(Zap/32,40)
>exten => 2500,2,VoiceMail2(u2500)
>exten => 2500,3,Hangup
>exten => 2500,102,VoiceMail2(b2500)
>exten => 2500,103,Hangup
>
>exten => 2501,1,Dial(Zap/33,40)
>exten => 2501,2,VoiceMail2(u2500)
>exten => 2501,3,Hangup
>exten => 2501,102,VoiceMail2(b2501)
>exten => 2501,103,Hangup
>
>exten => 81,1,AddQueueMember(salesq|Zap/32)
>exten => 81,2,wait(1)
>exten => 81,3,Playback(agent-loginok)
>exten => 81,4,wait(1)
>exten => 81,5,Hangup
>
>exten => 82,1,RemoveQueueMember(salesq|Zap/32)
>exten => 82,2,wait(1)
>exten => 82,3,Playback(agent-loggedoff)
>exten => 82,4,wait(1)
>exten => 82,5,Hangup
>
>exten => 95,3,Playback(agent-loginok)
>exten => 95,4,wait(1)
>exten => 95,5,Hangup
>
>exten => 96,1,RemoveQueueMember(salesq|SIP/6001)
>exten => 96,2,wait(1)
>exten => 96,3,Playback(agent-loggedoff)
>exten => 96,4,wait(1)
>exten => 96,5,Hangup
>
>exten => 97,1,AddQueueMember(salesq|SIP/6002)
>exten => 97,2,wait(1)
>exten => 97,3,Playback(agent-loginok)
>exten => 97,4,wait(1)
>exten => 97,5,Hangup
>
>exten => 98,1,RemoveQueueMember(salesq|SIP/6002)
>exten => 98,2,wait(1)
>exten => 98,3,Playback(agent-loggedoff)
>exten => 98,4,wait(1)
>exten => 98,5,Hangup
>
>[macro-stdexten]
>exten => s,1,Dial(${ARG2},20) ; Ring
the
>interface, 20 seconds maximum
>exten => s,2,Voicemail(u${ARG1}) ; If
>unavailable, send to voicemail w/ unavail announce
>exten => s,3,Goto(default,s,1) ; If
they
>press #, return to start
>exten => s,102,Voicemail(b${ARG1}) ; If
busy,
>send to voicemail w/ busy announce
>exten => s,103,Goto(default,s,1) ; If
they
>press #, return to start
>
>;[mainmenu]
>;
>; Example "main menu" context with submenu
>;
>;exten => s,1,Answer
>;exten => s,2,Background(thanks) ; "Thanks for calling
press
>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 sort
of
>>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 to
add
>>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 on
extensions.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, expects
number+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 at 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 at 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 at digitalcon.ca>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Document - contains malware
Reply-To: asterisk-users at lists.digium.com
hank wrote:
>is this a virus?
>
Yes.
--__--__--
Message: 3
From: "Jay Milk" <jay at skimmilk.net>
To: <asterisk-users at 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 at lists.digium.com
Google on "asterisk sms" -- the first result links to a working example.
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Gary Ruddock
Sent: Tuesday, May 25, 2004 1:22 AM
To: asterisk-users at 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 made
to
>/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, expects
number+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${CALLERIDNUM} = foo]?7:5)
>exten =>
>077777,5,Queue(salesq,,https://swiftdrinks.com/admin/callerid.php?calle
rid=${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(${ARG2},20) ; Ring
the
>interface, 20 seconds maximum
>exten => s,2,Voicemail(u${ARG1}) ; If
>unavailable, send to voicemail w/ unavail announce
>exten => s,3,Goto(default,s,1) ; If
they
>press #, return to start
>exten => s,102,Voicemail(b${ARG1}) ; If
busy,
>send to voicemail w/ busy announce
>exten => s,103,Goto(default,s,1) ; If
they
>press #, return to start
>
>;[mainmenu]
>;
>; Example "main menu" context with submenu
>;
>;exten => s,1,Answer
>;exten => s,2,Background(thanks) ; "Thanks for calling
press
>1 for sales, 2 for support, ..."
>;exten => 1,1,Goto(submenu,s,1)
>;exten => 2,1,Hangup
>;include => default
>;
>;[submenu]
>;exten => s,1,Ringing ; Make them
>comfortable with 2 seconds of ringback
>;exten => s,2,Wait,2
>;exten => s,3,Background(submenuopts) ; "Thanks for calling the sales
>department. Press 1 for steve, 2 for..."
>;exten => 1,1,Goto(default,steve,1)
>;exten => 2,1,Goto(default,mark,2)
>
>[default]
>;empty
>
>
>>I want to include a new context in my exensions.conf
>>
>>I have read this page
>>http://www.voip-info.org/wiki-Asterisk+howto+dial+plan and I can sort
of
>>follow it?!
>>
>>I have a context [local] that I know zapata.conf points to, I have
>>edited
>>extensions.conf and put in my phone, sip and iax extensions. I want to
add
>>an sms context.
>>
>>I understand that all calls go through my [local] context and I have
>>other
>>contexts that get included into [local] for long distance and freefone
>>numbers.
>>
>>At a guess would I put the code below in extensions.conf and include
>>[smsdial] into the [local] context? I have read a page on
extensions.conf
>>parsing, would I include [smsdial] at the end of [local]?
>>
>>Please help, cos I have to do the same for [fax].
>>
>>[smsdial] ; create and send a text message, expects
number+message
>>and
>>connect to 17094009
>>exten = _X.,1,SMS(${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 at 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 at 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 at 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 at edvina.net>
Organization: Edvina AB
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Sip Registration Problem
Reply-To: asterisk-users at 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 at yahoo.com.sg>
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Using Ser and Asterisk together
Reply-To: asterisk-users at 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 at yointernet.com>
To: <asterisk-users at 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 at 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 at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Paul Mahler
Sent: 25 May 2004 03:33
To: asterisk-users at 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 at signate.com
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training & Consulting
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of
> Jeff Gustafson
> Sent: Monday, May 24, 2004 4:21 PM
> To: asterisk-users at 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 at lists.digium.com
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Message: 7
From: "Nick Grindley" <npg at itvv.co.uk>
To: <asterisk-users at 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 at 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 at griffin.com>
To: <asterisk-users at lists.digium.com>
Reply-To: asterisk-users at lists.digium.com
This is a multi-part message in MIME format.
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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 at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Chris
Sullivan
Sent: 24 May 2004 17:16
To: asterisk-users at 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
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<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
n=
t><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 =
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10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
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size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>
asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at 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 at 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>
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