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Tim Thompson

2003-Oct-22 13:31 UTC

[Asterisk-Users] Meetme

Yes.

Tim Thompson
http://www.amatechtel.com
(806) 722-2227


-----Original Message-----
From: CW_ASN - Gus [mailto:cw_asn@fibertel.com.ar] 
Sent: Wednesday, October 22, 2003 1:12 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Meetme

Do you have ztdummy or zaptel device in your system?


----- Original Message ----- 
From: "Panny Malialis" <panny@hotlinks.co.uk>
To: <asterisk-users@lists.digium.com>
Sent: Wednesday, October 22, 2003 1:43 PM
Subject: [Asterisk-Users] Meetme

> Is app_meetme broken?
> 
> I seem to get invalid conference number all the time :(
> 
> Panny
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> 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

Tim Sailer

2004-Mar-04 07:41 UTC

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[Asterisk-Users] Meetme

OK, maybe I need more coffee. Or less. Either way, I'm stumped.

I have a Meetme conference room configured. Meetme(|M|) to enable
the MOH. When you are the first one to go into the conf, you get the
announcement that you are the only one, and then a *male* voice gives
a little talk about 'Why are we putting you on hold?'. Where does that
come from and how do I get rid of it? It's not any of the sound files 
that I can see/hear, and running 'asterisk -vvvr' doesn't show any
file being played...

Tim

-- >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
>> Tim Sailer                       ><  Coastal Internet, Inc.      
<<
>> Network and Systems Operations   ><  PO Box 726                  
<<
>> http://www.buoy.com              ><  Moriches, NY 11955          
<<
>> tps@buoy.com                     ><  (631) 399-2910  (888)
924-3728  <<
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<

Pablo Endres

2004-Jun-28 06:43 UTC

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[Asterisk-Users] Meetme

Hi people,

I have a user that forgets to hangup his conference calls, so they go on
forever.  Is there a way of limiting the duration of a conf call?

Thanks in advance,

Pablo
-- 
Pablo Endres <epablo@comvoz.com>
ComVoz Communications

USA: 	   +1 954 343-2085 Ext 199
Venezuela: +58 212 7713195 Ext 199
Colombia:  +57 1 3256840 Ext 199

Senad Jordanovic

2004-Jun-28 06:54 UTC

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[Asterisk-Users] Meetme

Pablo Endres wrote:> Hi people,
> 
> I have a user that forgets to hangup his conference calls, so they go
> on forever.  Is there a way of limiting the duration of a conf call? 
> 
> Thanks in advance,
> 
> Pablo
Try using ABSOLUTETIMEOUT before starting the conference?

Holger Schurig

2004-Jun-28 07:08 UTC

head link

[Asterisk-Users] Meetme

*CLI> help application AbsoluteTimeout

Travis Conway

2004-Aug-06 11:53 UTC

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[Asterisk-Users] meetme

I am trying to get a simple MeetMe running between a few SIP phones here in our
office.

Here is a clip from my extensions.conf


exten   => 9000,1,Ringing
exten   => 9000,2,Answer
exten   => 9000,3,Wait(1)
exten   => 9000,4,MeetMe(|Md)

Here is what my console says:

    -- Executing Ringing("SIP/loni-b550", "") in new stack
    -- Executing Answer("SIP/loni-b550", "") in new stack
    -- Executing Wait("SIP/loni-b550", "1") in new stack
Aug  6 13:48:56 WARNING[-298230864]: pbx.c:1257 pbx_extension_helper: No
application 'MeetMe' for extension (from-sip, 9000, 4)
  == Spawn extension (from-sip, 9000, 4) exited non-zero on
'SIP/loni-b550'

My meetme.conf file looks correct (nothing really in it, just a [rooms] block).

Perhaps meetme isnt installed?  I used an RPM for RC-1 I found on a mirror.

--
Travis Conway
travisconway@charter.net
FWD: 414668
+1 334 220-7519 (T-Mobile)
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Mitul Sen

2004-Aug-17 12:36 UTC

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[Asterisk-Users] MeetMe

I have just started using * and have been trying to set up MeetMe. So far I
have not been able to start a conference. When I dial the conference
extension (I am using X-Lite softphone), the call hangs up. I am not using
Zaptel cards so I uncommented the ztdummy in the Zaptel Makefile. My
configuration is as follows:

In meetme.conf:
conf => 1234

In extensions.conf:

exten => 2002,1,Answer
exten => 2002,2,Wait(1)
exten => 2002,3,MeetMe,1234
exten => 2002,4,Hangup

At the CLI prompt I get the following:
Aug 16 15:19:43 WARNING[565265]: pbx.c:1257 pbx_extension_helper: No
application 'MeetMe' for extension (sip, 2002, 3)

Is there anything I am missing? Any help will be greatly appreciated.

Regards,
Mitul

Gurdeep Singh Bagga Guru

2004-Aug-17 13:20 UTC

head link

[Asterisk-Users] MeetMe

Hi,

You need ztdummy. Follow this link, it worked for me,
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy

Thanks,
Guru.
> I have just started using * and have been trying to set up MeetMe. So 
far I> have not been able to start a conference. When I dial the conference
> extension (I am using X-Lite softphone), the call hangs up. I am not 
using> Zaptel cards so I uncommented the ztdummy in the Zaptel Makefile. My
> configuration is as follows:
> 
> In meetme.conf:
> conf => 1234
> 
> In extensions.conf:
> 
> exten => 2002,1,Answer
> exten => 2002,2,Wait(1)
> exten => 2002,3,MeetMe,1234
> exten => 2002,4,Hangup
> 
> At the CLI prompt I get the following:
> Aug 16 15:19:43 WARNING[565265]: pbx.c:1257 pbx_extension_helper: No
> application 'MeetMe' for extension (sip, 2002, 3)
> 
> Is there anything I am missing? Any help will be greatly appreciated.
> 
> Regards,
> Mitul
> 
> _______________________________________________
> 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
> 
> 
Thanks & Regards,
Gurdeep (Guru)
+91-11-35372111
guru@baggafamily.net

Gurdeep Singh Bagga Guru

2004-Aug-17 13:21 UTC

head link

[Asterisk-Users] MeetMe

Hi,

You need ztdummy. Follow this link, it worked for me,
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy

Thanks,
Guru.
> I have just started using * and have been trying to set up MeetMe. So 
far I> have not been able to start a conference. When I dial the conference
> extension (I am using X-Lite softphone), the call hangs up. I am not 
using> Zaptel cards so I uncommented the ztdummy in the Zaptel Makefile. My
> configuration is as follows:
> 
> In meetme.conf:
> conf => 1234
> 
> In extensions.conf:
> 
> exten => 2002,1,Answer
> exten => 2002,2,Wait(1)
> exten => 2002,3,MeetMe,1234
> exten => 2002,4,Hangup
> 
> At the CLI prompt I get the following:
> Aug 16 15:19:43 WARNING[565265]: pbx.c:1257 pbx_extension_helper: No
> application 'MeetMe' for extension (sip, 2002, 3)
> 
> Is there anything I am missing? Any help will be greatly appreciated.
> 
> Regards,
> Mitul
> 
> _______________________________________________
> 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
> 
> 
Thanks & Regards,
Gurdeep (Guru)
+91-11-35372111
guru@baggafamily.net

Gurdeep Singh Bagga Guru

2004-Aug-17 13:26 UTC

head link

[Asterisk-Users] MeetMe

Hi,

You need ztdummy. Follow this link, it worked for me,
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy

Thanks,
Guru.
> I have just started using * and have been trying to set up MeetMe. So 
far I> have not been able to start a conference. When I dial the conference
> extension (I am using X-Lite softphone), the call hangs up. I am not 
using> Zaptel cards so I uncommented the ztdummy in the Zaptel Makefile. My
> configuration is as follows:
> 
> In meetme.conf:
> conf => 1234
> 
> In extensions.conf:
> 
> exten => 2002,1,Answer
> exten => 2002,2,Wait(1)
> exten => 2002,3,MeetMe,1234
> exten => 2002,4,Hangup
> 
> At the CLI prompt I get the following:
> Aug 16 15:19:43 WARNING[565265]: pbx.c:1257 pbx_extension_helper: No
> application 'MeetMe' for extension (sip, 2002, 3)
> 
> Is there anything I am missing? Any help will be greatly appreciated.
> 
> Regards,
> Mitul
> 
> _______________________________________________
> 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
> 
> 
Thanks & Regards,
Gurdeep (Guru)
+91-11-35372111
guru@baggafamily.net

Gurdeep Singh Bagga Guru

2004-Aug-17 13:28 UTC

head link

[Asterisk-Users] MeetMe

Hi,

You need ztdummy. Follow this link, it worked for me,
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy

Thanks,
Guru.
> I have just started using * and have been trying to set up MeetMe. So 
far I> have not been able to start a conference. When I dial the conference
> extension (I am using X-Lite softphone), the call hangs up. I am not 
using> Zaptel cards so I uncommented the ztdummy in the Zaptel Makefile. My
> configuration is as follows:
> 
> In meetme.conf:
> conf => 1234
> 
> In extensions.conf:
> 
> exten => 2002,1,Answer
> exten => 2002,2,Wait(1)
> exten => 2002,3,MeetMe,1234
> exten => 2002,4,Hangup
> 
> At the CLI prompt I get the following:
> Aug 16 15:19:43 WARNING[565265]: pbx.c:1257 pbx_extension_helper: No
> application 'MeetMe' for extension (sip, 2002, 3)
> 
> Is there anything I am missing? Any help will be greatly appreciated.
> 
> Regards,
> Mitul
> 
> _______________________________________________
> 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
> 
> 
Thanks & Regards,
Gurdeep (Guru)
+91-11-35372111
guru@baggafamily.net

Darren Wiebe

2004-Sep-21 20:13 UTC

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[Asterisk-Users] Meetme

I'm presently using meetme extensively on my server.  I have a rather 
strange question.  I'm using it with one person in "talk-only"
mode and
everybody else in "monitor" mode.  I'm running an athlon xp 2800
with
1gb of ram.  I can handle 40 users.  Does anybody know of any 
adjustments that could be made to the asterisk to increase this limit?  
Or  other changes.  I can buy a newer system but I'm not sure how much 
that will help.  I'm using only u-law presently.  Is that a good thing 
or bad?

Darren Wiebe
darren@hagenhomes.com

Tim Jackson

2004-Oct-27 14:58 UTC

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[Asterisk-Users] MeetMe

I've got a problem with MeetMe. I dial the extension that dynamically
creates the new conf, but it just hangs up on me after telling me I'm
the only person in the conference. Here's my extensions.conf and what
its doing:


    -- Executing Answer("SIP/101-74c0", "") in new stack
    -- Executing Wait("SIP/101-74c0", "1") in new stack
    -- Executing MeetMe("SIP/101-74c0", "|Dx") in new stack
    -- Playing 'conf-getconfno' (language 'en')
    -- Playing 'conf-getpin' (language 'en')
    -- Created MeetMe conference 1023 for conference '100'
    -- Playing 'conf-onlyperson' (language 'en')
    -- Hungup 'Zap/pseudo-874077465'
  == Spawn extension (default, 800, 3) exited non-zero on 'SIP/101-74c0'

[meetme-int]
exten => 800,1,Answer
exten => 800,2,Wait(1)
exten => 800,3,MeetMe(|Dx)


Tim Jackson
Network Engineer
Angelina County, Texas
(936)639-4827x101 office
(936)414-6723 mobile

Tim Jackson

2004-Oct-27 15:25 UTC

head link

[Asterisk-Users] MeetMe

I figured it out. I don't need the 'x' in there. 

'x' - close the conference and hangup on all others when last marked
user exits

I'm an idiot, sorry.

-Tim

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kristian
Kielhofner
Sent: Wednesday, October 27, 2004 5:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MeetMe

Tim Jackson wrote:
> I've got a problem with MeetMe. I dial the extension that dynamically
> creates the new conf, but it just hangs up on me after telling me I'm
> the only person in the conference. Here's my extensions.conf and what
> its doing:
> 
> 
>     -- Executing Answer("SIP/101-74c0", "") in new
stack
>     -- Executing Wait("SIP/101-74c0", "1") in new stack
>     -- Executing MeetMe("SIP/101-74c0", "|Dx") in new
stack
>     -- Playing 'conf-getconfno' (language 'en')
>     -- Playing 'conf-getpin' (language 'en')
>     -- Created MeetMe conference 1023 for conference '100'
>     -- Playing 'conf-onlyperson' (language 'en')
>     -- Hungup 'Zap/pseudo-874077465'
>   == Spawn extension (default, 800, 3) exited non-zero on
'SIP/101-74c0'> 
> [meetme-int]
> exten => 800,1,Answer
> exten => 800,2,Wait(1)
> exten => 800,3,MeetMe(|Dx)
> 
> 
> Tim Jackson
> Network Engineer
> Angelina County, Texas
> (936)639-4827x101 office
> (936)414-6723 mobile
> 
Tim,

	Even though the conference is dynamic, (I think) you still need
to 
define a room number:

exten => 800,3,MeetMe(800|Dx)

Try that and let us know.

--
Kristian Kielhofner
_______________________________________________
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

Henry Devito

2004-Nov-23 08:08 UTC

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[Asterisk-Users] MeetMe

I know it's on the wiki somewhere, but I've been searching for 2 days on
the
wiki and google.  I just can't seem to find the web interface for the meetme
application.  Does anyone have a link I could use.  I found ASGI's dynamic
conferences which is neat.  Thanks in advance. 
 
 
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Henry Devito

2004-Nov-23 14:37 UTC

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[Asterisk-Users] MeetMe

>Can you please not post html.  I can't correctly reply to this message. 
 
Sorry about the HTML,  I had to reload my windoze machine and I forgot to
turn it off and turn plain text on.
 >>I know it's on the wiki somewhere, but I've been searching for 2
days onthe wiki and google.  I just can't seem to find the web interface for the
meetme application.  Does anyone have a link I could use.  I found ASGI's
dynamic conferences which is neat.  Thanks in advance. 
 >I think the app you are thinking of is meetme2.http://www.areski.net/asterisk-meetme/about.php?s=0
 
Thanks.   For the reply
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Serge Schumacher

2005-Jan-02 04:55 UTC

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[Asterisk-Users] MeetMe

Can it be that the MeetMe application is not installed by default even if
there is a meetme.conf ?

 

pbx.c:1280 pbx_extension_helper: No application 'MeetMe' for extension
(from-sip, 550, 4)

 

 

Regards,

 

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Serge Schumacher

2005-Jan-02 16:34 UTC

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[Asterisk-Users] Meetme

Hi,

Can someone see what's wrong here please ?

I've installed the ztdummy driver to enable meetme, put his in my
extension.conf

exten => 550,1,Answer
exten => 550,2,Wait(1)
exten => 550,4,MeetMe(18|Md)
exten => 550,5,Hangup

this in my meetme.conf

[rooms]
;
; Usage is conf => confno[,pin]
;
conf => 18

and when I call 550 I get this error and the MusicOnHold (exten =>
550,4,MeetMe(18|Md))  also doesn't work:

    -- Executing Answer("SIP/ses-0730", "") in new stack
    -- Executing Wait("SIP/ses-0730", "1") in new stack
    Jan  3 01:40:38 WARNING[12000]: pbx.c:1934 ast_pbx_run: Timeout, but no
rule 't' in context 'from-sip'


Thnx,

Nitesh Divecha

2005-Feb-11 18:32 UTC

head link

[Asterisk-Users] Meetme

Hey All,

 

Just finished installing Asterisk and configured all the necessary
parameters to start.

I can't seem to find the Meetme application in my asterisk directory. 

 

I downloaded asterisk from CVS and installed it and all my Snom phones are
working and voicemail too.

 

I am getting error: -

Feb 11 17:10:19 WARNING[13042]: pbx.c:1280 pbx_extension_helper: No
application 'Meetme' for extension (sip, 5557, 1)

  == Spawn extension (sip, 5557, 1) exited non-zero on 'SIP/phone1-f88d'

 

Do I need zaptel to be installed?

 

Any help will be appreciated.

 

 

Nitesh 

 

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Matt Schwartz

2005-Apr-17 17:47 UTC

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[Asterisk-Users] MeetMe

Hi, I just recently installed Asterisk 1.0.7 but I cannot figure out how to
install the MeetMe application.  I don't think it installed with the
standard 'make install' command.  If not, how do I accomplish this?
 
Thanks,
Matt
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John Bittner

2006-Mar-13 19:45 UTC

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[Asterisk-Users] Meetme

Anyone notice on latest SVN trunk that meetme no longer asks for a pin when
you try to enter a room?
I want to verify it before I post it as a bug.


John Bittner
Simlab.net

Miles Scruggs

2006-Apr-09 05:39 UTC

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[Asterisk-Users] meetme

I'm having issues getting meetme to work:

Apr  9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No 
application 'MeetMe' for extension (internal, 9999, 2)
  == Spawn extension (internal, 9999, 2) exited non-zero on 
'SIP/mileslap-569b'

the only thing I could find was this:

http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extension&hl=en&gl=us&ct=clnk&cd=4&lr=lang_en&client=firefox-a

but I have the timer working (I think):

lsmod | grep dummy
ztdummy                 2608  -

I'm really confused as to what to do next, if someone could help me out 
that would be great:

I'm using gentoo with kernel 2.6.15.  asterisk has been compiled from 
scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5

Thanks

Miles

Miles Scruggs

2006-Apr-09 05:49 UTC

head link

[Asterisk-Users] meetme

I'm having issues getting meetme to work:

Apr  9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No 
application 'MeetMe' for extension (internal, 9999, 2)
 == Spawn extension (internal, 9999, 2) exited non-zero on 
'SIP/mileslap-569b'

the only thing I could find was this:

http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extension&hl=en&gl=us&ct=clnk&cd=4&lr=lang_en&client=firefox-a


but I have the timer working (I think):

lsmod | grep dummy
ztdummy                 2608  -

I'm really confused as to what to do next, if someone could help me out 
that would be great:

I'm using gentoo with kernel 2.6.15.  asterisk has been compiled from 
scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5

Thanks

Miles

Alexander Lopez

2006-Apr-09 08:30 UTC

head link

[Asterisk-Users] meetme

Snip..
> >>> Thanks
> >>>
> >>> Miles
> >>
> >> If you type "modprobe zaptel" "modprobe
ztdummy" at the Linux CLI,
> >> what do you get?
> > Nothing, they were loaded before, and loaded just fine.
> >
> > lsmod      Module                  Size  Used by
> > ztdummy                 2608  -
> > rtc                    10620  -
> > zaptel                186468  -
> > crc_ccitt               1576  -
> > 3c59x                  40240  -
> > _______________________________________________
> >
> And your dialplan for extension 9999?
> Also post a 'show applications' form your asterisk CLI> prompt.
>

Alexander Lopez

2006-Apr-09 12:16 UTC

head link

[Asterisk-Users] meetme

Snip.. 
> > you could try dialing 9999 from another phone and to dial 
> either 1023 
> > or 0, my guess is 1023 is what the other people will have to dial.
> I would assume that it would work like that, but nope.  9999 
> from a different phone just creates a new conf, and 1023 is 
> never announced it is only in in the logs.
> 
> where would a person find out about how meetme() works?
>From the CLI you can do show application meetme.
I would suggest you add a conference number to the extension. 

IE

Exten => 750,1,MeetMe(750| your options go here )
Exten => 751,1,MeetMe(750| your options go here )
Exten => 752,1,MeetMe(750| your options go here )
Exten => 753,1,MeetMe(750| your options go here )


That will give you 4 meetme rooms. Play with that and then add and
remove the different options that are in MeetMe to 'tweak' your install.

Alex

Paul A Brown

2006-Apr-09 12:44 UTC

head link

[Asterisk-Users] Problem - Voicemail resets phone

Hi Everyone,

Things seem to work fine (except my phone audio issue in a previous mail)

I can leave a vmail message and it emails it out fine. However when I dial 
the vmail server from any phone it usually resets the phone half way 
through. There is no single point where it starts to do this, it can vary 
but it happens sometimes after I connect to the vmail server.

Has anyone seen this? What other details can I post?

Thanks

Paul

Giuseppe

2006-Apr-10 02:19 UTC

head link

[Asterisk-Users] meetme

Hi,
when I try to use meetme I always hear this error message
"this is not a valid conference number, please try again",
but my configuration seems to be correct... Here it is:

-- extensions.conf --
exten => 6000,1,MeetMe(1234,ciMp) ; entra nella meetme room 1234

-- meetme.conf --
[rooms]
conf => 1234

Does anyone has the same problem? Any idea?

Thanks in advance!

Giuseppe

Paul A Brown

2006-Apr-10 11:12 UTC

head link

[Asterisk-Users] Problem - Voicemail resets phone

Just reposting as I know a lot of fresh faces are online :-)

Any help appreciated

Thanks

Paul
----- Original Message ----- 
From: "Paul A Brown" <paul@fowlmere.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users@lists.digium.com>
Sent: Sunday, April 09, 2006 8:44 PM
Subject: [Asterisk-Users] Problem - Voicemail resets phone

> Hi Everyone,
>
> Things seem to work fine (except my phone audio issue in a previous mail)
>
> I can leave a vmail message and it emails it out fine. However when I dial 
> the vmail server from any phone it usually resets the phone half way 
> through. There is no single point where it starts to do this, it can vary 
> but it happens sometimes after I connect to the vmail server.
>
> Has anyone seen this? What other details can I post?
>
> Thanks
>
> Paul
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Paul A Brown

2006-Apr-11 13:26 UTC

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[Asterisk-Users] Cisco 7960 problems

Anyone any ideas?

They are SIP phones. I am not sure if its an asterisk or phone problem. Any 
help to isolate would be good.

Thanks

Paul
----- Original Message ----- 
From: "Paul A Brown" <paul@fowlmere.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users@lists.digium.com>
Sent: Sunday, April 09, 2006 8:41 PM
Subject: [Asterisk-Users] Cisco 7960 problems

> Hi All,
>
> Not sure if this is a phone problem or an Asterisk problem.
>
> Basically after a period of time (around 30 minutes but not too sure of 
> the time) the phone no longer delivers any sounds. What I mean by that 
> is.....
>
> if I pick up the phone after a reset I get a dialtone. After around 30 
> minutes and I pick up phone I get no dial tone but I can still dial. I 
> dialled the voicemail number, I can see on the asterisk console its asking 
> for which vmail box and password but I hear nothing. Anyone heard anything 
> like this before?
>
> Thanks
>
> Paul
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

Dovid Bender

2006-Jun-21 10:45 UTC

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[Asterisk-Users] Meetme

Hi List,
  I have the following in my extensions.conf. For some reason if the user enters
a room that does not exist instead of going to the next pri. it just says room
invalid and dumps the call. Can it be a bug ?
   
  Exten => _*5XXX,1,MeetMe(${EXTEN:1},D)
  Exten => _5XXX,1,MeetMe(${EXTEN},cMrpsq)
  Exten => _5XXX,2,Goto(MainIVR,s,1)
  I also tried addding the following which didnt work:
  Exten => _5XXX,102,Goto(MainIVR,s,1)
   
  Thanks a lot.
   
  Dovid

 		
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