[asterisk-users] Problems with meetme in 1.4.26

2010-08-12 Thread Danny Nicholas
Hi list,

  I was going through my dialplan today and found these 2 oddities
with meetme using DAHDI to join the conference.

1.  Although music on hold is indicated, I don't get any sound until I press
*.  Then the conference menu plays and all is well except -

2.  According to the instructions, pressing 8 should leave the conference.
No dice.

 

Problem persists with 1.4.30

 

Any suggestions?

Thanks in Advance 

 

Danny Nicholas

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Re: [asterisk-users] Problems with meetme in 1.4.26

2010-08-12 Thread covici
Danny Nicholas da...@debsinc.com wrote:

 Hi list,
 
   I was going through my dialplan today and found these 2 oddities
 with meetme using DAHDI to join the conference.
 
 1.  Although music on hold is indicated, I don't get any sound until I press
 *.  Then the conference menu plays and all is well except -
 
 2.  According to the instructions, pressing 8 should leave the conference.
 No dice.
8 exits the menu, not the conference.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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Re: [asterisk-users] Problems with meetme in 1.4.26

2010-08-12 Thread Danny Nicholas
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
cov...@ccs.covici.com
Subject: Re: [asterisk-users] Problems with meetme in 1.4.26

Danny Nicholas da...@debsinc.com wrote:

 Hi list,
 
   I was going through my dialplan today and found these 2
oddities
 with meetme using DAHDI to join the conference.
 
 1.  Although music on hold is indicated, I don't get any sound until I
press
 *.  Then the conference menu plays and all is well except -
 
 2.  According to the instructions, pressing 8 should leave the
conference.
 No dice.
8 exits the menu, not the conference.

Arrgh.  Found the problem.  Thought I was running meetme with just small I,
turns out I was using msqx; changed this to spqx and now life is good.


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[Asterisk-Users] Problems with meetme dropping audio during call

2005-11-03 Thread James Moore
I'm using meetme with three SIP calls, updated to the latest Asterisk CVS
(as of around 10am Nov 3).  After a minute or two we start getting
substantial cutouts of the audio during the call.  I'm using the ztdummy
timer, also the latest CVS release.

Suggestions for things to look at?

There are some warnings on the console - I've included that output below.

Fedora Core 4, 2.6.13-1.1532_FC4 kernel.

- James

---

-- Created MeetMe conference 1023 for conference '1234'
-- Playing 'conf-onlyperson' (language 'en')
-- Executing MeetMe(SIP/xx.xx.xx.xx-097cc028, 1234) in new stack
-- Executing MeetMe(SIP/xx.xx.xx.xx-097e6b30, 1234) in new stack
-- Executing MeetMe(SIP/xx.xx.xx.xx-097e0ce0, 1234) in new stack
set verbose 9
Verbosity is at least 9
*CLI Nov  3 10:31:58 WARNING[17653]: app_meetme.c:1467 conf_run: Unable to
writ
e frame to channel: Success
  == Spawn extension (default, 12069737581, 1) exited non-zero on
'SIP/xx.xx.xxx
.141-097cc028'
Nov  3 10:31:58 WARNING[17673]: app_meetme.c:1467 conf_run: Unable to write
fram
e to channel: Success
Nov  3 10:31:58 WARNING[17647]: app_meetme.c:1467 conf_run: Unable to write
fram
e to channel: No such file or directory
  == Spawn extension (default, 12069737581, 1) exited non-zero on
'SIP/xx.xx.xxx
.141-097e0ce0'
  == Spawn extension (default, 12069737581, 1) exited non-zero on
'SIP/xx.xx.xxx
.141-097b7a00'
Nov  3 10:31:59 WARNING[17658]: app_meetme.c:1467 conf_run: Unable to write
fram
e to channel: Success
-- Hungup 'Zap/pseudo-477583753'
  == Spawn extension (default, 12069737581, 1) exited non-zero on
'SIP/xx.xx.xxx
.141-097e6b30'

*CLI

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[Asterisk-Users] Problems with meetme.

2005-04-10 Thread Xisco (Personal)



Hi everybody,

I'm new in *, i have installed over fedora core 3,
with kernel version 2.6 and ztdummy.

I have created one conference in meetme.conf and I
have modified properly extension.conf. But when I try to do a call to this
extension I get the following errors:

 -- Executing
Answer("SIP/xmg-cba9", "") in new stack -- Executing
Wait("SIP/xmg-cba9", "3") in new stackApr 9 15:53:19 NOTICE[12916]:
rtp.c:453 ast_rtp_read: RTP: Received packet with bad UDP
checksum -- Executing MeetMe("SIP/xmg-cba9", "1234") in
new stack == Parsing '/etc/asterisk/meetme.conf': FoundApr 9
15:53:20 WARNING[12916]: chan_zap.c:841 zt_open: Unable to open
'/dev/zap/pseudo': No such file or directoryApr 9 15:53:20
ERROR[12916]: chan_zap.c:6959 chandup: Unable to dup channel: No such file or
directoryApr 9 15:53:20 WARNING[12916]: app_meetme.c:304 build_conf:
Unable to open pseudo channel - trying deviceApr 9 15:53:20
WARNING[12916]: app_meetme.c:307 build_conf: Unable to open pseudo
device -- Playing 'conf-invalid' (language 'en')
== Spawn extension (sip-incoming, 1000, 3) exited non-zero on
'SIP/xmg-cba9'

Can anybody help me? O tell me where I have to look
for...

tnxs in advance.
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Re: [Asterisk-Users] Problems with meetme.

2005-04-10 Thread Adam Rybak
You need to compile zaptel with ztdummy module.
Uncomment ztdummy in Makefile.
You need have to loaded this module as kernel module before executing meetme.
Here you will find more detailed info:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy

Adam

Cytowanie Xisco (Personal) [EMAIL PROTECTED]:

 Hi everybody,

 I'm new in *, i have installed over fedora core 3, with kernel version 2.6
 and ztdummy.

 I have created one conference in meetme.conf and I have modified properly
 extension.conf. But when I try to do a call to this extension I get the
 following errors:

 -- Executing Answer(SIP/xmg-cba9, ) in new stack
 -- Executing Wait(SIP/xmg-cba9, 3) in new stack
 Apr  9 15:53:19 NOTICE[12916]: rtp.c:453 ast_rtp_read: RTP: Received packet
 with bad UDP checksum
 -- Executing MeetMe(SIP/xmg-cba9, 1234) in new stack
   == Parsing '/etc/asterisk/meetme.conf': Found
 Apr  9 15:53:20 WARNING[12916]: chan_zap.c:841 zt_open: Unable to open
 '/dev/zap/pseudo': No such file or directory
 Apr  9 15:53:20 ERROR[12916]: chan_zap.c:6959 chandup: Unable to dup channel:
 No such file or directory
 Apr  9 15:53:20 WARNING[12916]: app_meetme.c:304 build_conf: Unable to open
 pseudo channel - trying device
 Apr  9 15:53:20 WARNING[12916]: app_meetme.c:307 build_conf: Unable to open
 pseudo device
 -- Playing 'conf-invalid' (language 'en')
   == Spawn extension (sip-incoming, 1000, 3) exited non-zero on
 'SIP/xmg-cba9'

 Can anybody help me? O tell me where I have to look for...

 tnxs in advance.


Pozdrawiam,
Adam Rybak
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Re: [Asterisk-Users] Problems with MeetMe accepting conference PIN

2005-01-07 Thread Dean Thompson
Hi!,
[...]
I know this question may have been asked before (although the archives
don't seem to suggest it), but has anyone had any problems with Asterisk
accepting a PIN number for a conference room.
At this point in time I have established the conference definition in
the meetme.conf file as well as specifying the appropriate lines in the
extensions.conf file.
meetme.conf file:
conf = 1234,1574
extensions.conf file:
exten = 1234,1,MeetMe(5678) OR exten = 1234,1,MeetMe,5678
In your extensions.conf file you need to call the meeting that you created
in the meetme.conf file.
So you in your example you need to call MeetMe(1234), which is the meeting
you created in the meetme.conf file.
Thanks for pointing that out. I have made the appropriate 
configurations, but the system stil refuses to accept the pin for the 
conference.

The definition now looks like the following:
meetme.conf file:
conf = 1234 OR
conf = 1234,5678
 +-- When the PIN is specified in the meetme.conf file, the system does
  not recognise the conference at all.
extensions.conf file:
exten = 1000,1,Answer
exten = 1000,2,Wait(1)
exten = 1000,3,Authenticate(5678)
exten = 1000,4,MeetMe(1324)
exten = 1000,5,Playback(vm-goodbye)
exten = 1000,6,Hangup
The PIN number (5678) is still not recognised.  I should point out that 
I am using the SJPhone client and the INBAND DTMF configuration, for the 
phone to work.

I still can't really work out why the pin 5678 (with the #) key at the 
end is still not being recognised by the system.  Both the ztdummy and 
the zaprtc modules for Linux 2.6 have been used to provide RTC support. 
 Native RTC has been removed from the kernel.

See ya
Dean Thompson
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Re: [Asterisk-Users] Problems with MeetMe accepting conference PIN

2005-01-07 Thread Dinesh Nair
On 08/01/2005 11:35 Dean Thompson said the following:
Thanks for pointing that out. I have made the appropriate 
configurations, but the system stil refuses to accept the pin for the 
conference.

The definition now looks like the following:
meetme.conf file:
conf = 1234 OR
conf = 1234,5678
 +-- When the PIN is specified in the meetme.conf file, the system does
  not recognise the conference at all.
extensions.conf file:
exten = 1000,1,Answer
exten = 1000,2,Wait(1)
exten = 1000,3,Authenticate(5678)
exten = 1000,4,MeetMe(1324)
is MeetMe(1324) a typo or is it really MeetMe(1234) ? :)
--
Regards,   /\_/\   All dogs go to heaven.
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
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| for a in past present future; do|
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+=+
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Re: [Asterisk-Users] Problems with MeetMe accepting conference PIN

2005-01-07 Thread Dean Thompson
Hi,
Dinesh Nair wrote:
On 08/01/2005 11:35 Dean Thompson said the following:
Thanks for pointing that out. I have made the appropriate 
configurations, but the system stil refuses to accept the pin for the 
conference.

The definition now looks like the following:
meetme.conf file:
conf = 1234 OR
conf = 1234,5678
 +-- When the PIN is specified in the meetme.conf file, the system does
  not recognise the conference at all.
extensions.conf file:
exten = 1000,1,Answer
exten = 1000,2,Wait(1)
exten = 1000,3,Authenticate(5678)
exten = 1000,4,MeetMe(1324)
is MeetMe(1324) a typo or is it really MeetMe(1234) ? :)
Okay, yes that was a typo it should have been: MeetMe(1234).
Either way, that 5678 isn't being accepted as the PIN number for the 
conference.

Just not sure whether it is SJPhone playing up and not sending the right 
signal for the #, or whether there is a more underlying problem here.

Regards,
Dean Thompson
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Re: [Asterisk-Users] Problems with MeetMe accepting conference PIN

2005-01-07 Thread Dinesh Nair
On 08/01/2005 15:01 Dean Thompson said the following:
Just not sure whether it is SJPhone playing up and not sending the right 
signal for the #, or whether there is a more underlying problem here.
in which case you could try using a dialplan with Read() and SayDigits() to 
see of the SJPhone is actually sending the correct DTMF across and asterisk 
is recognizing it.

--
Regards,   /\_/\   All dogs go to heaven.
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo The opinions here in no way reflect the opinions of my $a $b.  |
| done; done  |
+=+
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[Asterisk-Users] Problems with MeetMe accepting conference PIN

2005-01-06 Thread Dean Thompson
Hi,
I know this question may have been asked before (although the archives 
don't seem to suggest it), but has anyone had any problems with Asterisk 
accepting a PIN number for a conference room.

At this point in time I have established the conference definition in 
the meetme.conf file as well as specifying the appropriate lines in the 
extensions.conf file.

meetme.conf file:
conf = 1234,1574
extensions.conf file:
exten = 1234,1,MeetMe(5678) OR exten = 1234,1,MeetMe,5678
In both cases I can get the SIP client SJPhone to actually dial the 
conference number and I hear the operator say, enter the pin number 
followed by a pound sign.

After entering in the PIN number and entering in the HASH button, the 
system waits for a couple of seconds and then tells me that I have 
entered an invalid PIN number for the conference, please try again.

I have read articles on the web which say that you need a proper RTC 
clocking source for MeetMe to work and I have tried both the ztdummy 
module and the zaprtc-for-2.6 module, but neither seem to be able to 
solve the issue.

At this point in time this is Asterisk CVS version (non-stable) as of 
4/1/05 running on a Fedora Core 3 installation without any ZapTel 
hardware cards.  I have recompiled the kernel and am using 2.6.9 and I 
have both the ztdummy module, zaptel, zaprtc and other modules loaded. 
Asterisk will process normal SIP calls without any problem and allow me 
to leave VoiceMail at the other end, but it refuses to accept the PIN 
number to a conference.

Has anyone had issues similar to this and be able to point me in the 
right direction.  I have also tried using the Asterisk Live CD as well 
and had the same problem regarding the use of conference rooms.

I was beginning to think that SJPhone may well be the problem and that 
it isn't sending the correct tone for the pound sign, but I was 
interested to know whether anyone else had this issue when they were 
setting up their Asterisk PBX.

Looking forward to your replies,
Dean Thompson
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RE: [Asterisk-Users] Problems with MeetMe accepting conference PIN

2005-01-06 Thread Geoff Nordli
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Dean Thompson
 Sent: Thursday, January 06, 2005 6:01 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Problems with MeetMe accepting conference PIN
 
 
 Hi,
 
 I know this question may have been asked before (although the archives
 don't seem to suggest it), but has anyone had any problems with Asterisk
 accepting a PIN number for a conference room.
 
 At this point in time I have established the conference definition in
 the meetme.conf file as well as specifying the appropriate lines in the
 extensions.conf file.
 
 meetme.conf file:
 conf = 1234,1574
 
 extensions.conf file:
 exten = 1234,1,MeetMe(5678) OR exten = 1234,1,MeetMe,5678
 

Hi Dean.

In your extensions.conf file you need to call the meeting that you created
in the meetme.conf file.

So you in your example you need to call MeetMe(1234), which is the meeting
you created in the meetme.conf file.

Geoff


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[Asterisk-Users] Problems with MeetMe SIP / Mobile

2003-10-15 Thread XISCOAIR
Hi everybody,

I'm trying to use MeetMe(2000|p) in order to enter in a conference 
room. But when a mobile or a SIP call press '#' to go out from it, 
everybody goes out. Instead when a analg press it, all works fine.

Anyone else have this problem?? Can anybody help me please??

Thanks a lot.

Regards.


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[Asterisk-Users] Problems with MeetMe.

2003-10-13 Thread XISCOAIR
Good afternoon,
 
I'm trying to use MeetMe in an AGI script written in Perl, as follows:
 
print EXEC MeetMe 2000|p \n;
$res = checkresult();
 
The problem that I have is that when a user press '#' in order to exit 
from the conference, everybody goes out. This is randomized because 
sometimes doesn't happened.
 
My current version of asterisk is:
 
Asterisk CVS-05/22/03-11:14:50 built by [EMAIL PROTECTED] on a i686 running 
Linux
 
Does somebody knows the problem? It's a version problem of *???
 
I would be very pleasured if somebody can help me or told me if I'm 
mistaked about this functionality of MeetMe.
 
Thanks a lot.


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[Asterisk-Users] Problems with meetme.

2003-10-13 Thread XISCOAIR
Good afternoon,

I'm trying to use MeetMe in an AGI script written in Perl, as follows:

print EXEC MeetMe 2000|p \n;
$res = checkresult();

The problem that I have is that when a user press '#' in order to exit 
from the conference, everybody goes out. This is randomized because 
sometimes doesn't happened.

My current version of asterisk is:

Asterisk CVS-05/22/03-11:14:50 built by [EMAIL PROTECTED] on a i686 running 
Linux

Does somebody knows the problem? It's a version problem of *???

I would be very pleasured if somebody can help me or told me if I'm 
mistaked about this functionality of MeetMe.

Thanks a lot.
 



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[Asterisk-Users] Problems with meetme application.

2003-03-17 Thread Xisco Mateu



Hi,

Now I'm trying to use the meetme application like 
as follows:

exten =_917878822,1,Answer.
exten =_917878824,2,MeetMe,1|p

The problem that I have is when a user press '#' in 
order to exit from de conference room, all users go out and the conference room 
finish for all it.

There any system in order to exit onlythe 
user that press a key???

Thks a lot.