subject:"\[asterisk\-users\] Meetme problems"

[asterisk-users] Meetme problems

2007-06-01 Thread ram


Hi

I have reading the voiip side i found some document says

"
The conference bridge runs Ulaw codec by default. If you let people connect
with GSM or other codecs, Asterisk will use CPU power to convert audio
between codecs "


iam using vicidial and meetme for callcenter application. iam geting choppy
voice, and voice breaks.

iam using connecting VoIP SIP provider using g729 codec, since i can save
bandwidth

iam using client side also g729, so no translation required

but after i see this document, will meetme convert the g729 to GSM or ULAW
internall, and
i have will have cpu load, is this correct.

if i dont want to CPU loadup more, i should use GSM or ULAW at client side
is this correct.

can some one correct me if iam wrong

suggestions welcome

ram
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Re: [Asterisk-Users] MeetMe Problems

2005-06-23 Thread Waldo Rubinstein

Doing further tests, I discovered that I can successfully do MeetMe  
on both server B and server C, AS LONG AS all parties are SIP  
extensions registered on the same server (e.g. server B or server C).  
However, when I try to bring a call from server A into a MeetMe in  
server B or server C, that's when the problem shows up. Hope this  
helps anyone who can help me.


Thanks,
Waldo

On Jun 22, 2005, at 3:06 PM, Waldo Rubinstein wrote:

I decided to test a similar scenario against another machine  
(server C). This machine behaves in a similar way as server B. It  
is also running on Gentoo. When I try to transfer a call into a  
conference room, it fails. Below is the CLI output of an inbound  
call coming from server A into server C, ringing extension SIP/ 
3211. Once answered, I try transferring to MeetMe room 0211 which  
fails.


bacardi init.d # aa
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.7, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
== 
===

Connected to Asterisk 1.0.7 currently running on bacardi (pid = 10925)
Verbosity was 0 and is now 10
-- Accepting AUTHENTICATED call from 10.0.10.9, requested  
format = 4, actual format = 4
-- Executing Dial("IAX2/[EMAIL PROTECTED]/16384", "SIP/3211") in  
new stack

-- Called 3211
-- SIP/3211-1bd8 is ringing
-- SIP/3211-1bd8 answered IAX2/[EMAIL PROTECTED]/16384
-- Started music on hold, class 'default', on IAX2/ 
[EMAIL PROTECTED]/16384

-- Executing MeetMe("SIP/3211-e3c6", "0211|qM") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '0211'
-- Started music on hold, class 'default', on SIP/3211-e3c6
-- Stopped music on hold on SIP/3211-e3c6
-- Stopped music on hold on IAX2/[EMAIL PROTECTED]/16384
Jun 22 14:59:40 WARNING[10952]: app_meetme.c:667 conf_run: Error  
getting conference

-- Hungup 'Zap/pseudo-1721629866'
  == Spawn extension (default, 0211, 1) exited non-zero on 'IAX2/ 
[EMAIL PROTECTED]/16384'

-- Hungup 'IAX2/[EMAIL PROTECTED]/16384'
-- Attempting native bridge of SIP/3211-e3c6 and SIP/ 
3211-1bd8
Jun 22 14:59:40 WARNING[10951]: rtp.c:1365 ast_rtp_bridge: Can't  
find native functions for channel 'SIP/3211-e3c6'
Jun 22 14:59:40 WARNING[10951]: channel.c:2634 ast_channel_bridge:  
Private bridge between SIP/3211-e3c6 and SIP/3211-1bd8 failed
  == Spawn extension (default, 3211, 1) exited non-zero on 'SIP/ 
3211-e3c6'


I don't know if it has anything to do with the  channel.  
lsmod shows that both zaptel and ztdummy are loaded. Any ideas?


Thanks,
Waldo

On Jun 22, 2005, at 10:41 AM, Waldo Rubinstein wrote:


Absolutely. Here is the CLI output. I made two attempts. First, I  
dialed inbound into an extension and then tried using meetme room  
0201 from Server B, which didn't work. Then I dialed inbound into  
the same extension and then tried using meetme room 0215 which  
resides in Server A. Note that all inbound calls come into Server  
A, for it has the Digium card.


SERVER A
=

gateway0:~# aa
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk CVS-Nv1-0-7-06/01/05-01:27:25, Copyright (C) 1999-2004  
Digium.

Written by Mark Spencer <[EMAIL PROTECTED]>
= 

Connected to Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 currently  
running on gateway0 (pid = 2653)

Verbosity is at least 10
-- Starting simple switch on 'Zap/1-1'
-- Executing NoOp("Zap/1-1", "Inbound Call - 3211 - ") in new  
stack

-- Executing Dial("Zap/1-1", "IAX2/corona/3211||r") in new stack
-- Called corona/3211
-- Call accepted by 10.0.10.13 (format ulaw)
-- Format for call is ulaw
-- IAX2/corona/16386 is ringing
-- IAX2/corona/16386 answered Zap/1-1
-- Hungup 'IAX2/corona/16386'
  == Spawn extension (inbound, 3211, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
-- Executing NoOp("Zap/1-1", "Inbound Call - 3211 - ") in new  
stack

-- Executing Dial("Zap/1-1", "IAX2/corona/3211||r") in new stack
-- Called corona/3211
-- Call accepted by 10.0.10.13 (format ulaw)
-- Format for call is ulaw
-- IAX2/corona/16388 is ringing
-- IAX2/corona/16388 answered Zap/1-1
  == Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '0215'
-- Started music on hold, class 'default', on IAX2/ 
[EMAIL PROTECTED]/16390

-- Hungup 'Zap/31-1'
-- Hungup 'IAX2/corona/16388'
  == Spawn extension (inbound, 3211, 2) exited non-zero on 'Zap/1-1'
-- Stopped music on hold on IAX2/[EMAIL PROTECTED]/16390
-- Hungup 'Zap/pseudo-1262753463'
  == Spawn extension (meetme, 0215, 1) exited non-zero on 'IAX2/ 
[EMAIL PROTECTE

Re: [Asterisk-Users] MeetMe Problems

2005-06-22 Thread Waldo Rubinstein

I decided to test a similar scenario against another machine (server  
C). This machine behaves in a similar way as server B. It is also  
running on Gentoo. When I try to transfer a call into a conference  
room, it fails. Below is the CLI output of an inbound call coming  
from server A into server C, ringing extension SIP/3211. Once  
answered, I try transferring to MeetMe room 0211 which fails.


bacardi init.d # aa
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.7, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
 
=

Connected to Asterisk 1.0.7 currently running on bacardi (pid = 10925)
Verbosity was 0 and is now 10
-- Accepting AUTHENTICATED call from 10.0.10.9, requested format  
= 4, actual format = 4
-- Executing Dial("IAX2/[EMAIL PROTECTED]/16384", "SIP/3211") in  
new stack

-- Called 3211
-- SIP/3211-1bd8 is ringing
-- SIP/3211-1bd8 answered IAX2/[EMAIL PROTECTED]/16384
-- Started music on hold, class 'default', on IAX2/ 
[EMAIL PROTECTED]/16384

-- Executing MeetMe("SIP/3211-e3c6", "0211|qM") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '0211'
-- Started music on hold, class 'default', on SIP/3211-e3c6
-- Stopped music on hold on SIP/3211-e3c6
-- Stopped music on hold on IAX2/[EMAIL PROTECTED]/16384
Jun 22 14:59:40 WARNING[10952]: app_meetme.c:667 conf_run: Error  
getting conference

-- Hungup 'Zap/pseudo-1721629866'
  == Spawn extension (default, 0211, 1) exited non-zero on 'IAX2/ 
[EMAIL PROTECTED]/16384'

-- Hungup 'IAX2/[EMAIL PROTECTED]/16384'
-- Attempting native bridge of SIP/3211-e3c6 and SIP/ 
3211-1bd8
Jun 22 14:59:40 WARNING[10951]: rtp.c:1365 ast_rtp_bridge: Can't find  
native functions for channel 'SIP/3211-e3c6'
Jun 22 14:59:40 WARNING[10951]: channel.c:2634 ast_channel_bridge:  
Private bridge between SIP/3211-e3c6 and SIP/3211-1bd8 failed
  == Spawn extension (default, 3211, 1) exited non-zero on 'SIP/3211- 
e3c6'


I don't know if it has anything to do with the  channel.  
lsmod shows that both zaptel and ztdummy are loaded. Any ideas?


Thanks,
Waldo

On Jun 22, 2005, at 10:41 AM, Waldo Rubinstein wrote:

Absolutely. Here is the CLI output. I made two attempts. First, I  
dialed inbound into an extension and then tried using meetme room  
0201 from Server B, which didn't work. Then I dialed inbound into  
the same extension and then tried using meetme room 0215 which  
resides in Server A. Note that all inbound calls come into Server  
A, for it has the Digium card.


SERVER A
=

gateway0:~# aa
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk CVS-Nv1-0-7-06/01/05-01:27:25, Copyright (C) 1999-2004  
Digium.

Written by Mark Spencer <[EMAIL PROTECTED]>
== 
===
Connected to Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 currently  
running on gateway0 (pid = 2653)

Verbosity is at least 10
-- Starting simple switch on 'Zap/1-1'
-- Executing NoOp("Zap/1-1", "Inbound Call - 3211 - ") in new  
stack

-- Executing Dial("Zap/1-1", "IAX2/corona/3211||r") in new stack
-- Called corona/3211
-- Call accepted by 10.0.10.13 (format ulaw)
-- Format for call is ulaw
-- IAX2/corona/16386 is ringing
-- IAX2/corona/16386 answered Zap/1-1
-- Hungup 'IAX2/corona/16386'
  == Spawn extension (inbound, 3211, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
-- Executing NoOp("Zap/1-1", "Inbound Call - 3211 - ") in new  
stack

-- Executing Dial("Zap/1-1", "IAX2/corona/3211||r") in new stack
-- Called corona/3211
-- Call accepted by 10.0.10.13 (format ulaw)
-- Format for call is ulaw
-- IAX2/corona/16388 is ringing
-- IAX2/corona/16388 answered Zap/1-1
  == Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '0215'
-- Started music on hold, class 'default', on IAX2/ 
[EMAIL PROTECTED]/16390

-- Hungup 'Zap/31-1'
-- Hungup 'IAX2/corona/16388'
  == Spawn extension (inbound, 3211, 2) exited non-zero on 'Zap/1-1'
-- Stopped music on hold on IAX2/[EMAIL PROTECTED]/16390
-- Hungup 'Zap/pseudo-1262753463'
  == Spawn extension (meetme, 0215, 1) exited non-zero on 'IAX2/ 
[EMAIL PROTECTED]/16390'

-- Hungup 'Zap/1-1'
-- Hungup 'IAX2/[EMAIL PROTECTED]/16390'

Here are the relevant sections in the .conf files:

meetme.conf:
[rooms]
conf => 0215

extensions.conf:
[meetme]
exten => 0215,1,MeetMe(0215|qM)
exten => 0215,2,Hangup


SERVER B
=

corona root # aa
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.7, Copyright (C) 1999-2004 Digium.
Written by

Re: [Asterisk-Users] MeetMe Problems

2005-06-22 Thread Waldo Rubinstein

Absolutely. Here is the CLI output. I made two attempts. First, I  
dialed inbound into an extension and then tried using meetme room  
0201 from Server B, which didn't work. Then I dialed inbound into the  
same extension and then tried using meetme room 0215 which resides in  
Server A. Note that all inbound calls come into Server A, for it has  
the Digium card.


SERVER A
=

gateway0:~# aa
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk CVS-Nv1-0-7-06/01/05-01:27:25, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
 
=
Connected to Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 currently running  
on gateway0 (pid = 2653)

Verbosity is at least 10
-- Starting simple switch on 'Zap/1-1'
-- Executing NoOp("Zap/1-1", "Inbound Call - 3211 - ") in new stack
-- Executing Dial("Zap/1-1", "IAX2/corona/3211||r") in new stack
-- Called corona/3211
-- Call accepted by 10.0.10.13 (format ulaw)
-- Format for call is ulaw
-- IAX2/corona/16386 is ringing
-- IAX2/corona/16386 answered Zap/1-1
-- Hungup 'IAX2/corona/16386'
  == Spawn extension (inbound, 3211, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
-- Executing NoOp("Zap/1-1", "Inbound Call - 3211 - ") in new stack
-- Executing Dial("Zap/1-1", "IAX2/corona/3211||r") in new stack
-- Called corona/3211
-- Call accepted by 10.0.10.13 (format ulaw)
-- Format for call is ulaw
-- IAX2/corona/16388 is ringing
-- IAX2/corona/16388 answered Zap/1-1
  == Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '0215'
-- Started music on hold, class 'default', on IAX2/ 
[EMAIL PROTECTED]/16390

-- Hungup 'Zap/31-1'
-- Hungup 'IAX2/corona/16388'
  == Spawn extension (inbound, 3211, 2) exited non-zero on 'Zap/1-1'
-- Stopped music on hold on IAX2/[EMAIL PROTECTED]/16390
-- Hungup 'Zap/pseudo-1262753463'
  == Spawn extension (meetme, 0215, 1) exited non-zero on 'IAX2/ 
[EMAIL PROTECTED]/16390'

-- Hungup 'Zap/1-1'
-- Hungup 'IAX2/[EMAIL PROTECTED]/16390'

Here are the relevant sections in the .conf files:

meetme.conf:
[rooms]
conf => 0215

extensions.conf:
[meetme]
exten => 0215,1,MeetMe(0215|qM)
exten => 0215,2,Hangup


SERVER B
=

corona root # aa
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.7, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
 
=

Connected to Asterisk 1.0.7 currently running on corona (pid = 5105)
Verbosity is at least 10
-- Remote UNIX connection
-- Call accepted by 10.0.10.9 (format ulaw)
-- Format for call is ulaw
-- Accepting unauthenticated call from 10.0.10.9, requested  
format = 4, actual format = 4

-- Executing Goto("IAX2/[EMAIL PROTECTED]/16395", "211|1") in new stack
-- Goto (client,211,1)
-- Executing Macro("IAX2/[EMAIL PROTECTED]/16395", "stdexten|211|SIP/ 
3211") in new stack
-- Executing Dial("IAX2/[EMAIL PROTECTED]/16395", "SIP/3211|20|t")  
in new stack

-- Called 3211
-- SIP/3211-3c74 is ringing
-- SIP/3211-3c74 answered IAX2/[EMAIL PROTECTED]/16395
-- Started music on hold, class 'default', on  
IAX2/[EMAIL PROTECTED]/16395

-- Executing MeetMe("SIP/3211-4ed5", "0201|qM") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '0201'
-- Started music on hold, class 'default', on SIP/3211-4ed5
-- Stopped music on hold on SIP/3211-4ed5
-- Stopped music on hold on IAX2/[EMAIL PROTECTED]/16395
Jun 21 18:57:41 WARNING[8254]: app_meetme.c:667 conf_run: Error  
getting conference

-- Hungup 'Zap/pseudo-510190782'
  == Spawn extension (client_INT, 0201, 1) exited non-zero on  
'IAX2/[EMAIL PROTECTED]/16395'

-- Executing Hangup("IAX2/[EMAIL PROTECTED]/16395", "") in new stack
  == Spawn extension (client_INT, h, 1) exited non-zero on  
'IAX2/[EMAIL PROTECTED]/16395'
  == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/ 
3211-4ed5' in macro 'stdexten'
  == Spawn extension (client, 211, 1) exited non-zero on 'SIP/ 
3211-4ed5'

-- Executing Hangup("SIP/3211-4ed5", "") in new stack
  == Spawn extension (client, h, 1) exited non-zero on 'SIP/ 
3211-4ed5'

-- Hungup 'IAX2/[EMAIL PROTECTED]/16395'
-- Hungup 'IAX2/gateway0/16384'
-- Accepting unauthenticated call from 10.0.10.9, requested  
format = 4, actual format = 4

-- Executing Goto("IAX2/[EMAIL PROTECTED]/16386", "211|1") in new stack
-- Goto (client,211,1)
-- Executing Macro("IAX2/[EMAIL PROTECTED]/16386", "stdexten|211|SIP/ 
3211") in new stack
-- Executing Dial("IAX2/[EMAIL PROTECTED]/16386", "SIP/3211|20|t"

Re: [Asterisk-Users] MeetMe Problems

2005-06-21 Thread Moises Silva

it would be very helpfull (IMHO) if you post the output of the
Asterisk console with a high verbosity level. Also, show us how the
important code in your extensions.conf

best regards

On 6/21/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> I have two asterisk machines. One of them has a Digium board (server
> A) and the other is simply using ztdummy (server B). Server A is
> running on Debian and Server B is running Gentoo. Server A is running
> Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 and Server B is running
> Asterisk 1.0.7.
> 
> The problem I have is that when I try to transfer a call into a
> meetme room in server B, it simply hangs up the call. To be specific,
> when I press transfer (XFER on the Uniden UIP200) and then the meetme
> room number, the meetme room answers (I hear MOH), but when I hang
> up, it drops all calls and not just transfers the call to the meetme
> room.
> 
> Now, if I configure the meetme rooms indentically in server A, I can
> transfer the calls from server B to server A's meetme room and
> everything works just fine.
> 
> I would like for the meetme rooms to work in server B and not having
> to depend on server A for it.
> 
> Can anyone shed some light into why this is happening and, more
> importantly, how to fix it?
> 
> Thanks,
> Waldo
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[Asterisk-Users] MeetMe Problems

2005-06-21 Thread Waldo Rubinstein

I have two asterisk machines. One of them has a Digium board (server  
A) and the other is simply using ztdummy (server B). Server A is  
running on Debian and Server B is running Gentoo. Server A is running  
Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 and Server B is running  
Asterisk 1.0.7.


The problem I have is that when I try to transfer a call into a  
meetme room in server B, it simply hangs up the call. To be specific,  
when I press transfer (XFER on the Uniden UIP200) and then the meetme  
room number, the meetme room answers (I hear MOH), but when I hang  
up, it drops all calls and not just transfers the call to the meetme  
room.


Now, if I configure the meetme rooms indentically in server A, I can  
transfer the calls from server B to server A's meetme room and  
everything works just fine.


I would like for the meetme rooms to work in server B and not having  
to depend on server A for it.


Can anyone shed some light into why this is happening and, more  
importantly, how to fix it?


Thanks,
Waldo
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[asterisk-users] Meetme problems

Re: [Asterisk-Users] MeetMe Problems

Re: [Asterisk-Users] MeetMe Problems

Re: [Asterisk-Users] MeetMe Problems

Re: [Asterisk-Users] MeetMe Problems

[Asterisk-Users] MeetMe Problems

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