[asterisk-users] Problem with queue

2007-05-14 Thread gc 
Asterisk 1.2.17

I am starting  to have problem with one of my queue. Everytime when I try to 
login an agent with AgentCallBackLogin(), it will play periodic announcement 
for the queue during this function call. Also when this agent answer the call, 
during the conversation, the agent also hear the periodic announcement. I tried 
to delete the agent completely from the queue or recreate the queue, the 
problem still persist. I have not yet restart the asterisk because this is our 
production server.

Gary

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Re: [asterisk-users] Confference function

2007-05-11 Thread gc 

  - Original Message - 
  From: Ed Nuñez 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Monday, April 30, 2007 1:36 PM
  Subject: [asterisk-users] Confference function


  I would like to know if anyone here knows the answer to the following question

   

  I need to implement the following conferencing feature for my agents.

   

  1.   Agent receives call from caller

  2.   Agent conferences a verification service

  3.   After finishing the verification, agent needs to drop third party 
(Verification service) and continue on the line with caller.

   

  My problem right now is being able to disconnect the third party and keeping 
the caller on the line.  Would this be a function of Asterisk or the SIP / IAX 
phone?  Any comments would be appreciated.

   

  Thank you

   

  Ed Nuñez 

  The following page may help you with this:

  http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
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Re: [asterisk-users] Runaway MOH/mp3123 process?

2007-05-04 Thread gc 


- Original Message - 
From: Alex Balashov [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Wednesday, May 02, 2007 2:35 AM
Subject: [asterisk-users] Runaway MOH/mp3123 process?




Has anyone noticed a problem with runaway mpg123 processes for 
music-on-hold eating up ~100% CPU and driving the load on the

machine way up?

I've seen this problem consistently with multiple Asterisk
installs, 1.2.x and 1.4.x, although admittedly it was more
common with 1.2.x as far as I can tell.

There is no clearly identifiable sequence of events that causes
this to occur, although it obviously involves utilisation of the
MOH audio blend at some point, which I use both in queues and for
hold.  But the precise chain of events is never consistent,
predictable, nor triggered in any particular temporal relation
to when MOH is last used--at least, not one that I can pin down.
It does not appear to arise immediately following the activation
of a MOH sequence.


We had the same problem on our Asterisk ACD. After switching to native mode 
of MOH, problem  goes away.


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Re: [asterisk-users] Call prority (QUEUE_PRO) in the queues

2007-04-27 Thread gc 
Whenever I turn the weight option on, it locked the *.  It happens several 
times a day ( abount every two to three hours). When this happen, the 
incoming call can still connect to * but will not hear any music on hold. If 
I issue the 'show channels' command, it shows the connected channels 
continues going up and never released and eventfully * will run out of file 
descriptors and completely lock the *. When this happen, I have to use 
'kill -9 ' to kill * .  When I turn the weight option off, everything works 
fine. I searched the web and several people have the same problem. I also 
found a patch to fix this problem. Right now I am running * using this 
patch. It is up and running for about 24 hours and everything looks good 
right now. Some people have concern about this patch so it has never been 
put into * release. Since we do really need weight option, we have no choice 
but try this patch.


Gary Chen

- Original Message - 
From: BJ Weschke [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Thursday, April 26, 2007 5:36 PM
Subject: Re: [asterisk-users] Call prority (QUEUE_PRO) in the queues



On 4/26/07, gc [EMAIL PROTECTED] wrote:



Suppose I have one agent login into two different queue and there are 
calls

waiting in both queues. If the calls in one queue has higher call prority
(set QUEUE_PRO to higher value) than the calls in other queue, will the
agent get the higher prority call first or the QUEUE_PRO has no effect?
We have an Asterisk server( 1.2.17 with CentOS) running as ACD. We are
having problem using weight option in the queue. I figure maybe I can use
QUEUE_PRO instead.



Queue priority will, unfortunately, only cover one queue. It cannot
cover and account for priorities of calls from more than one queue.
You will want weight for that. What's the problem you were having
with it?

BJ


--
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[asterisk-users] Problem of configuring musiconhold.conf file

2007-04-27 Thread gc 
Asterisk 1.2.17

When try to play moh, I can only use old format in musiconhold.conf file to 
play moh like this:

[moh_files]
default = /var/lib/asterisk/mohmp3,r

If I use the new format like this:

[default]
mode=quietmp3
directory=/var/lib/asterisk/mohmp3

I hear no music at all.

Can anybody tell me what is wrong?

Gary Chen




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Re: [asterisk-users] Problem of configuring musiconhold.conf file

2007-04-27 Thread gc 


- Original Message - 
From: Tzafrir Cohen [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Friday, April 27, 2007 12:55 PM
Subject: Re: [asterisk-users] Problem of configuring musiconhold.conf file



On Fri, Apr 27, 2007 at 12:31:16PM -0400, gc wrote:

Asterisk 1.2.17

When try to play moh, I can only use old format in musiconhold.conf file 
to play moh like this:


[moh_files]
default = /var/lib/asterisk/mohmp3,r

If I use the new format like this:

[default]
mode=quietmp3
directory=/var/lib/asterisk/mohmp3

I hear no music at all.

Can anybody tell me what is wrong?


ls -l /var/lib/asterisk/mohmp3

Do you see any relevant messages in the CLI when a channel is on hold?


Here is the message from logger:
Apr 27 14:09:41 VERBOSE[11172] logger.c: -- Executing 
Wait(SIP/lycin.net-b79332f8, 2) in new stack
Apr 27 14:09:41 DEBUG[11088] chan_sip.c: Stopping retransmission on 
'[EMAIL PROTECTED]' of Response 1: Match Found
Apr 27 14:09:41 DEBUG[11172] channel.c: Generator got voice, switching to 
phase locked mode
Apr 27 14:09:41 DEBUG[11172] channel.c: Scheduling timer at 0 sample 
intervals
Apr 27 14:09:43 VERBOSE[11172] logger.c: -- Executing 
Queue(SIP/lycin.net-b79332f8, queue1|t) in new stack
Apr 27 14:09:43 VERBOSE[11172] logger.c: -- Started music on hold, class 
'default', on channel 'SIP/lycin.net-b79332f8'
Apr 27 14:09:43 DEBUG[11172] channel.c: Scheduling timer at 160 sample 
intervals

Apr 27 14:09:43 DEBUG[11172] app_queue.c: Everyone is busy at this time
Apr 27 14:09:43 DEBUG[11172] channel.c: Generator got voice, switching to 
phase locked mode
Apr 27 14:09:43 DEBUG[11172] channel.c: Scheduling timer at 0 sample 
intervals

Apr 27 14:09:43 DEBUG[11172] channel.c: Auto-deactivating generator
Apr 27 14:09:43 VERBOSE[11172] logger.c: -- Stopped music on hold on 
SIP/lycin.net-b79332f8
Apr 27 14:09:43 DEBUG[11172] channel.c: Scheduling timer at 0 sample 
intervals

Apr 27 14:09:48 DEBUG[11172] app_queue.c: Everyone is busy at this time
Apr 27 14:09:53 DEBUG[11172] app_queue.c: Everyone is busy at this time

What is the 'Auto-deactivation generator' ? It seems that this one cause moh 
to stop immediatly.


Gary Chen 
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[asterisk-users] Test

2007-04-26 Thread gc 
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[asterisk-users] Call prority (QUEUE_PRO) in the queues

2007-04-26 Thread gc 
Suppose I have one agent login into two different queue and there are calls 
waiting in both queues. If the calls in one queue has higher call prority (set 
QUEUE_PRO to higher value) than the calls in other queue, will the agent get 
the higher prority call first or the QUEUE_PRO has no effect?
We have an Asterisk server( 1.2.17 with CentOS) running as ACD. We are having 
problem using weight option in the queue. I figure maybe I can use QUEUE_PRO 
instead. 

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[asterisk-users] test

2007-04-25 Thread gc 
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Re: [asterisk-users] AgentCallBackLogin vs AddQueueMember

2007-02-14 Thread gc 
So you have to hard code each queue name in the dialplan for an agent to login. 
What about hundreds of agents login 30-40 different queues?  If this is the 
only way to do it,  I will not use AddQueueMember at all. I  do not know the 
reason for deprecating AgentCallBackLogin. But I do think remove it without 
appropriate replacement is bad idea.

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Re: [asterisk-users] AgentCallBackLogin vs AddQueueMember

2007-02-14 Thread gc 

  - Original Message - 
  From: James FitzGibbon 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Wednesday, February 14, 2007 10:34 AM
  Subject: Re: [asterisk-users] AgentCallBackLogin vs AddQueueMember


  On 2/13/07, gc [EMAIL PROTECTED] wrote:


I am developing an ACD front end using Asterisk 1.2.14. I heard that 
AgentCallBackLogin will be deprecated in future version of *.
Is this true? If it is, how can I use AddQueueMember to replace 
AgentCallBackLogin? I mean to login an agent in multiple queues at once. I have 
multiple queues and a lot of agents defined in  queues.conf and agents.conf. 
Each agent may login more than one queue. It seem that AgentCallBackLogin  is 
much easier than AddQueueMember to manage this kind of situation. 

  The setup to use AddQueueMember isn't terribly difficult.

  Here's a contrived example where dialing *11[23]XX adds channel SIP[23]XX to 
queue sales, and *21 with the same suffix removes them.  *12/*22 is for 
custserv and *13/*23 is for techsupp.  There's no authentication here, but 
that's not the difficult part of the exercise: 

  exten = _*11[23]XX,1,AddQueueMember(sales,SIP/${EXTEN:3})
  exten = _*11[23]XX,n,Saydigits(${EXTEN:3})
  exten = _*11[23]XX,n,Hangup()
  exten = _*21[23]XX,1,RemoveQueueMember(sales,SIP/${EXTEN:3}) 
  exten = _*21[23]XX,n,Saydigits(${EXTEN:3})
  exten = _*21[23]XX,n,Hangup()

  exten = _*12[23]XX,1,AddQueueMember(custserv,SIP/${EXTEN:3})
  exten = _*12[23]XX,n,Saydigits(${EXTEN:3})
  exten = _*12[23]XX,n,Hangup() 
  exten = _*22[23]XX,1,RemoveQueueMember(custserv,SIP/${EXTEN:3})
  exten = _*22[23]XX,n,Saydigits(${EXTEN:3})
  exten = _*22[23]XX,n,Hangup()

  exten = _*13[23]XX,1,AddQueueMember(techsupp,SIP/${EXTEN:3}) 
  exten = _*13[23]XX,n,Saydigits(${EXTEN:3})
  exten = _*13[23]XX,n,Hangup()
  exten = _*23[23]XX,1,RemoveQueueMember(techsupp,SIP/${EXTEN:3})
  exten = _*23[23]XX,n,Saydigits(${EXTEN:3})
  exten = _*23[23]XX,n,Hangup() 

  Then, calls to Queue(queuename) will work like AgentCallbackLogin() do.

  The problem I am having is that the channel that shows up in the CDR and the 
queue log is the phone that took the call, not the agent on the phone.  It 
seems that I will have to establish a mapping between agents and channels and 
remove down the mapping at agent logoff, then use the map to determine which 
actual agent was on SIP/200 when the call came in in order to produce 
meaningful per-agent reports. 

  Any suggestions on how to make that part easier are welcome.


  -- 
  j. 


--


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Re: [asterisk-users] AgentCallBackLogin vs AddQueueMember

2007-02-14 Thread gc 
So you have to hard code the each queue name in the dialplan for an agent to 
login. What about hundreds of agents login 30-40 different queues?  If this is 
the only way to do it,  I will not use AddQueueMember at all. I  do not know 
the reason for deprecating AgentCallBackLogin. But I do think remove it without 
appropriate replacement is bad idea.

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[asterisk-users] AgentCallBackLogin vs AddQueueMember

2007-02-13 Thread gc 
I am developing an ACD front end using Asterisk 1.2.14. I heard that 
AgentCallBackLogin will be deprecated in future version of *.
Is this true? If it is, how can I use AddQueueMember to replace 
AgentCallBackLogin? I mean to login an agent in multiple queues at once. I have 
multiple queues and a lot of agents defined in  queues.conf and agents.conf. 
Each agent may login more than one queue. It seem that AgentCallBackLogin  is 
much easier than AddQueueMember to manage this kind of situation. 

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Re: [asterisk-users] SER/OpenSER + Asterisk + Queue

2006-12-07 Thread gc 


- Original Message - 
From: [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Tuesday, December 05, 2006 10:56 AM
Subject: [asterisk-users] SER/OpenSER + Asterisk + Queue


We are in the process of redesigning our single Asterisk server that
handles several queues for our clients. We offer our clients hosted
queueing/call center basic services. All the agents are in remote
locations behind NATs using either softphones or PAP2-like devices.

What we would like to accomplish is setup a SER or OpenSER (SER)
server(s) in front of our Asterisk box such that all incoming and outgoing
calls are handled by SER.

The basic idea is to get set up for scaleability and redundancy. The goal
is to be able to add additional Asterisk servers to spread our queue
loads. Nothing fancy, maybe just separate clients on different boxes (not
load balancing queues across multiple Asterisk boxes since that a totally
different scope of project).

We could then add additional SER boxes to protect our inbound and outbound
SIP gateways to our SIP providers (all our calls are SIP-based - e.g. no
TDM circuits).

Lastly, all our agents would register against the SER server(s) instead of
directly to the Asterisk boxes.

Has anyone done this? Can anyone point me to some tips/documentation? Does
anyone care to comment? If agents login using AgentCallBackLogin, will
Asterisk know where the agents are and send the calls to them via SER?

Thank you so much in advanced.

- Daniel

Yes, you can do this. We have our own SIP proxy server. We only use Asterisk
as ACD.
It works good.
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[asterisk-users] Need help on AgentCallbackLogin()

2006-12-07 Thread gc 
When I use AgentCallbackLogin() to logout an agent, it always ask for new 
extension. I can press # to logout. But I'd like the remove this new extension 
prompt so when agents are trying to logout, they do not have to press #.
Does anybody know how to do this?

I am using Asterisk 1.2.12.1

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[asterisk-users] Problem with agent AgentCallbackLogin()

2006-12-01 Thread gc 
When I use AgentCallbackLogin() function to login an agent, I got following 
warning message saying the agent is not valid for auto login while my other 
extensions work fine for this function.
Does anybody know why?  This extension has the same settings as the other ones 
like agents.conf and queues.conf.

Here is the message from asterisk:
-- Executing AgentCallbackLogin(SIP/gc3-08c7ad58, 611222||[EMAIL 
PROTECTED]) in new stack
-- Playing 'agent-pass' (language 'en')
Dec  1 11:03:09 WARNING[30060]: chan_agent.c:1844 __login_exec: Extension 
'611222' is not valid for automatic login of agent '611222'


gary

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[asterisk-users] Need help on Music on Hold

2006-11-17 Thread gc 
I am testing asterisk (version 1.2.12.1) on a Dell 1950 server and have this 
strange problem on music on hold.
When I called into a queue using SIP from PSTN line which goes through our 
cisco gateway (cisco 5300), asterisk will start play music on hold. But this 
MOH seems at voice activation mode. That is only when I make noice on my end 
then I can hear music otherwise I will hear silence. I have another asterisk 
(version 1.2.9.1) running on an older Dell server and MOH works fine for call 
from PSTN. So my guess is that maybe there is some settings in asterisk cause 
this problem.

Any suggestion about this problem?

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[asterisk-users] How to setup announce attibute in queues.conf

2006-09-12 Thread gc 



I have this line in my queues.conf:
announce= support-department
and I have an recording file 
support-department-recording.wav file.
Can anybody tell me how to setup support-department 
so it play the .wav file when agent pickup the phone? Where should I define 
support-department so asterisk will play support-department-recording.wav? Is 
this in musiconhold.conf?

gc

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[asterisk-users] Use PauseQueueMember

2006-09-08 Thread gc 



After using PauseQueueMember in my dialplan. 
I used 'show agents' cli to show the agent status. It is still show that agent 
available. Here is the output from asterisk console:

5156597 (Agent1 ) available at '[EMAIL PROTECTED]' 
(musiconhold is 'default')5156598 (Agent2 ) not logged in 
(musiconhold is 'default')

Although Agent1 is indeed in pause 
mode.

Here is my dialplan:
exten = 
881112,1,PauseQueueMember(|Agent/${CALLERIDNUM})exten = 
881112,n,Playback(vm-goodbye)exten = 881112,n,Hanup
Am I doing somthing wrong? 

I am using asterisk 1.2.9.1



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Re: [asterisk-users] Use PauseQueueMember

2006-09-08 Thread gc 

The calling extension is 5156598.
After I dial into 881112 from this phone. It no longer accept call from
queue but the 'show agents' still show it is available.

- Original Message - 
From: Julian Lyndon-Smith [EMAIL PROTECTED]

To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, September 08, 2006 1:50 PM
Subject: Re: [asterisk-users] Use PauseQueueMember




gc wrote:

 After using PauseQueueMember in my dialplan. I used 'show agents' cli to
show the agent status. It is still show that agent available. Here is the
output from asterisk console:
 5156597   (Agent1 ) available at '[EMAIL PROTECTED]'
mailto:'[EMAIL PROTECTED]' (musiconhold is 'default')
5156598   (Agent2 ) not logged in (musiconhold is 'default')
 Although Agent1 is indeed in pause mode.
 Here is my dialplan:
exten = 881112,1,PauseQueueMember(|Agent/${CALLERIDNUM})


What's the extension of the calling phone ?
Unless you are calling from Extension 1, this won't work. you need
something like
exten = 881112,1,PauseQueueMember(|Agent/1})


exten = 881112,n,Playback(vm-goodbye)
exten = 881112,n,Hanup
Am I doing somthing wrong?
 I am using asterisk 1.2.9.1


Julian.


 

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[asterisk-users] Got error when compiling asterisk 1.2.11

2006-08-31 Thread gc 




I got follwing error when tried to compile asterisk 
1.2.11 on redhat linux 9:
make[1]: Entering directory 
`/home/voipuser/asterisk-1.2.11/db1-ast'make[1]: `libdb1.a' is up to 
date.make[1]: Leaving directory 
`/home/voipuser/asterisk-1.2.11/db1-ast'make[1]: Entering directory 
`/home/voipuser/asterisk-1.2.11/stdtime'make[1]: `libtime.a' is up to 
date.make[1]: Leaving directory 
`/home/voipuser/asterisk-1.2.11/stdtime'for x in res channels pbx apps 
codecs formats agi cdr funcs utils stdtime; do make -C $x || exit 1 ; 
donemake[1]: Entering directory 
`/home/voipuser/asterisk-1.2.11/res'make[1]: Nothing to be done for 
`all'.make[1]: Leaving directory 
`/home/voipuser/asterisk-1.2.11/res'make[1]: Entering directory 
`/home/voipuser/asterisk-1.2.11/channels'gcc -c -pipe -Wall 
-Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 
-DZAPTEL_OPTIMIZATIONS 
-fomit-frame-pointer -Wno-missing-prototypes -Wno-missing-declarations 
-DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC -o chan_zap.o 
chan_zap.cchan_zap.c: In function `pri_dchannel':chan_zap.c:9025: 
structure has no member named `call'make[1]: *** [chan_zap.o] Error 
1make[1]: Leaving directory 
`/home/voipuser/asterisk-1.2.11/channels'make: *** [subdirs] Error 
1
How can I fix it?

gc
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[asterisk-users] Question about context for incoming calls

2006-08-28 Thread gc 



I am new to asterisk. After studying the book and 
the tutorial on the Internet, I am still confued about how to use context for 
incoming calls. 
Here is my question:
If I create s extensions in two different contexts 
for incoming calls which one will be used? When a call comes in, which context 
will asterisk use to search the extension match first? 

Gary 
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Re: [Asterisk-Users] Error on using queue.

2005-12-02 Thread gc 



My aterisk is working now. I had some spelling 
mistakes in queues.conf. 
Thanks for your help.

  - Original Message - 
  From: 
  Dov Bigio 

  To: gc ; Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Thursday, December 01, 2005 12:22 
  PM
  Subject: Re: [Asterisk-Users] Error on 
  using queue.
  
  How is your agents.conf ? How is your login in 
  extensions.conf?
  
- Original Message - 
From: 
gc 
To: Dov Bigio ; Asterisk Users Mailing List - 
Non-Commercial Discussion 
Sent: Thursday, December 01, 2005 2:53 
PM
Subject: Re: [Asterisk-Users] Error on 
using queue.

Thanks. I made change to joinempty=yes. And now 
I can hear the music on hold. But it would not ring the agent even if I 
login agent in. When I run show queue command under CLI, I got these 
messages:
queue1 has 
1 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:2, 
SL:0.0% within 0s Members: 
Agent/555997 (Unavailable) has taken no calls 
yet Agent/555998 (Unavailable) has 
taken no calls yet
It seems that something wrong with my config 
file, it did not login any agent.



  - Original Message - 
  From: 
  Dov Bigio 
  
  To: Asterisk Users Mailing List 
  - Non-Commercial Discussion 
  Sent: Thursday, December 01, 2005 
  8:33 AM
  Subject: Re: [Asterisk-Users] Error 
  on using queue.
  
  If you are using 1.2, it might be the 
  joinempty and leavewhenempty parameters.
  Their default are different than the 1.0.x 
  releases
  
- Original Message - 
From: 
gc 

To: Asterisk Users Mailing 
List - Non-Commercial Discussion 
Sent: Thursday, December 01, 2005 
11:27 AM
Subject: [Asterisk-Users] Error on 
using queue.

I am trying to use * as ACD server for our 
sip proxy.
I first dial 55 to login 98 
as ACD agent it worked fine and then when I dialed 98, 
I got these messages from * 
CLI:

 -- Executing 
Answer("SIP/98-f718", "") in new stack -- 
Executing Ringing("SIP/98-f718", "") in new 
stack -- Executing Wait("SIP/98-f718", 
"2") in new stack -- Executing 
Queue("SIP/98-f718", "queue1") in new stackNov 30 16:54:12 
WARNING[7579]: app_queue.c:3078 queue_exec: Unable to join queue 
'queue1' -- Executing 
Hangup("SIP/98-f718", "") in new stack == Spawn 
extension (default, 99, 5) exited non-zero on 
'SIP/5025155598-f718'
Can anybody tell me what cause this 
problem?
The followings are my configuration 
files:

extensions.conf:
[default]
;For incoming call to ring into the 
queue.exten= 99,1,Answerexten= 
99,2,Ringingexten= 99,3,Wait(2)exten= 
99,4,Queue(queue1)exten= 
99,5,Hangup
;Agent loginexten = 
55,1,AgentCallBackLogin(|[EMAIL PROTECTED]);Agent 
logoutexten = 55,1,AgentCallBackLogin(|1)

exten = 
97,1,Dial(SIP/97)exten = 
98,1,Dial(SIP/98)

agents.conf:
[Agent1]agent = 
97,,Gary1agent = 98,,Gary2

queues.conf:
[queue1]musiconhold = 
defaultstrategy=ringalltimeout=15retry=5wrapuptime=0maxlen 
= 0announce-frequency = 0announce-holdtime = nomember = 
Agent1/555997member = Agent1/555998
sip.conf:
port=5060bindaddr=192.168.111.11context=defaultallow=ulaw

[97]type=friendusername=97insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2

[98]type=friendusername=98insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2











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[Asterisk-Users] Error on using queue.

2005-12-01 Thread gc 



I am trying to use * as ACD server for our sip 
proxy.
I first dial 55 to login 98 as ACD 
agent it worked fine and then when I dialed 98, I got these messages from * CLI:

 -- Executing 
Answer("SIP/98-f718", "") in new stack -- 
Executing Ringing("SIP/98-f718", "") in new stack 
-- Executing Wait("SIP/98-f718", "2") in new stack 
-- Executing Queue("SIP/98-f718", "queue1") in new stackNov 30 
16:54:12 WARNING[7579]: app_queue.c:3078 queue_exec: Unable to join queue 
'queue1' -- Executing Hangup("SIP/98-f718", "") in 
new stack == Spawn extension (default, 99, 5) exited non-zero 
on 'SIP/5025155598-f718'
Can anybody tell me what cause this 
problem?
The followings are my configuration 
files:

extensions.conf:
[default]
;For incoming call to ring into the 
queue.exten= 99,1,Answerexten= 
99,2,Ringingexten= 99,3,Wait(2)exten= 
99,4,Queue(queue1)exten= 99,5,Hangup
;Agent loginexten = 
55,1,AgentCallBackLogin(|[EMAIL PROTECTED]);Agent 
logoutexten = 55,1,AgentCallBackLogin(|1)

exten = 
97,1,Dial(SIP/97)exten = 
98,1,Dial(SIP/98)

agents.conf:
[Agent1]agent = 
97,,Gary1agent = 98,,Gary2

queues.conf:
[queue1]musiconhold = 
defaultstrategy=ringalltimeout=15retry=5wrapuptime=0maxlen = 
0announce-frequency = 0announce-holdtime = nomember = 
Agent1/555997member = Agent1/555998
sip.conf:
port=5060bindaddr=192.168.111.11context=defaultallow=ulaw

[97]type=friendusername=97insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2

[98]type=friendusername=98insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2








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Re: [Asterisk-Users] Error on using queue.

2005-12-01 Thread gc 



Thanks. I made change to joinempty=yes. And now I 
can hear the music on hold. But it would not ring the agent even if I login 
agent in. When I run show queue command under CLI, I got these 
messages:
queue1 has 1 
calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:2, 
SL:0.0% within 0s Members: 
Agent/555997 (Unavailable) has taken no calls 
yet Agent/555998 (Unavailable) has taken 
no calls yet
It seems that something wrong with my config file, 
it did not login any agent.



  - Original Message - 
  From: 
  Dov Bigio 

  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Thursday, December 01, 2005 8:33 
  AM
  Subject: Re: [Asterisk-Users] Error on 
  using queue.
  
  If you are using 1.2, it might be the joinempty 
  and leavewhenempty parameters.
  Their default are different than the 1.0.x 
  releases
  
- Original Message - 
From: 
gc 
To: Asterisk Users Mailing List - 
Non-Commercial Discussion 
Sent: Thursday, December 01, 2005 11:27 
AM
Subject: [Asterisk-Users] Error on 
using queue.

I am trying to use * as ACD server for our sip 
proxy.
I first dial 55 to login 98 as 
ACD agent it worked fine and then when I dialed 98, I got these messages from * CLI:

 -- Executing 
Answer("SIP/98-f718", "") in new stack -- 
Executing Ringing("SIP/98-f718", "") in new 
stack -- Executing Wait("SIP/98-f718", "2") in 
new stack -- Executing Queue("SIP/98-f718", 
"queue1") in new stackNov 30 16:54:12 WARNING[7579]: app_queue.c:3078 
queue_exec: Unable to join queue 'queue1' -- Executing 
Hangup("SIP/98-f718", "") in new stack == Spawn extension 
(default, 99, 5) exited non-zero on 
'SIP/5025155598-f718'
Can anybody tell me what cause this 
problem?
The followings are my configuration 
files:

extensions.conf:
[default]
;For incoming call to ring into the 
queue.exten= 99,1,Answerexten= 
99,2,Ringingexten= 99,3,Wait(2)exten= 
99,4,Queue(queue1)exten= 99,5,Hangup
;Agent loginexten = 
55,1,AgentCallBackLogin(|[EMAIL PROTECTED]);Agent 
logoutexten = 55,1,AgentCallBackLogin(|1)

exten = 
97,1,Dial(SIP/97)exten = 
98,1,Dial(SIP/98)

agents.conf:
[Agent1]agent = 
97,,Gary1agent = 98,,Gary2

queues.conf:
[queue1]musiconhold = 
defaultstrategy=ringalltimeout=15retry=5wrapuptime=0maxlen 
= 0announce-frequency = 0announce-holdtime = nomember = 
Agent1/555997member = Agent1/555998
sip.conf:
port=5060bindaddr=192.168.111.11context=defaultallow=ulaw

[97]type=friendusername=97insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2

[98]type=friendusername=98insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2











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[Asterisk-Users] How to exit from Asterisk console.

2005-11-30 Thread gc 



I am new to Asterisk.
Asterisk 1.2

I started * like this: asterisk 
-vgc
now I am in CLI mode: *CLI

How do I get out this CLI mode to linux shell 
without kill asterisk process?

I tried EXIT, QUIT, exit and quit. None of them 
work.

If I use ^c, this also kill asterisk 
process.

GC
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[Asterisk-Users] Remove older version of Asterisk

2005-11-18 Thread gc 




Ihave an older version (0.9.0)of 
Asterisk on my linux box. Do I need to remove it before I install version 1.2? 
How do I remove it? Does Asterisk make file contain the uninstall process? Or I 
have to manully remove all the directory structure. 

Gary
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[Asterisk-Users] Using asterisk as voicemail system for SER

2004-06-09 Thread gc 



I ma new to Asterisk.

I'd like to setup * as voicemail system for SER.
Let's say I have an phone number registered in ser as 5554321. 
When somebody dial to ser for this number and nobody answer, the ser will 
forward the call to asterisk and get into voicemail box 5554321. I already 
have asterisk up and running with mysql setup for asterisk 
voicemail.
Can somebody show me how to do it? Or show me some examples 
ofsip.conf, voicemail.conf and extensions.conf.

Gary