Re: [asterisk-users] mISDN problem
On 6/13/07, Josu Lazkano <[EMAIL PROTECTED]> wrote: > > How can I saw the status of the ISDN??? > ...try "misdn show stacks" or "misdn show config". You can also increase debug level in /etc/misdn-init.conf... Output will end up in /var/log/asterisk/misdn.log Cheers, -- Ex Vito ___ --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
Re: [asterisk-users] mISDN problem
You have only one extension in the [incoming] context and that is 's'. You probably need a different one -- the one the telco sends you... Ideas: 1. Try using a generic wildcard such as '_X.' instead of 's', then check the CLI after incrementing verbosity to at least 3 (BTW: don't forget reloading extensions!) 2. Enable misdn debugging to leve 3 and check its log at /var/log/asterisk/misdn.log. You will have the "destination extension" as the "dad" field, IIRC. Good luck -- Ex Vito On 6/20/07, Josu Lazkano <[EMAIL PROTECTED]> wrote: > Hello everybody. > > I have an other problem with mISDN. > The outgoing calls goes perfect, but the incoming no. > > When people call in the CLI puts that: > > *CLI> Jun 20 12:32:08 WARNING[2315]: chan_misdn.c:4920 chan_misdn_log: > Extension can never match, so disconnecting > ___ --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
Re: [asterisk-users] Forcing Dial application to skip if called server is unreachable
...not really sure, maybe ChanIsAvail can be of use ? -- Ex Vito On 6/20/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Is it possible to force the Dial function to skip to the next priority if it > doesn't find the server of the called contact within a few seconds? > > I know I can use: > Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]) > where I can use some short timeout in the "timeout" option, but if I do so, > when some call is well succeeded, it will only ring for that time! > > Any ideas? > ___ --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
[asterisk-users] PCI, PCI-X and PCI-e -- Server / Interface Card Selection
Hello List, The main issue is server selection regarding PCI bus connectivity for Asterisk solutions. Most offerings on HP Proliants, something I've been looking into, include PCI-X and/or PCI-e expansion slots. I know PCI-e is totally different from PCI and PCI-X so, for now, that's not an issue. However, regarding PCI and PCI-X, and after googling for a while and checking wikipedia and whatnot, I'm still not clear on my main issue: Can one use a PCI interface card in a PCI-X slot ? If so, under what conditions ? (ex: 3.3v cards only ? PCI-X bus speed is brought down ? what ?) The objective would be to use Digium's echo cancelling PRI and BRI cards and/or beroNet's BRI cards on, for example, a Proliant ML350 G4 or G5 containing PCI-X slots -- would such combination be technically feasible ? If not, where are you guys getting servers for your PCI based solutions ? Can anyone shed some light into my doubts ? Pointers, documentation, experiences ? The second, kind of "attatched" issue, is associated to the growing PCI-e buses in the current servers. I know Sangoma already has a PRI card for PCI-e. What about Digium ? Other PRI, BRI manufacturers ? Thanks in advance and regards to all, -- Ex Vito ___ --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
Re: [asterisk-users] PCI, PCI-X and PCI-e -- Server / Interface Card Selection
...well, thanks Andrew + Thomas, but that's exactly what I am trying to avoid: knowing by trying ! ;-) I can't risk spending a few thousand just to reach the conclusion that Digium's PRI or BRI cards do not work with a particular system's PCI-X slots/bus... Or, worse, staying with a dead card / system board in my hands ! :-( Anyone ? Thanks + regards, -- Ex Vito On 12/10/06, Thomas Kenyon <[EMAIL PROTECTED]> wrote: Andrew D Kirch wrote: > The rule of thumb is "if it fits you can use it" unless it doesn't work, > there are few that won't (Creative's soundcards being an example of ones > that don't) > I remember reading up on it and (other than there being 2 different types of PCI-66 slots and then there's the PCI-100 and PCI-X ones), discovering that a particular PCI card should work in a PCI slot. When I put the card in I blew a £300 server board. Oops. ___ ___ --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
[asterisk-users] Queue Monitor not mixing if using UNIQUEID in MONITOR_FILENAME
Hello Asterisk Users,
I guess the subject says the most of it; here goes some more
detail:
- Running Asterisk 1.2.14
- Objective: record all calls managed by a specific queue
- Name those files ${TIMESTAMP}-${CALLERIDNUM}-${UNIQUEID}
Facts:
- If the UNIQUEID chan var is used in the MONITOR_FILENAME,
before calling the Queue() application, the two legs of the call are
not mixed and I end up with the two separate -in / -out files
- If it isn't used (for example: using only ${TIMESTAMP]-${CALLERIDNUM})
then, the legs are mixed together...
Note:
- In my first attempt I never managed to get the legs mixed... Only
after some experiment, I understood (well, not 100% clear why!)
that I had to also to add to include "recordagentcalls=yes" and
"monitor-join=yes" in agents.conf !
Can anyone provide some insight into this ? Thanks in advance!
(see below for config)
--
Ex Vito
queues.conf:
[general]
persistentmembers = yes
[the_queue]
musiconhold = default
announce = the_announcement
strategy = ringall
servicelevel = 20
context = the_context
wrapuptime = 10
announce-frequency = 30
announce-holdtime = once
monitor-format = wav
monitor-join = yes
eventwhencalled = yes
eventmemberstatus = no
reportholdtime = no
member => SIP/sip0001
agents.conf:
[general]
persistentagents=yes
recordagencalls=yes
monitor-join = yes
[agents]
(no agents declared, as they are directly configured in the
queues.conf file)
extensions.conf:
...
[globals]
SUPPORT_MONITOR_PATH=/var/spool/asterisk/monitor/support
[the_context]
exten => 305,1,Answer()
exten =>
305,n,Set(MONITOR_FILENAME=${SUPPORT_MONITOR_PATH}/${TIMESTAMP}-${CALLERIDNUM})
exten => 305,n,Queue(the_queue,t)
exten => 305,n,Hangup()
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[asterisk-users] Queue Monitor not mixing if using UNIQUEID in MONITOR_FILENAME
(1st attempt was rejected by postfix @lists.digium.com, here goes the 2nd)
-- Forwarded message --
From: Ex Vitorino <[EMAIL PROTECTED]>
Date: Dec 18, 2006 11:41 PM
Subject: Queue Monitor not mixing if using UNIQUEID in MONITOR_FILENAME
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello Asterisk Users,
I guess the subject says the most of it; here goes some more
detail:
- Running Asterisk 1.2.14
- Objective: record all calls managed by a specific queue
- Name those files ${TIMESTAMP}-${CALLERIDNUM}-${UNIQUEID}
Facts:
- If the UNIQUEID chan var is used in the MONITOR_FILENAME,
before calling the Queue() application, the two legs of the call are
not mixed and I end up with the two separate -in / -out files
- If it isn't used (for example: using only ${TIMESTAMP]-${CALLERIDNUM})
then, the legs are mixed together...
Note:
- In my first attempt I never managed to get the legs mixed... Only
after some experiment, I understood (well, not 100% clear why!)
that I had to also to add to include "recordagentcalls=yes" and
"monitor-join=yes" in agents.conf !
Can anyone provide some insight into this ? Thanks in advance!
(see below for config)
--
Ex Vito
queues.conf:
[general]
persistentmembers = yes
[the_queue]
musiconhold = default
announce = the_announcement
strategy = ringall
servicelevel = 20
context = the_context
wrapuptime = 10
announce-frequency = 30
announce-holdtime = once
monitor-format = wav
monitor-join = yes
eventwhencalled = yes
eventmemberstatus = no
reportholdtime = no
member => SIP/sip0001
agents.conf:
[general]
persistentagents=yes
recordagencalls=yes
monitor-join = yes
[agents]
(no agents declared, as they are directly configured in the
queues.conf file)
extensions.conf:
...
[globals]
SUPPORT_MONITOR_PATH=/var/spool/asterisk/monitor/support
[the_context]
exten => 305,1,Answer()
exten =>
305,n,Set(MONITOR_FILENAME=${SUPPORT_MONITOR_PATH}/${TIMESTAMP}-${CALLERIDNUM})
exten => 305,n,Queue(the_queue,t)
exten => 305,n,Hangup()
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Re: [asterisk-users] Queue Monitor not mixing if using UNIQUEID in MONITOR_FILENAME
James,
Thanks a lot for sharing the result of those debugging hours ! :-)
I'm now left with two choices to begin with:
1. Replacing the "." with a "-" within the dialplan
2. Replacing the "Ubuntu Server" packaged sox version (12.17.9)
with the most recent (12.18.2) which no longer seems to
suffer from that sillyness...
(yes, I did a quick new sox download/compile/test in a separate system
and "soxmix file1.this.ext file2.that.ext mix.good.ext" started working
with the new version !)
Kind Regards,
--
Ex Vito
On 12/19/06, James Fromm <[EMAIL PROTECTED]> wrote:
I spent hours debugging this a few weeks ago.
The ${UNIQUEID} contains a period ("."). Mine are something like
.xx. When soxmix is executed to mix the in and out files, the
file types are not specified. This causes soxmix to attempt to
determine the file type by the filename's extension. The routine in sox
that looks for the filename's extension doesn't expect multiple periods
in the filename. So it finds the file type to be xx.wav (or xx.gsm) and
that's not a format sox can handle.
You can add an AGI call to your dialplan immediately after the Queue
application to join the files.
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Re: [asterisk-users] MixMonitor and Queues
Jay,
I had a similar issue recently... My filename had more than one "."
(dot / period)
and the sox version I was using failed to mix files in such conditions...
If that is your case, try:
- Using a filename with no "."
- Upgrade sox to the latest version which fixes the funny behaviour
Cheers,
--
Ex Vito
On 12/28/06, Jay Moore <[EMAIL PROTECTED]> wrote:
Recompiled Asterisk after installing sox and it's still not merging the
two streams into a single recorded file. What am I doing wrong?
Jay
Jay Moore wrote:
> Ed,
>
> Thanks for the help. One more question, however. Everything is working
> fine with the exception of sox joining the calls. I have sox installed
> and monitor-join set to yes in both queues.conf and agents.conf
>
> I installed sox after I installed Asterisk. Do I need to recompile
> Asterisk for it to work with sox?
>
> This is the last hurdle I need to overcome (I hope) before I can use my
> Asterisk box in a live situation. Any help would be much appreciated.
>
> Regards,
> Jay
>
> Ed Nuñez wrote:
>> In queues.conf you must have the following under the queues you want
>> to record.
>>
>> monitor-format=wav49 ; you may also use wav or gsm formats
>> monitor-join=yes; if you have the latest sox installed,
>> this will join the in and out files into one.
>>
>> In agents.conf
>>
>> recordagencalls=yes
>> monitor-join = yes
>> recordformat=wav49
>> savecallsin=/var/www/html/calls;this is the path where call
>> will be recorded.
>>
>> That's all
>>
>> If you want to change the file name place this in your extensions.conf
>> on a line prior to sending the call to the queue.
>>
>> exten=> 1097,4,Set(MONITOR_FILENAME=QUEUE-${CALLERID}-${TIMESTAMP})
>>
>>
>> Ed Nuñez
>> IT/Telecom Engineer
>>
>> 4037 Metric Drive
>> Winter Park, FL
>>
>> (o) 407-384-4200 x 1656
>> (f) 407-384-4222
>> (c) 732-925-0730
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore
>> Sent: Wednesday, December 13, 2006 10:15 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] MixMonitor and Queues
>>
>> Greetings, all.
>>
>> I would like to record calls that are entered into queues and I'm not
>> quite sure how to do it. Here's how I'm currently set up:
>>
>> - Call comes in and is placed into Queue #1 (which rings all phones
>> for 15 sec).
>> - If call drops out of this queue, it is placed into Queue #2 (which
>> plays MoH until the call is picked up).
>>
>> I've tinkered with MixMonitor and I have my queues set up, but I'm not
>> sure how to combine the two. Ideally, I'd like to only record once
>> the call comes out of queue (no point in recording hold music, unless
>> I want to hear people mumble about how lousy a company we are for
>> placing them on hold ;) )
>>
>> On a semi-related note, is it possible to determine the extension that
>> pull the call out of queue before the call is bridged? The reason I
>> ask is that I'd like to put the receiving extension in the name of the
>> file that MixMonitor creates. If not, no biggie.
>>
>> Recap:
>>
>> Two queues. First rings for 15 seconds then drops into the second.
>> Second plays music on hold till the call is answered. I want to
>> record the call when it's pulled out of either queue using
>> MixMonitor. Bonus points if I can determine the answering extension
>> before MixMonitor starts (if possible).
>>
>> Any help would be greatly appreciated.
>>
>> Thanks,
>> Jay
>> ___
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>> asterisk-users mailing list
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
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>>
>>
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>
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Re: [asterisk-users] queues - limiting ringing calls to queue members
Nikola, Check the maxlen parameter for the queue... Also check the sample queues.conf distributed with Asterisk source, which somehow includes queue parameter documentation. If set, maxlen will limit the number of calls in the queue. Cheers, -- Ex Vito On 1/2/07, Nikola Ciprich <[EMAIL PROTECTED]> wrote: Hello, I'm using asterisk queues, for reception phone, and I have small problem: I have only one phone as queue member, and the problem is, that ALL channels waiting in queue are ringing on it. And if there are too many people ringing on it, it's not possible to use attended transfer then... Is it possible to limit maximum ringing calls from queue? or some other tip? thanks a lot in advance! best regards Nik ___ --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
Re: [asterisk-users] Some queries on g729 license.
/var/lib/asterisk/licenses :-) On 1/8/07, Xue Liangliang <[EMAIL PROTECTED]> wrote: Hi, leo, I will try the following solution that seperate /usr/lib/asterisk/modules in another patition other than drbd, then register the licenses on both server. not sure where the license key acutally lies in? Regards, Liangliang Leo Ann Boon wrote: > Xue Liangliang wrote: > >> Hi, actutally it is kind of shareing storage, because we use drbd and >> vserver technology, the fail over is at vserver level, and vserver is >> synced through drbd storage. > > drdb - that's what I suspected. Off the top of my head, the fastest > way is to reactivate using the new master's MAC. The proper solution > is to only use drdb for data that should be shared like the conf and > database. The license key portion should not be on a device that's > being mirrored by drdb. > > Leo > > ___ > --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 > ___ --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 ___ --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
Re: [asterisk-users] Manage 'full' log file
We've been using logrotate without any issue... We're using
the below quoted configuration. Notice the invocation of
Asterisk's CLI "logger reload" command so as to close the
old files and open new ones.
Cheers,
--
Ex Vito
/var/log/asterisk/messages /var/log/asterisk/queue_log
/var/log/asterisk/event_log {
weekly
rotate 52
dateext
compress
delaycompress
nocreate
missingok
sharedscripts
postrotate
/usr/sbin/asterisk -rx "logger reload"
endscript
}
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Re: SV: [asterisk-users] Manage 'full' log file
On 1/8/07, lenz <[EMAIL PROTECTED]> wrote: You know that if you rename an open Unix file, it will stay open - i.e. if you rename the logfile "full" to "full.1", Asterisk will continue writing to "full.1" thinking it was "full". The "logger rotate" command forces all log files to be closed and reopened with their canonical names, so your file is actually rotated. Hope this helps l. CORRECT about UNIX files, INCORRECT about "logger rotate" command. CORRECTION: "logger rotate" does: 1. Closes the files 2. Renames them (actually rotating them) 3. Reopnes the canonical named files "logger reload" does effectively work as you described: 1. Closes files 2. Reopens canonical named files This is the command that should be used with logrotate, for example. EXAMPLE: (consider asterisk running and writing to messages file) # cd /var/log/asterisk # mv messages messages.old (asterisk still running and now writing to messages.old) (there is no file named messages) # /usr/sbin/asterisk -rx "logger reload" (asterisk closed messages.old and created a new messages file) Hope this is clear enough as its really late now... Please correct me if I'm wrong Cheers -- Ex Vito ___ --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
[asterisk-users] Queues without music on hold ?
Hello List, This must be an easy one... I'd like to setup a queue without music on hold - just give the callers the traditional ringing tones. However, not setting the musiconhold parameter in queues.conf does not seem to do the trick: it defaults to "default" moh class which: a) Gets played if it exists b) Doesn't get played if it doesn't, but the caller still gets no ringing tones Any ideas ? Thanks in advance, and kind regards, -- Ex Vito ___ --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
[asterisk-users] Trying to understand mISDN kernel buffer messages...
Hello List, I have got some questions about mISDN kernel buffer messages which I posted to the mISDN users list a while ago. Unfortunatelly I received no feedback at all, so I though I'd share them in this (wider) context to see if anyone can give any input... Thanks in advance for any feedback and kind regards, -- Ex Vito -- Forwarded message -- From: Ex Vitorino <[EMAIL PROTECTED]> Date: Jan 2, 2007 1:22 PM Subject: Trying to understand kernel buffer messages... To: misdn-asterisk@lists.beronet.com Hello list, I've been successfully running an Asterisk 1.2.14 + bundled chan_misdn + mISDN 1.0.4 + mISDNuser 1.0.3 + beroNet 4 port BRI card solution for about two weeks. It's running on top of Ubuntu Server 6.06 LTS, kernel 2.6.15-26-server and everything installed, to my knowledge, without a hitch. After this period, the kernel buffer is holding several different mISDN related messages which I'd like to fully understand. Can anyone shed some light, please ? 1. These show up on boot Modular ISDN Stack core $Revision: 1.37 $ mISDNd: kernel daemon started (current:f7e53a90) mISDNd: test event done mISDN_isac: disagrees about version of symbol mISDN_debugprint mISDN_isac: Unknown symbol mISDN_debugprint netjetpci: Unknown symbol mISDN_isac_free netjetpci: Unknown symbol mISDN_isac_interrupt netjetpci: Unknown symbol mISDN_clear_isac netjetpci: Unknown symbol mISDN_isac_init netjetpci: Unknown symbol mISDN_ISAC_l1hw ...something about mISDN_isac and netjetpci does not seem right; neither of the modules is currently loaded. 2. MISDN free_device: entitylist not empty Does this indicate some cleaning up did no go as it should ? They seem to appear everytime Asterisk is restarted (not shure if on shutdown or on startup!) 3. ie not handled ie [76] l [b] I've got 7 of these: all of them have the [76], two have [b], two have [2] and three have [5]... I can't associate any of them to a particular Asterisk log event but, honestly, this seems to indicate something did not go as planned ! :-( 4. Found ie in set which we do not support ie [27] Once. Right after one of the previous messages "ie not handled ie [76] l [2]". Again, what went wrong here ? 5. These ones seem related and show up in pairs, like: mISDN_FsmAddTimer: timer already active! lapd 1 mISDN_FsmAddTimer already active! ...or... mISDN_FsmAddTimer: timer already active! lapd 2 mISDN_FsmAddTimer already active! I've got 4 of these pairs... None of them show up on times which can be related to any specific Asterisk activity. 6. Last, but not least, this one... DSS1 1 mISDN dss1 sending RELEASE_COMPLETE without proc pr=35a80 dinof(90001) ...which I already know shows up after every call. The question: does it make any sense to fill the kernel message buffer with them ? Shouldn't they rather be sent to a log file only, and only if activated ? I'd like to thank everyone in advance for any help or pointers they can provide. The objective is simple: have a good understanding of what is going on, so as to keep the system running as smoothly as possible. Kind regards and Happy New Year, -- Ex Vito ___ --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
Re: [asterisk-users] Problems with mISDN TE line
I've been through this once and, IIRC, I'd say your [outside] context in the dialplan does not have an extension matching the incoming msn. Suggestion: - Recheck your extensions.conf [outside] context Otherwise: - Increase misdn debug to level 3 (something like "misdn set debug 3" at the CLI) - Then share the misdn.log + [outside] context dialplan with the list so that further feedback can be provided Cheers + Good luck, -- Ex Vito ___ --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
Re: [asterisk-users] About BRI / ISDN hardware. What to buy?
On 1/18/07, Cosmin Prund <[EMAIL PROTECTED]> wrote: How about the "Digium Wildcard B410P" card? It seems to be Digium, it has hardware echo cancel and I can buy this in Romania. Is this card any good? ...well, it's essentially a 4 port HFC based card with builtin HW echo canceler. I'd say if Digium backs it up, it should be reasonable... (note: I'm running a 4 port HFC from beroNet perfectly fine) Good Luck, -- Ex Vito ___ --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
Re: [asterisk-users] Test to Speech
> > Someone has worked with any test to speech software with aceptable > quality in spanish? Probably in english the text to speech quality > will be better. > Witch test to speech software gave you the best results in spanish? > Hi Andres, Check www.loquendo.com out... They have a nice web front end for demoing their product's abilites. Good Luck, -- Ex Vito ___ --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
Re: [asterisk-users] Sending sound to an open channel....
In a dialplan, after i set an autohangup (with AGI) , how could i send a sound (stream a sound ) into an open channel at X seconds before the autohangup time get to 0 for that channel? (Like public phones, that gives u a 'beep!!!' before ur time runs out, just like that...) Check the L option to the Dial application... Try "show application dial" at the Asterisk CLI. My guess is that this is exactly what you want. Good Luck, -- Ex Vito ___ --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
Re: [asterisk-users] Queues do not accept calls if all agent are busy?
On 2/15/07, Angel Heart <[EMAIL PROTECTED]> wrote: cud any one help me figuring out the problem... When the agent in a queue is engaged, it cannot accept anymore calls, below is the script; Angel, Check your queues.conf, specifically the "joinempty" parameter. See below the relevant part in the queues.conf sample file: ... ; This setting controls whether callers can join a queue with no members. There ; are three choices: ; ; yes- callers can join a queue with no members or only unavailable members ; no - callers cannot join a queue with no members ; strict - callers cannot join a queue with no members or only unavailable ; members ; ; joinempty = yes ... Cheers, -- Ex Vito ___ --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
