I think you may have missed a few files...
[EMAIL PROTECTED] asterisk]# ls -lR /root/sounds | wc -l
372
[EMAIL PROTECTED] asterisk]# ls -lR sounds | wc -l
1710
Looks like the original number of files is 1710, but the new ulaw format files
only number 372...
Doug.
-Original Message-
-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk native sounds now available!
Douglas Garstang wrote:
I think you may have missed a few files...
[EMAIL PROTECTED] asterisk]# ls -lR /root/sounds | wc -l
372
[EMAIL PROTECTED] asterisk]# ls -lR sounds | wc -l
1710
Looks like
Stay away from Alliance Systems. We ordered $15k worth of Polycom's over a
month ago and we're still waiting. Our account rep's communication with us on
what the delay has been, has been terrible.
Doug.
-Original Message-
From: Gavin Adams [mailto:[EMAIL PROTECTED]
Sent: Friday,
We've using 1.6.3.0067, and not experiencing this problem.
-Original Message-
From: Ron Senykoff [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 26, 2006 2:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom 501 horrible echo
We
[mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 25, 2006 3:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ACD with polycom ip phones
On 1/24/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Thanks, but there's an avail/unavail button on the Polycom
Some questions regarding calling Fast AGI from the dial plan.
Considering that the server side of the Fast AGI has to be able to a) use
threading and b) connect to MySQL, this causes some serious limitations. I'm
not a C programmer, so development options are either perl or python.
It
On 1/25/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I've tried that. Setting acd-login-logout and acd-agent-available to
1 causes the appearance to automatically log in when the phones comes up, and
stays up the entire time. I'll have another shot it in a bit tho
Has anyone tried to (recently) install asterisk in a location not relative to
/, as a non-root user? Ie editting the PREFIX directive in Makefile.
Why? Several quite obvious reasons:
a). Allows an asterisk user to be created, and operators to log into the box as
asterisk user, without having
Has anyone tried to (recently) install asterisk in a location not relative to
/, as a non-root user? Ie editting the PREFIX directive in Makefile.
Why? Several quite obvious reasons:
a). Allows an asterisk user to be created, and operators to log into the box as
asterisk user, without having
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] Fast AGI Options. Eeeek!
On 1/25/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Some questions regarding calling Fast AGI from
Skip this post. My bad. I had an old asterisk in /usr/sbin and it was in the
path before $HOME:/usr/sbin. :) Still would like to know if anyone is running
asterisk as a non-root user unde /home/asterisk or similar. Thanks.
-Original Message-
From: Douglas Garstang
-Commercial Discussion
Subject: Re: [Asterisk-Users] MOH Server
I setup Slimserver to stream online radio stations to asterisk. Not all
online stations work, but once you find a couple good ones you just
stick with it.
Kyle
Douglas Garstang wrote:
Has anyone managed to set up a moh server
Why do you need a patch? We have ACD/Asterisk 1.2.1 working well with Polycom
IP phones. Haven't done much with 1.2.2 yet. Is there some sort of issue?
About the only thing that doesn't work is the appearances don't display the
login/logout status with the icon of an agent in an ACD Queue.
-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 24, 2006 12:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ACD with polycom ip phones
On 1/24/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Why do you need a patch? We have ACD
-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 24, 2006 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ACD with polycom ip phones
On 1/24/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Oo I thought that this was a Polycom
We conducted focus groups, looking at several different vendors, before we
decided to go with the Polycom. From the user interface perspective, the
Polycom's won hands down. I was never involved with it, but apparently to
configure the Cisco's you need to be converting hex??? Yuk!
Example:
log (L_INFO,test)
It will go to syslog, ie /var/log/messages.
Douglas.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Monday, January 23, 2006 6:48 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] debug with ser
hi
how can i
I'm trying to think of a way to store/represent the Asterisk .conf files. One
method is to store them in MySQL in some format, and then write some scripts to
query MySQL and generate the conf files before doing a reload.
MySQL is pretty heavy handed though. I'm looking for something a bit more
You aren't making calls from one phone to another, with them right next to each
other on the same desk are you?
Doug.
-Original Message-
From: Jeff Herring [mailto:[EMAIL PROTECTED]
Sent: Mon 1/23/2006 6:46 PM
To: asterisk-users@lists.digium.com
; Asterisk
Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List -
Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] Config File Storage
At 08:42 PM 1/23/2006, Douglas Garstang wrote:
Content-Class: urn:content
:[EMAIL PROTECTED]
Sent: Mon 1/23/2006 10:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: RE: [Asterisk-Users] Re: Polycom FW
On Mon, 23 Jan 2006, Douglas Garstang wrote:
We conducted
Polycom SoundPoint 601 has 4 'lines'. :)
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Mon 1/23/2006 9:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users]
Do the linksys phones support BLF? A lot of businesses require/expect BLF. Do
the linksys phones support Asterisk setting the ring-type to auto answer so
that you can do paging and intercom? Businesses expect this too.
-Original Message-
From: Cory Andrews
Has anyone managed to set up a moh server for Asterisk? Reason would be to
offload processing off the asterisk box, onto another system.
The wiki is a bit light on details. If anyone managed to get it up and working,
what software did you use on the server side, and what client app did you use?
Matt,
Wouldn't they have to actually enter a forwarded number for the forward to
activate? I've hit the forward button myself many times after a call ends, and
the phone asks you for a new number to forward to.
Douglas.
-Original Message-
From: Matt Darnell
I don't think you can beat the Polycom's for design, features, configuration
options and functionality tho. :)
-Original Message-
From: Wilson Pickett [mailto:[EMAIL PROTECTED]
Sent: Sun 1/22/2006 10:32 AM
To: Asterisk Users Mailing List -
You could also achieve the same result with phones that support some type of
failover, such as a simple list of systems to try in order, or DNS SRV lookups.
-Original Message-
From: Jon Radon [mailto:[EMAIL PROTECTED]
Sent: Sun 1/22/2006 4:27 PM
We purchased our phones through Alliance Systems, a Polycom certified reseller.
Getting firmware was difficult, and they where very unresponsive, probably
because we didn't pay them additional money for a support contract. Such is
life.
-Original Message-
From: The VoIP Connection
That's
really old SIP software.
-Original Message-From: Adam Dobrin
[mailto:[EMAIL PROTECTED]Sent: Friday, January 20, 2006 10:33
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionCc: [EMAIL PROTECTED]Subject:
Re: [Asterisk-Users] Polycom
Yes, heartbeat is good at monitoring system and network availability, but to
monitor applications as well, you need to jump through hoops and do some custom
development. A shame really because without that it's useless.
Also, heartbeat only works in a primary/secondary fashion. Ie you can't
Polycom are analy retentive when it comes to this. You should be able to get
the older versions on their web site though.
Doug.
-Original Message-
From: Bill Michaelson [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 19, 2006 2:13 PM
To: Asterisk Users Mailing List - Non-Commercial
The O'Reilly TFOT book is full of errors. Two that pop into my head instantly
are it's referring to regcontext being able to execute dialplan commands upon
SIP registration and it's use of auth= in sip.conf in the DUNDi section. I
wouldn't trust it.
-Original Message-
From: Leif Madsen
Has anyone gotten announce-holdtime in queues.conf to work? Doesn't seem to
matter what combination of options I use, I can't get this particular setting
to do what the docs say.
Thanks.
Doug.
___
--Bandwidth and Colocation provided by Easynews.com --
What's the maximum number of #include statments I can have in extensions.conf?
I'm getting an error at the 11th one. I tried breaking twelve #include's into 2
different contexts, and still got the same error. These aren't nested
includes... they're only one level deep.
Thanks,
Doug.
I was having problems too. Mine stopped at 5:19am MST this morning and just
picked up a few minutes ago. Isn't the first time it's happened either.
-Original Message-
From: Francesco Peeters [mailto:[EMAIL PROTECTED]
Sent: Monday, January 16, 2006 3:17 AM
To:
Looks like it's a bug.
If I have #include's going to non-existant files, Asterisk doesn't complain
that the file wasn't found. It just says that I have more than 10 includes.
Weird...
-Original Message-
From: Douglas Garstang
Sent: Monday, January 16, 2006 7:57 AM
To: Asterisk Users
Actually the docs for the Queue application say:
'w' -- allow the called user to write the conversation to disk via Monitor
'W' -- allow the calling user to write the conversation to disk via
Monitor
couldn't get these to work tho. Does this mean I can do one touch recording
with
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] Max Number of #include statements
Don't know the max, but I have many more than 11 with never a problem.
On Jan 16, 2006, at 8:57 AM, Douglas
Why not direct your question to Digium? It's their protocol after all. :)
-Original Message-
From: John Falk [mailto:[EMAIL PROTECTED]
Sent: Mon 1/16/2006 12:46 PM
To: asterisk-users@lists.digium.com
Cc:
Subject: [Asterisk-Users] Dundi
=_3xxx,2,Dial(SIP/${EXTEN},30,wth).
Hope this is of some help.
Regards
Jenn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Tuesday, January 17
Reinvite doesn't happen until after the call is picked up. After it's picked
up, new invites' are sent and the phones communicate directly. Sorry, I forget
the details. It was a few weeks ago.
Doug
-Original Message-
From: Mike Hammett [mailto:[EMAIL PROTECTED]
I've noticed that with FastAGI (and maybe AGI) that when you sequentially send
a sequence of dial commands, if the call is picked up, that after the call
ends, the Fast AGI script keeps executing the commands!
Is there anyway to stop execution once a call is picked up? I think looking at
the
[mailto:[EMAIL PROTECTED]
Sent: Friday, January 13, 2006 9:46 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: FastAGI Command Execution
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
I've noticed that with FastAGI (and maybe AGI) that when you
Does ael support #include statements yet?
Is there any way to perform a reload in asterisk and reload extensions.ael?
If both of these aren't available yet, then AEL isn't ready for real-world use.
Doug.
-Original Message-
From: Steve Murphy [mailto:[EMAIL PROTECTED]
Sent: Friday,
)
... won't work.
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Friday, January 13, 2006 10:37 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: FastAGI Command Execution
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote
Can someone tell me why the following from extensions.conf generates this error
on asterisk load?
Jan 13 09:27:24 WARNING[31701]: config.c:938 ast_config_internal_load: Maximum
Include level (10) exceeded
As far as I can tell I don't have a DEPTH of 10 includes. I certainly have more
than 10
increased this limit?
I'm flabbergasted. Please, someone tell me I have got this all wrong.
Doug.
-Original Message-
From: Douglas Garstang
Sent: Friday, January 13, 2006 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Extensions.conf error
fairly logical, and no problem , since if you need more than 9
includes per context, simply use goto's and jump out to other context's
or use AGI first to select a final context and return to the dial plan
So what was the issue again ?
Douglas Garstang
Subject: Re: [Asterisk-Users] Re: Nested MySQL Commands
Douglas Garstang wrote:
I also don't believe perl DBI is thread safe
The lastest docs says that DBI does support multithread connection
pooling. Otherwise, you are always free to implement your AGI in
'modern' :) programming languages
Andreas.
I tried that. Still didn't work. It just appears that Asterisk doesn't like
letting you execute another query while it's holding on to the state of a
previous one.
Doug.
-Original Message-
From: Andreas Sikkema [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 12, 2006 2:01
I have a fast agi python script that reads some numbers from MySQL, and then
instructs asterisk to try those numbers in sequential order.
ie:
def run(self):
agi = AGI(self.client)
db =
MySQLdb.connect(host=192.168.10.15,user=user,passwd=password,db=somedb)
c =
:
Douglas Garstang ha scritto:
So I really wish there was some way to measure how well the worst
case scenario would perform. This would be 120 simultaneous calls
(don't know how many per second) on a Dual 3.8Ghz Dell PowerEdge 1850
with 2GB RAM
No an option. Too slow and too resource intensive.
-Original Message-
From: Gonzalo Servat [mailto:[EMAIL PROTECTED]
Sent: Thu 1/12/2006 9:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re:
Polycom phones need a reboot after making configuration changes.
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 11, 2006 7:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Why remotely reboot SIP
Is it
possible to have nested MySQL queries in extensions.conf?
Ie,
perform a query, grab a value, and then jump to another location in the dialplan
and do another query based on that original value. I'm having problems with the
result and fetchid's and I'm not sure if it's even
:[EMAIL PROTECTED]
Sent: Wednesday, January 11, 2006 11:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Nested MySQL Commands
On 11/01/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Is it possible to have nested MySQL queries in extensions.conf?
Ie
We are trying to implement findme/followme.
The dialplan first queries the dialled number in the database and determines if
it is OnNet(IP) or OffNet(PSTN). If it is OnNet, it then queries a
findme/followme table, which looks like this:
mysql select * from ast_findme;
, Douglas Garstang [EMAIL PROTECTED] wrote:
Peter,
Too slow! We're going to potentially be doing several MySQL lookups for
routing even the most basic of calls, and if every one of those queries has
to make a call out to an AGI script, it would become a performance problem.
I mean, an AGI to do
Daragon [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 11, 2006 2:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Nested MySQL Commands
Douglas Garstang wrote:
Peter, I assume you mean something like this in extensions.conf:
exten = _X.,1,AGI
Users Mailing List - Non-Commercial Discussion
Cc: Douglas Garstang
Subject: Re: [Asterisk-Users] Nested MySQL Commands
Douglas Garstang wrote:
jd but. but Asterisk still fires up a process
each time you make an AGI call in the dialplan. You could
still have 120 of these lightweight
We have no control over this. Asterisk is the one that starts the new process
upon a call to AGI in extensions.conf...
-Original Message-
From: Mike Fedyk [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 11, 2006 3:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
So I really wish there was some way to measure how well the worst case scenario
would perform. This would be 120 simultaneous calls (don't know how many per
second) on a Dual 3.8Ghz Dell PowerEdge 1850 with 2GB RAM. Asterisk would call
an AGI script, written in perl, to route all calls. The
-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Nested MySQL Commands
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
Peter,
Too slow! We're going to potentially be doing several MySQL lookups for
routing even the
most basic of calls
Well, is Perl DBI thread-safe for a start???
-Original Message-
From: Leo Ann Boon [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 11, 2006 5:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Nested MySQL Commands
Douglas Garstang wrote
-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] Re: Nested MySQL Commands
Douglas Garstang wrote:
I don't get the whole concept of FastAGI. It's nothing special.
Asterisk just opens a connection to a TCP port instead of executing a binary
I also don't believe perl DBI is thread safe
-Original Message-
From: Douglas Garstang
Sent: Wed 1/11/2006 9:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: RE: [Asterisk-Users] Re: Nested MySQL
- 716.907.4054
email - [EMAIL PROTECTED]
AIM - b2Cory
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Try running ngrep or (t)ethereal on your NTP server and see if you are even
getting requests for the time via (S)NTP.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Sun 1/8/2006 1:52 PM
To: asterisk-users@lists.digium.com
I'm not so sure about that. I tried putting a Monitor() command right before
the Dial in extensions.conf that AgentCallbacklogin() uses to call an agent
back. I get a very small file of recorded audio before recording stops. I
assume the Queue application doesn't like Monitor() being called on
]
Sent: Sun 1/8/2006 3:50 PM
To: asterisk-users@lists.digium.com
Cc:
Subject: Re: [Asterisk-Users] Asterisk Jobs
On Sun, 2006-01-08 at 13:58 +0400, Jean-Michel Hiver wrote:
Douglas Garstang a écrit :
Actually, I've found
on website. go in trade shows. Demo and
make $
Steve kalcevich
Douglas Garstang wrote:
I'm curious why the number of jobs out there requiring Asterisk seems
to be pretty low. After looking around dice, monster, careerbuilder etc, I
that would be most popular
make marketing material, dump on website. go in trade shows. Demo and
make $
Steve kalcevich
Douglas Garstang wrote:
I'm curious why the number of jobs out there requiring Asterisk seems
to be pretty low
configs that would be most
popular
make marketing material, dump on website. go in trade shows.
Demo
and make $
Steve kalcevich
Douglas Garstang wrote:
I'm curious why the number
I'm curious why the number of jobs out there requiring Asterisk seems to be
pretty low. After looking around dice, monster, careerbuilder etc, I was
surprised to find no more than 3-4 employment opportunities with Asterisk
throughout the US.
Is it really that low? There seems to be a job of
:
Subject: Re: [Asterisk-Users] Asterisk Jobs
Douglas Garstang wrote:
I'm curious why the number of jobs out there requiring Asterisk seems
to be pretty low. After looking around dice, monster, careerbuilder etc, I was
surprised to find no more than 3-4
I have a question on this. It isn't readily obvious to me, upon issueing a 'sip
show channels' command which call legs are related to which call.
For example:
*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last
Message
192.168.10.121 a00090601
On Demand-monitoring? If your referring to monitoring specific agents calls,
I'm still trying to work out how to do that. You can either monitor all calls
for a queue, or all calls for all agents, but not all calls for a specific
agent. I tried to use the Monitor() command on it's own to start
I recollect that there used to be a fixed, finite limit to the number of call
groups that could exist. Does anyone know if that limitation still exists in
1.2.1, or maybe where I could look in the code to find out if it's a fixed
length array or not? Thanks.
Doug.
Is this what you're looking for?
On 1/5/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I recollect that there used to be a fixed, finite limit to the number of call
groups that could exist. Does anyone know if that limitation still exists in
1.2.1, or maybe where I could look in the code to find out
I'd like to have Asterisk log useful messages during operation.
Is there any way in extensions.conf that I can manually log messages to a file,
say via syslog()? The console output is ugly, with all the extra Executing
NoOp(SIP/pstn.voip.com-08a28bd0, crud at the front of each line. I'm not
Well,
I want the output that the NoOp's generate. I want to be able to manually log
lines to a file through some mechanism. I just wish I could do it without all
the extra NoOp stuff at the front.
I just
tried using:
mylogfile = verbose
in
logger.conf but all I got was the
Not
everyone is a C programmer extraordinairre.
-Original Message-From: Alyed Tzompa
[mailto:[EMAIL PROTECTED]Sent: Thursday, January 05, 2006
11:59 AMTo: Douglas Garstang;
asterisk-users@lists.digium.comSubject: RE: [Asterisk-Users]
Asterisk DebuggingThen stop
I'm trying to
record calls for SPECFIC agents, which queues.conf and agents.conf don't seem to
support. Someone suggested I just put a monitor() command before the Dial() so
that when the Queue dials the agent, it will start
recording.
exten =
a00090101,1,Monitor(wav||m)
exten =
are pretty easy and will cleanup your file for
you.On Jan 5, 2006, at 1:07 PM, Douglas Garstang wrote:
Not everyone is a C programmer extraordinairre. -Original
Message- From: Alyed Tzompa
[mailto:[EMAIL PROTECTED] Sent: Thursday, January 05, 2006
11:59 AM To: Douglas Garstang
The dispatcher module in OpenSER can load balance calls based on a hash of the
SIP call-id. Supposedly the latest version even supports failover. O, fancy.
Doug.
-Original Message-
From: tijmen van den brink [mailto:[EMAIL PROTECTED]
Sent: Wed 1/4/2006
I've asked this question several times before and was always told it wasn't
possible. However, after reading a thread posted to the list today, I'm not so
sure my question was understood.
So, here I go again...
Is it possible to have multiple Asterisk systems share a common realtime
Haven't seen a post
to this list since last night. Don't know if there'sa problem or
not.
I'm trying to record
calls for SPECFIC agents, which queues.conf and agents.conf don't seem to
support. Someone suggested I just put a monitor() command before the Dial() so
that when the Queue dials
Just a curiosity really. Anyone know how I can do this?
exten = page,1,SetVar(_ALERT_INFO=ring-answer)
exten = page,2,Page(SIP/a00090101SIP/a00090301)
exten = page,3,Playback(tt-weasels)
ie Play back the sound file after the phones receiving the page have answered?
I know page is really
Reposting. I forgot to change the subject. Oops.
Just a curiosity really. Anyone know how I can do this?
exten = page,1,SetVar(_ALERT_INFO=ring-answer)
exten = page,2,Page(SIP/a00090101SIP/a00090301)
exten = page,3,Playback(tt-weasels)
ie Play back the sound file after the phones receiving the
Is it possible to record calls for specific ACD Agents?
From looking at queues.conf and agents.conf, it appears that all calls for a
specific queue can be record, or all calls for all agents can be recorded.
I'd like to be able to specify that calls for a _specific_ agent are recorded.
Case in
for Specific ACD Agents
On 12/30/05, Douglas Garstang [EMAIL PROTECTED] wrote:
Is it possible to record calls for specific ACD Agents?
From looking at queues.conf and agents.conf, it appears that all calls for a
specific queue can be record, or all calls for all agents can be recorded.
I'd like
I remember reading that there was a limitation on the number of call groups
that could exist of around 32. Anyone know if that limitation still exists in
1.2.1?
Thanks,
Doug.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users
The
word from Kevin Fleming and Digium is that the use of realtime to support
multiple Asterisk boxes sharing sip is not supported or even known to work at
this point.
-Original Message-From: Asterisk
[mailto:[EMAIL PROTECTED]Sent: Thursday, December 29, 2005 12:14
PMTo:
The
'status' is only as goodas the frequency of the qualify periodand
you can say hello to a LOT of SIP OPTIONS messages being sent from Asterisk to
each phone.
-Original Message-From: Adrian Carter
[mailto:[EMAIL PROTECTED]Sent: Wednesday, December 28, 2005 8:36
AMTo:
You can also construct a packet manually with sipsak and send that directly to
the phone.
-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 28, 2005 3:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Add it
to/etc/ld.so.conf
-Original Message-From: Kanishka Somaratne
[mailto:[EMAIL PROTECTED]Sent: Monday, December 26, 2005 9:51
PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] LD_LIBRARY_PATH
HiI set the LD_LIBRARY_PATH and when i reboot i have to set it
It seems that Asterisk gives priority to extensions in the extensions.conf file
over what's access in the db via the switch statement. For example, if you have
an entry in extensions.conf and realtime for the same extension, Asterisk won't
look in the db.
Anyone know if there's a way to
Weschke [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 27, 2005 10:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime Static/Dynamic Preference
On 12/27/05, Douglas Garstang [EMAIL PROTECTED] wrote:
It seems that Asterisk gives priority
Subject: Re: [Asterisk-Users] Realtime Static/Dynamic Preference
Douglas Garstang wrote:
It seems that Asterisk gives priority to extensions in the extensions.conf
file over what's access in the db via the switch statement. For example, if
you have an entry in extensions.conf and realtime
-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 27, 2005 10:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime Static/Dynamic Preference
Douglas Garstang wrote:
It seems that Asterisk gives
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