Google search reveals a fairly dated reference to the same carrier switch
tag message being delivered in a Skype for Business forum thread.
https://social.microsoft.com/Forums/en-US/e25b3198-b5a0-4a43-9328-4a1aff5f6ed0/1800-number-dialing-issue?forum=communicationsservertelephony
On Thu, Apr 20,
On Fri, Mar 5, 2010 at 10:33 AM, Cory Andrews ipcbc...@gmail.com wrote:
Is there a way to strip the normal features out of FollowMe (call
acceptance, etc), and just set followme up to to blind transfer any call to
an extension's associated cell number if it is not answered on the extension
On Wed, Oct 28, 2009 at 2:09 PM, Robert Grignon rgrig...@fleetone.com wrote:
This has been a rollercoaster ride
Building a new gateway (Asterisk 1.6.1 / Sangoma A108D 3.5.8 drivers)
Where I stand right now, I have a PRI on the gateway and circuit is working
I can make calls through the
JD wrote:
I've got a challenge (or clarification request if I am mistaken) for the
group.
I have a non-profit customer on asterisk 1.4 that has multiple
volunteers that work from home. The volunteers are willing to take calls
to help out the organization.
So, a formal queue is out. They
nik600 wrote:
Hi to all.
What can i do if a customer needs to log in the CDR all the dialpan
actions related to a call?
I mean, not only the lastapp e the lastdata but all the dialpan actions!
I know that the actual CDR system store one record for each call (and
for billing purposes this
Dean Collins wrote:
http://www.pcworld.com/article/160653/skype_gives_away_highquality_audio_codec.html?tk=rss_news
any thoughts?
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
mailto:d...@cognation.net+1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
OCG Technical Support wrote:
Perhaps if he threw in a paperclip and some tictacs people would respond...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: March 3, 2009 7:37 PM
To:
Vikas wrote:
I have been using the inbound 800 services from vitelity. Slowly the
usage has been rising and in the month of Jan the bill was for $650. I
am currently on a 1.9 cents a minute plan. Am I paying too much ?
Some suggestions my team generated to reduce the toll free incoming
call
Alex Balashov wrote:
RE Kushner List Account wrote:
The question is, what are you actually paying for as a customer? To
discriminate against bits just because they actually use what they are
paying for is beyond me.
At least a bandwidth cap is easier to understand. You get what you
Ade Vickers wrote:
-Original Message-
Hi Folks,
I'm using the rawplayer program to provide my
music-on-hold, and it works very well, with one small
drawback: every time I reset Asterisk, for any reason, the
MoH resets to the beginning of the list. Since MoH isn't used
that
Jeremy Mann wrote:
Any advise on troubleshooting this:
Oct 29 02:25:58 nurscarepbx kernel: wanpipe2: OOF alarm is OFF
Oct 29 02:25:58 nurscarepbx kernel: wanpipe2: RED alarm is OFF
Oct 29 02:26:05 nurscarepbx kernel: wanpipe2: RAI alarm is OFF
Oct 29 02:26:05 nurscarepbx kernel:
Tilghman Lesher wrote:
On Wednesday 08 October 2008 13:22:25 Rob Hillis wrote:
Wesley Haut wrote:
Yell at me if you will, but I hate func_realtime - it's not very
usable nor is it change-friendly (update your database and your
dialplan completely breaks).
I agree
Dean Collins wrote:
Who is Chris Langford in Huntsville Alabama and is he seeking Digium’s
permission in order to report the asterisk mailing lists out onto the
internet
http://asteriskbizrss.blogspot.com/
http://www.blogger.com/profile/04174728129647374395
What can be done to stop
Mark Phillips wrote:
Hi all,
When one uses the follow-me logic to forward calls to lots of phone
devices do subsequent calls get routed to the VM (or whatever the 10x
is)?
In other words, if I want my office, house and cell phones to ring
whenever a call comes in and I answer it on my
Igor Hernandez wrote:
I was thinking the same thing I believe Tzafrir just alluded to. If the
passwords are encrypted in the DB with a public key then...asterisk
needs to have the private key stored somewhere to be able to decrypt the
values to authenticate the user. In this way there is
Alex Balashov wrote:
randulo wrote:
On Sat, Jul 19, 2008 at 7:09 PM, Alex Balashov
[EMAIL PROTECTED] wrote:
I've asked a number of others I know in real life who got the beach
balls and all are reported as being fully functional.
So this is not a case for the bug tracker?
Sherwood McGowan wrote:
Mark Hamilton wrote:
Hi,
Yesterday I made a change in queues.conf and so tried doing a reload
app_queue.so in the CLI. (Using 1.4.18). It didn’t seem to do
anything, infact all action on CLI stopped.
Then, I did a reload. Same thing.
After that there was no
BJ Weschke wrote:
Sherwood McGowan wrote:
Mark Hamilton wrote:
Hi,
Yesterday I made a change in queues.conf and so tried doing a reload
app_queue.so in the CLI. (Using 1.4.18). It didn’t seem to do
anything, infact all action on CLI stopped.
Then, I did a reload. Same thing.
After
Steve Totaro wrote:
On Fri, Apr 18, 2008 at 11:09 AM, John Signorello [EMAIL PROTECTED] wrote:
excuse me...
But did you not just post
[asterisk-users] OT Nice IBM 1U Server Gets Along w/Old and New
Digium Boards Cheap X305 $199
Did you not provide a link to a COMMERICAL entity?
equis software wrote:
Hi, I need to make Click-to-Call web application to connect with an
asterisk server.
I´m using Java
What solution recommend me?
I did a spiel on this at Astricon last year. The slide deck is
somewhere around for those interested, but now we also have some code to
With regard to (1), yes, very good point there and certainly reason
enough to leave it alone. I had completely forgotten about a use case
like that.
With regard to (2), I'm pretty sure there's been work done in the
recent past to make chan_local more state aware so that this might not
be
Rilawich Ango wrote:
Thanks. I have checked that the queue.conf. I keep the default
setting as autofill=yes in my tests. That's mean even autofill=yes,
the 1st caller will still stick the whole queue.
asterisk version : 1.4.18
--queue.conf--
; AutoFill Behavior
;The old/current
I'll give you an A+ for originality after I get done laughing and then
we'll still ask you to take this off list. :-)
BerkHolz, Steven wrote:
Asterisk work does not pay all of my bills, so I have joined up with a
company that has a very good payment plan.
I have recently become a Mona Vie
Add an answer() and a playback of 1 second of silence or something else
to make sure the RTP is nailed up. AMD can/will hang if it has no media
to analyze.
Carlos Chavez wrote:
We have an Asterisk server with a small outgoing call center. We use
AMD and it usually works very well on
Mike Coakley wrote:
I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed
the WIKI page on setting it up but I can't seem to get it to work.
Here is my Asterisk version:
pbx1*CLI core show version
Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on
Kevin Kiely wrote:
Great suggestion, thanks. The boot failed with the mac-phone.cfg removed. I
re-touched the file and followed your suggestion.
Any way of removing the call forwarding feature via the xml configs?
Kevin Kiely wrote:
I have a remote user on a Polycom IP Phone who has
Kevin Kiely wrote:
I have a remote user on a Polycom IP Phone who has set call forwarding
by accident and is away from the phone. Does anyone know of a way to
remotely un-forward the phone? I tried to reboot the phone but that
didn’t work and removing the mac-phone.cfg caused problems
Rajkumar S wrote:
Hi,
I have callcenter running with v 1.2 with AgentCallbackLogin and now
trying to move to 1.4 using the example doc,
doc/queues-with-callback-members.txt. From what I understand the basic
idea in the example is to
1. Authenticate a caller with VMAuthenticate
2. Get his
MatsK wrote:
Olivier wrote:
Hi,
I'm wondering whether or not it is achievable to build a web based
click2dial application that could automatically detect that a user is
connected from office or home.
Another option is to directly ask user or let them change default option
but having
I've just spent the last two hours Googling and searching the Wiki. I'm
trying to find if there are any listings of codes for the Avaya Definity
G3R, to allow for an Asterisk system to turn on/off a phones MWI that is
attached to a G3. We are looking to use an Asterisk system as a voice
Jerry Geis wrote:
Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a
64 bit 4200+ box
would there be any noticable lag or delay to bring each one of them into
a PAGE mode. so one speaker can talk out on all 230 SIP clients for a
message.
Would this work?
Thanks,
Michael J. Liberatore wrote:
Would this be normal? Could this be a problem with the line?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Balashov
Sent: Friday, November 23, 2007 8:20 PM
To: Asterisk Users Mailing List - Non-Commercial
Kyriakos wrote:
Guys can someone answer how the ACD works when it needs to decide which call
to take next from queues with equal weights? Does it take the call with the
longest period of watiting or does it work randomly?
Whichever thread from the queue that does its processing first is
Kyriakos wrote:
It would be nice to add an option of choosing to answer the call with the
longest waiting time, or answer randomly, or round robin, etc...
Agreed, but, understand that each queue defined in app_queue is separate. The
way the weights work is only by instructing a thread
Gary wrote:
I used ChanSpy( ) recorded some test conversations. It has .raw
extension.
What kind of audio file is this? How can I play it?
Gary
That's SLINEAR. I know CoolEdit, now Adobe Audition can play them. Not sure
about Audacity. I've never tried it with that.
--
Bird's The
Yes. That's supposed to to be the timeout value. In the case where it's
0 are you seeing a call reject or something else?
asterisk wrote:
In my queue log I see that on the RINGNOANSWER Event I get different
content. Some events soe the ring timeout (15000). Other events show
0. Other
Peder @ NetworkOblivion wrote:
Is there any way to see the called number when a call gets redirected to
the 'a' extension from voicemail? Say x123 calls x456 and it rolls to
voicemail. x123 hits * and gets dumped into the 'a' extension in the
original context. I need some logic in 'a' to
Dan Casey wrote:
does anyone know how to route a call coming in with ANI*DNIS*
Oct 31 12:47:41 VERBOSE[30056] logger.c: -- Executing
Set(Zap/49-1, DID=1231234*4812*) in new stack
I tried making a route for _.*4812* but that matched everything rather
then just the dnis i wanted.. any
Joe Acquisto wrote:
My Polycom phones are displaying time, off by one hour. Seems they are on
the old DST rules. How do I fix this?
joe a.
If you've got the files centrally managed, you can update the correct
tags in sip.cfg to correct the situation.
These are the correct settings
Turbo Fredriksson wrote:
Quoting Joe Acquisto [EMAIL PROTECTED]:
My thanks to all. Problem resolved with the assistance.
Would be nice if you posted HOW it was fixed to... I have this exact
same problem at home, but the work phones displays time correctly...
If you've got the
Hmm.. I think it should be cleaning it up post-call already. If not,
please open a bug on Mantis as that sounds like a bug.
On 10/19/07, Anthony Messina [EMAIL PROTECTED] wrote:
Is there a variable for the filename that is created by the FollowMe
application when a is specified as an option
On 9/21/07, Gregory Boehnlein [EMAIL PROTECTED] wrote:
Hello,
At one of our locations, we have started to see Polycom 501s
(running 1.6.7 firmware) randomly reboot. We have taken packet traces of the
phones to determine if there is something odd in the Layer 2 or 3 of the
network
On 9/4/07, Larry Costigan [EMAIL PROTECTED] wrote:
I've got an Asterisk switch that is going to run an IVR menu with a database
interface that will be doing lookups based on the user entered data and then
reading back strings with the appropriate data integrated into the text. I
have found
On 8/31/07, Joe Acquisto [EMAIL PROTECTED] wrote:
Any great disadvantage to using polycom 300/500/600 vs the 301/501/601?
I recall reading in the release notes of the latest release of the
firmware (2.2+) that I believe they've finally stopped supporting the
earlier models so it looks like you
On 8/29/07, James FitzGibbon [EMAIL PROTECTED] wrote:
Does anyone know what can cause queue members to go into a status of
Unknown?
pbxtel-01*CLI queue show
cshas 2 calls (max unlimited) in 'rrmemory' strategy (24s holdtime),
W:0, C:447, A:20, SL: 91.7% within 60s
Members:
On 8/17/07, Andrew Ruthven [EMAIL PROTECTED] wrote:
Hi,
With more recent version of v1.2 and with v1.4 are things like the
AstManProxy still recommended if you want to have a bunch of
applications talking directly to Asterisk?
If you're looking to have a number of clients monitor events,
On 6/27/07, Jason Martin [EMAIL PROTECTED] wrote:
Hello,
We use MS Access 2000 (I know, we're migrating away from it) as an application
to automatically dial phone numbers. The old phone system we have allowed the
call representative using the application to take their phone off hook, push
a
On 6/19/07, Lucian Romi [EMAIL PROTECTED] wrote:
In this scenario, how to make asterisk send the invite to
SIP/[EMAIL PROTECTED]:5064
instead of
Local/[EMAIL PROTECTED]
Thanks
Asterisk isn't a SIP proxy. As such, you need to use some workarounds
to make what you want to do work. One way
On 6/6/07, Elmar Haneke [EMAIL PROTECTED] wrote:
Hi,
On asterisk 1.4.4 I have an strange effect on agents answering queue calls:
If an agent does set current call on hold the phone
immediately gets connected to the next incoming call.
What might cause this effect?
How can it
On 5/24/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Thu, 24 May 2007, William Moore wrote:
On 5/24/07, Jeremy Mann [EMAIL PROTECTED] wrote:
Can an asterisk box equipped with a Digium T1 card handle Integrated T1
circuits? I have a T1 with 768k data and the remaining channels voice, can
On 5/16/07, TienSen Chong [EMAIL PROTECTED] wrote:
Hi all,
I am seeing a strange problem with Asterisk queue. I am not sure if it's my
configuration which is wrong or there's something with Asterisk.
I am using Asterisk 1.4.2 and i have a queue with one MGCP member. When i
tried to call the
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
On 4/9/07, ahester [EMAIL PROTECTED] wrote:
Hi all,
I am having the following trouble with recording calls:
When calls come into the support line did number, the call starts to
record on the first queue, but appears to hang up when the call actually
connects to the engineer (ie I see got hangup
On 4/9/07, Damon Estep [EMAIL PROTECTED] wrote:
There are 6 different ${QUEUESTATUS} variable values defined in asterisk
1.2, I am attempting to make sure I have a full understanting of when they
would be set;
If someone could correct errors with these definitions ot would be
appreciated;
On 3/31/07, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Peder @ NetworkOblivion wrote:
I also had a question about acking a call. It appears that acking a
call is under agents.conf. I want to specify members as SIP/1234, etc,
rather than having users login all the time. I don't want to have
On 3/29/07, Jordan Novak [EMAIL PROTECTED] wrote:
What is the most stable version supporting queue priority. I have had many
crashes, I am using 1.2.11 and have set the weight in queues.conf. is there
a better way or a patch. I can't seem to find much. Any suggestions?
Do you have a bug open
On 3/19/07, equis software [EMAIL PROTECTED] wrote:
Asterisk 1.4
I have strategy= leastrecent and autofill = yes
I have 2 agents, one is answering a call and the other is free and have
some calls waiting in the queue.
Only when the first agent hangup the second agent receive the first call
You can use a patch cable, yes. The T1 will look to use pins 1,2,4
and 5 while Ethernet will typically use 1,2,3 and 6 provided you're
not using POE or something simliar that requires additional pins.
On 3/18/07, Jeronimo Romero [EMAIL PROTECTED] wrote:
Is there any technical difference between
Yes. At that point, you're looking for a T1 cross-over.
The pinout is as follows:
1
4
RX/Ring/- --TX/Ring/-
2
5
RX/Tip/+ --TX/Tip/+
4
1
TX/Ring/- --RX/Ring/-
5
2
TX/Tip/+ --RX/Tip/+
3
3
Correct. Because Ethernet cross-over cables are crossing over 1,2,3
and 6; no 1,2,4 and 5.
On 3/18/07, Jeronimo Romero [EMAIL PROTECTED] wrote:
So a regular cross over cable wouldn't work?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
If you make a SIP device a queue member, that member will be rung so
long as the device state of the SIP interface shows as not in use.
With regard to voicemail, are you trying to get a queue call answered
by voicemail or is that not your intent?
On 3/17/07, Steve Kennedy [EMAIL PROTECTED]
On 3/16/07, Kevin Kiely [EMAIL PROTECTED] wrote:
I am having an issue with the follow me application in 1.4
The application description (below) indicates that if the specified
followmeid profile doesn't exist in followme.conf, execution will be
returned to the dialplan and call execution will
On 3/16/07, Ritesh Agrawal [EMAIL PROTECTED] wrote:
Hi Folks,
I want to setup a follow me routine so that asterisk can call me on the
multiple numbers.
I tried some of the samples at voip-info but there is a problem with those
examples.
I dont have coverage in my home area and my cell phone
All -
Next step here would probably be to open a bug on bugs.digium.com
with a full VERBOSE/DEBUG log along with associated config files so we
can troubleshoot this and fix it if there's a problem.
Thanks.
On 3/9/07, Drew Gibson [EMAIL PROTECTED] wrote:
Trevor G. Hammonds wrote:
From: Drew
if this
canreinvite=no; Leave this alone for now; see archives for details
nat=1
qualify=yes
Subscribecontext=sip
notifyringing=yes
call-limit=300
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Wednesday, March 07, 2007 10
agents configured [3 online , 13 offline]
If you tell me how to do a full DEBUG/VERBOSE I will be happy to send you one.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: Thursday, March 08, 2007 7:03 AM
To: Asterisk Users Mailing List
There's a lot more than just app_chanspy.c changes required to get
the full functionality backported to 1.2.
On 3/8/07, Wai Wu [EMAIL PROTECTED] wrote:
You must be talking about Chanspy. It is included in 1.4. Has anyone tried to
compiled for 1.2x?
-Original Message-
From: [EMAIL
I don't think this is a bug.
From UPGRADE.txt:
* Queues depend on the channel driver reporting the proper state
for each member of the queue. To get proper signalling on
queue members that use the SIP channel driver, you need to
enable a call limit (could be set to a high value so it
is not
On 3/2/07, Mike Lynchfield [EMAIL PROTECTED] wrote:
Please note that we are available to fix the current REMOTE crash that
affects Asterisk/openpbx/trixbox and crashes these systems via a malformed
packet
please contacts use if you need a hand to patch your systems.
And you'll do it for
On 2/2/07, Thomas Winter [EMAIL PROTECTED] wrote:
Hi,
I have an queue stored in relatime and defined members called through
LOCAL/
I found out that if I call the members through the LOCAL think the queue
statistics is not updated.
Any idea, or isnt possible to call members with LOCAL
On 1/15/07, Douglas Garstang [EMAIL PROTECTED] wrote:
-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Monday, January 15, 2007 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue cmd option 'i'
On 1/15/07
On 1/15/07, James Fromm [EMAIL PROTECTED] wrote:
Using Asterisk 1.4, on the console 'show application queue' mentions an
option 'i' that should ignore call forward requests from queue members
and do nothing when they are requested. Does this work?
My assumption is that the member whose next
On 1/5/07, Lenz [EMAIL PROTECTED] wrote:
I think we are going to do it if we get big problems with those many
queues. From what I'm seeing, the biggest problems seem to be related to
agents, so maybe we can have a try at using straight terminals instead of
agents.
l.
Being somewhat familiar
On 1/4/07, Douglas Garstang [EMAIL PROTECTED] wrote:
Richard,
We have underscores all over the place in our config files, including others in
queues.conf. I don't think that's the murder weapon.
I think, in general, queues are one of Asterisks biggest features, and also one
of it's shakiest.
On 12/1/06, Tomer Horn [EMAIL PROTECTED] wrote:
Michel R Vaillancourt wrote:
Tomer Horn wrote:
Hello list,
I am curious here if anybody here got an experience connecting Avaya
to Asterisk using H323 / T1. I am completely lack of experience with
Avaya and I wanna know if anybody here has
On 11/30/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Thursday 23 November 2006 11:44, Heidi Mendoza wrote:
We're looking at using 4 or 8 port T1 cards with echo cancellation and are
evaluating brands to go with. We know that Sangoma has excellent solutions
especially when it comes to
On 11/16/06, Tim Uckun [EMAIL PROTECTED] wrote:
98% of the people here don't use Trixbox.
I don't think this is something with trixbox. I asked the person
having the same problem as me if he was using trixbox to see if that
would narrow down the realm of the problem.
Anyway the error message
The functionality you're describing isn't currently available in
app_queue, but it probably could be done without too much trouble.
What does everyone else think about this kind of functionality? Is it
useful? Not useful? Thoughts on ringing going on for maybe a longer
period of time when the
On 10/25/06, Christopher Aloi [EMAIL PROTECTED] wrote:
Hello List,
Question: Has anyone been able to create multiple queue_log files in
/var/log/asterisk for multiple queues?
We are designing a multi-tenant system and separating the log files
would be useful, instead of dropping all queue
On 10/25/06, Kelvin Williams [EMAIL PROTECTED] wrote:
I have been forced to introduce a Cisco 7971G-GE into my network, because
it has a pretty screen. I have wasted nearly three days fighting with the
thing based upon the information on voip-info.org and a few other forums.
Asterisk is
On 10/20/06, Dean Collins [EMAIL PROTECTED] wrote:
Hi Jesse,
Do you know if latest bootrom (3.2.2) and firmware (2.0.1) loads up onto
Polycom IP500's?
Or are they only for the later models?
Do you know if you can still use TFTP for these software updates?
They are compatible with
On 10/20/06, Dean Collins [EMAIL PROTECTED] wrote:
Sweet thanks for that, is there any reason not to go to version 2.0.1
now?
I know people were concerned initially because you cant go back but is
there a reason to go back if I have a few Polycom IP 500's?
We've got clients running 501's on
On 10/13/06, Matt [EMAIL PROTECTED] wrote:
Hi,
Does anyone know what is going on with voipsupply? My sales guy
hasn't been online in several days, their 800 number is fasy busy, as
are their direct lines. And the canadian store website is down. What
the heck is going on?
There was a pretty
On 10/11/06, Dinesh Nair [EMAIL PROTECTED] wrote:
On 10/11/06 21:15 Joseph said the following:
I quits on my as well, when I try to make a second call.
There is a bug report on it:
http://bugs.digium.com/view.php?id=7972
this seems like a configuration error within FreePBX and isnt really
On 10/5/06, Todd Houle Asterisk [EMAIL PROTECTED] wrote:
Hi Guys-
While I played a little with Asterisk a year or so ago, I'm getting ready to
roll out a project now that I think is perfect for it. My friend with with
a commercial solution he has been very unhappy with and is thinking of
On 10/4/06, asterisk-user [EMAIL PROTECTED] wrote:
Hello,
Can someone help me with this please?
Attached is the log file.
thank you
Original Message
Subject:[Fwd: asterisk-users Digest, Vol 26, Issue 166]
Date: Fri, 29 Sep 2006 10:31:21 -0400
From: asterisk-user
On 9/28/06, Douglas Garstang [EMAIL PROTECTED] wrote:
All,
I've recently been told that the AgentCallBacklogin() application is buggy, and
I should not use it. Apparently I should use AddQueueMember() instead. I see
though that AddQueueMember() does not take the location to call back as an
On 9/28/06, asterisk-user [EMAIL PROTECTED] wrote:
Hello,
I have a problem with asterisk and trying to see if someone can help me
fix the issue...
Problem:
I couldn't join ATT's Tele Conference bridge directly without their
customer service interaction.
Instead of getting the automated prompts
On 9/25/06, Michelle Dupuis [EMAIL PROTECTED] wrote:
I have a asterisk box with some queues for a call center and need help on
two points:
1. I have a scenario where if a queue has no agents logged in, an inbound
call should immediately failover to the failover destination for that queue.
On 9/21/06, Denis Galvão [EMAIL PROTECTED] wrote:
bweschke, is there any news about using astdb to store the numbers to
be dialed?
This is related to this note on bug http://bugs.digium.com/
bug_view_advanced_page.php?bug_id=5574:
(0035684)
shmaltz - reporter
11-02-05 15:01
Also thinking
On 9/21/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Sep 21, 2006 at 08:41:37AM -0700, Elpidio Ramos wrote:
Ok, after requesting information to digium (no answer yet) and being informed
that asterisk-dev is *NOT* a support hot line, I am trying in this list to see if
someone has
On 9/22/06, C F [EMAIL PROTECTED] wrote:
BJ, I believe that asteiskdb is before realtime. It does not give the
same functionality, since asterisk apps can only update asteriskdb
thru the DP, and built in commands.
There was some discussion around this feature in app_followme in the
IRC chat
On 9/21/06, Denis Galvão [EMAIL PROTECTED] wrote:
bweschke, is there any news about using astdb to store the numbers to
be dialed?
This is related to this note on bug http://bugs.digium.com/
bug_view_advanced_page.php?bug_id=5574:
(0035684)
shmaltz - reporter
11-02-05 15:01
Also thinking
On 9/20/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi,
I have a problem with Asterisk AGI command.
I wrote a script which launches a shell command.
If I launch a normal command for example like ll /tmp/tmp.txt, the
AGI command launches the shell commands and then exits.
The problem is
On 9/20/06, nik600 [EMAIL PROTECTED] wrote:
hi
is there any software usable to simulate work on an asterisk server?
I'm interested in it to evaluate the level of currently calls that a
server can support
For SIP, see http://sipp.sourceforge.net/
--
Bird's The Word Technologies, Inc.
On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote:
Doug Lytle wrote:
Steve Totaro wrote:
I am trying to connect a Definity G3 to an asterisk system. I had it
working OK with the exception of the caller ID on the Definity
handsets just
He wants to know if your Definity is an S, SI or an R?
On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote:
BJ Weschke wrote:
On 9/13/06, Steve Totaro [EMAIL PROTECTED] wrote:
Doug Lytle wrote:
Steve Totaro wrote:
I am trying to connect a Definity G3 to an asterisk system. I had it
working OK with the exception of the caller ID on the Definity
On 9/13/06, Forum [EMAIL PROTECTED] wrote:
Unfortunately they pointed me back to Polycom and I have not yet heard back
from them.
Can somebody post a link to download sip2.0.1?
Your reseller should not be pointing you back to Polycom if they are
a certified reseller. The download is
On 9/12/06, Antoine Megalla [EMAIL PROTECTED] wrote:
Hi,
I have a client who wants a call center with 16 analog
FXO modules.
I offered him a solution with a 1U or 2U rack and
Digium TDM2400 card.
I know that there mother board compatability issue
with the Digium cards, so
can anyone suggest a
On 9/8/06, gc [EMAIL PROTECTED] 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]' (musiconhold is
1 - 100 of 514 matches
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