Alex Balashov a écrit :
Syed Nasruddin wrote:
1. 10 Call Center Agents.
2. All the calls coming in will ALWAYS be routed to specific 5 agents,
firstly.
4. IF ALL the first 5 agents are busy then ONLY then the call will be
routed to next 5 Agents.
Set up two queues. Call
On Thu, Jul 31, 2008 at 05:28:42PM -0700, Stephen Cattaneo wrote:
Both are sitting behind a Linksys IP PBX (SPA9000). On the Linksys IP
PBX I have set the outside number 5000 to connect to 3001. 3002 does
not have a similar external mapping (this would defeat the purpose of
the test I am
As for me i mostly saw spellings mistakes, but that's me :)
Grygoriy Dobrovolskyy a écrit :
i saw that billing iface somewhere else, maybe i am wrong...
2008/7/30 Mindaugas Kezys [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hello,
Based on our own and our clients' experience we
Syed Nasruddin a écrit :
Hi,
I have Asterisk 1.4.18 and I have been running call center queues on
it. Today it suddenly stopped adding inbound calls to queues. I am
facing with following error: _app_queue.c:3939 queue_exec: unable to
join queue “myqueue”_
In extension file:
On Tue, Jul 29, 2008 at 01:48:26PM +1000, Lee, John (Sydney) wrote:
On the box, first of all, I just installed Zaptel 1.4.10.1.
[..]
BUG: soft lockup - CPU#2 stuck for 16s! [ztcfg:4681]
Hi, just for (all of) you to know this is a known bug of zaptel
1.4.11, the
Al Baker a écrit :
Quote
Yet amazingly (if this is, indeed, a source of amazement for you), CCM
and other Cisco software can be just as buggy as anything OSS, if not
worse.
This is simply NOT TRUE and shows a complete lack of understanding of modern
software development.
CISCO software is
voip crazy a écrit :
Hello all,
A client of us, is thinking to migrate their actual PBX to a Cisco
CallManager. We want to sell him an asterisk box to complement the
Cisco PBX.
I think to use asterisk as a Voicemail server (Replazing the Cisco Unity)
Has asterisk all the functionalities to
[EMAIL PROTECTED] a écrit :
Call me crazy, but why are you so keen on selling them an Asterisk box
when you don't even know if its capable of doing what you want to sell
it for?
I won't, i had the same felling ...
thats kinda scray actually.
Yep
I would say you have two choices for that:
opt 1, let the carrier provider do the ring
and then answer, using Wait() or WaitForRing()
opt 2, do it yourself using PlayTones() or Progess()
broadband Voice a écrit :
Finally did it but only one more problem, I want it to ring
On Sat, Jun 21, 2008 at 05:30:51AM -0700, Vieri wrote:
Hi,
I'm having trouble connecting two Asterisk boxes via a IAX2 friend trunk.
iax2 show peers on both boxes seem to show that all's fine (Status OK on
qualify=yes peer).
voip1 is an Asterisk 1.2.27 production server.
voip2 is an
Is it possible on a TE220p to deactivate the hardware echo canceler at
will ? (With a function in the dialpan for example)
example for fax SDA ,beeing able to shutdown the echo canceler could
give better results don't you think ?
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On Tue, Jun 17, 2008 at 10:56:06AM -0500, Kevin P. Fleming wrote:
Benoit Plessis wrote:
Is it possible on a TE220p to deactivate the hardware echo canceler at
will ? (With a function in the dialpan for example)
example for fax SDA ,beeing able to shutdown the echo canceler could
give
Hi,
I'm having trouble with a TE220p PRI card and (outbond) caller
identification.
Previously with usecallingpres=no everything was Ok, one small
difference between the
BRI (B410p) was that the callerid needed to be stripped from one number.
But then came the need to make hidden calls, and
Benoit Plessis a écrit :
Hi,
I'm having trouble with a TE220p PRI card and (outbond) caller
identification.
Previously with usecallingpres=no everything was Ok, one small
difference between the
BRI (B410p) was that the callerid needed to be stripped from one number.
But then came the need
Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization :)
We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent calls.
My problem is how to combine multiple accounts and fail back to PSTN
lines
Gordon Henderson a écrit :
On Thu, 5 Jun 2008, benoit plessis wrote:
Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization :)
We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent
Brent Davidson a écrit :
...I wonder why more vendors haven't adopted IAX yet?
Well, even ZoIPer (ex IdeFisk) team, still recommend using SIP over IAX
as SIP is more mature and reliable in asterisk and zoiper,
--
Benoit
begin:vcard
fn:Benoit Plessis
n:Plessis;Benoit
email;internet:[EMAIL
Benoit Plessis a écrit :
Gordon Henderson a écrit :
On Thu, 5 Jun 2008, benoit plessis wrote:
Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization :)
We are actually testing a SIP provider, which has an interessting
limitation: each account support
You might be looking for that instead http://sip.free.fr/index.html.en
Andrea Cristofanini a écrit :
Hi
I just saw this now !
does the microphone and speaker works ?
Can you use it like softphone for recive calls ?
Regards Andrea
C F ha scritto:
() +
Playback(silence/1) to get around this kind
of thing. Since it's most probably problem while decoding first bunch of
audio sample, using Wait()
won't help.
--
Benoit Plessis +33 6 77 42 78 32
[EMAIL PROTECTED] +33 4 67 28 06 96
On Wed, May 14, 2008 at 01:27:38AM +0800, Rilawich Ango wrote:
I have a queue with the following setting.
total queue member =30, autofill=1, timeout=25, monitor_format=wav49
asterisk 1.4.18
In busy hour, the loading of CPU reaches over 300%. At that moment,
all members are occupied and many
look through the changelog / bugs tracker to see if your
problem has already been reported ?
--
Benoit Plessis
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--
Benoit Plessis +33 6 77 42 78 32
[EMAIL PROTECTED] +33 4 67 28 06 96
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+Queue+with+limited+calls+per+IAX+agent
Or if you want your queue agent to be joinable with internal calls while
in communication with a queue
member.
--
Benoit Plessis +33 6 77 42 78 32
[EMAIL PROTECTED] +33 4 67 28 06 96
Hi,
I'm using a Queue in asterisk with IAX2 peers and my agents are also
doing outbond calls.
Actually we are using a GROUP() function like that to prevent users from
beeing Dialed while
in communication:
exten = s,1,gotoif($[${GROUP_COUNT([EMAIL PROTECTED])}=0]?:busy)
exten = s,n,Set([EMAIL
Russell Bryant a écrit :
Benoit Plessis wrote:
So i'm wondering if someone already as made a dialplan function that
could toggle the 'Use' flag of
an agent ? or if this kind of function would be integrated into the core
if i build it ?
This is a slightly different approach
Benoit Plessis a écrit :
Russell Bryant a écrit :
Alternatively, if you would like to control the usability of an agent
through
the dialplan, then you could use the DEVICE_STATE() function to
create a custom
device state. Then, you could list your custom device as what
app_queue
, and returns if it
doesn't find one:
Why not modifying func_devstate instead to support Custom/ ?
btw why is it limited to some channel name ? it would be nice
to use the Agent/xxx channel directly
--
Benoit Plessis+33 4 67 36 42 59
[EMAIL PROTECTED
Russell Bryant a écrit :
I don't see why this wouldn't help. You just list the IAX2 peer as the device
Asterisk uses to determine the state of the agent.
Well i've read elsewhere that only SIP peers did support the use flag ?
--
Benoit Plessis +33 6 77 42
Tzafrir Cohen a écrit :
On Tue, May 06, 2008 at 09:42:17PM +0200, Benoit Plessis wrote:
Tzafrir Cohen a écrit :
On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote:
Here it is, but since the AsteriskNow release has stripped the binary
i fear it won't
Philipp Kempgen a écrit :
Benoit Plessis wrote:
Well i tried a debian/lenny with an mISDN patched for 2.6.24
Are those patches available somewhere? Pointers?
Regards,
Philipp Kempgen
It's a patch i got from the gentoo portage site, should be made of some
mISDN commit
that asterisk fall into the KISS rule, it's not
a reason to let it grow.
--
Benoit Plessis +33 6 77 42 78 32
[EMAIL PROTECTED] +33 4 67 28 06 96
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lordfuknowsyou a écrit :
Vinícius Fontes wrote:
I use 1.4.18 with no problems. We have quite a few users(125 total
between branches), but the call volume at the most has been around 15
active calls at a time.
Any IAX2 phone or mostly SIP ?
Do you use Call Queues ?
We have less user
Steve Totaro a écrit :
On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis [EMAIL PROTECTED] wrote:
Any IAX2 phone or mostly SIP ?
Do you use Call Queues ?
We have less user than that, less concurrent call but quite a few
crash/deadlock
Try SIP only if you can and report back. I think
Tilghman Lesher a écrit :
On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote:
lordfuknowsyou a écrit :
Vinícius Fontes wrote:
I use 1.4.18 with no problems. We have quite a few users(125 total
between branches), but the call volume at the most has been around 15
active calls
Tilghman Lesher a écrit :
On Tuesday 06 May 2008 09:30:50 Benoit Plessis wrote:
Tilghman Lesher a écrit :
On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote:
lordfuknowsyou a écrit :
Vinícius Fontes wrote:
I use 1.4.18 with no problems. We have quite a few
Steve Totaro a écrit :
On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis [EMAIL PROTECTED] wrote:
lordfuknowsyou a écrit :
Vinícius Fontes wrote:
I use 1.4.18 with no problems. We have quite a few users(125 total
between branches), but the call volume at the most has
Tzafrir Cohen a écrit :
On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote:
Here it is, but since the AsteriskNow release has stripped the binary
i fear it won't be of much use:
Is there any -debug package for asterisknow's asterisk package?
On RedHat
Hi,
I'm able to send an HTML Frame to an IAX2 phone like Zoiper, Using
Dial(IAX2/...,,,http://site/uri;)
but now i need to do this
Inbound Call == Zap == Asterisk-1 [main diaplan, build the needed
Url ] == IAX2 == Asterisk/2 == IAX2 == Phone
By using a tcp trace i can see that the HTML
lokotes2 a écrit :
Hi,
I'm using Asterisk 1.4.17 and 1.4.19 versions, some time ago I've
noticed that cli command 'core show channels' does not show all data.
It returns only header or one line of data.
After that, auto completition of commands (hitting TAB) freezes cli...
Does anybody has
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