Hey guys,
Iam having a peculiar problem with my asterisk installation. The specs
are..
[EMAIL PROTECTED] ~]# asterisk -V
Asterisk 1.2.7.1
Wildcard: Digium Wildcard TE110P T1/E1
Wildcard TDM: Wildcard TDM400P REV I (4 modules) ( 2 FXO, 2 FXS)
Wildcard TDM: Wildcard TDM400P REV I (4 modules) ( 1
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I would look at ventilation if I were you. Drive failures at the rate
you are talking about can usually be traced back to thermal failures.
Just a thought
Stu
Dushyanth wrote:
Hey guys,
Iam having a peculiar problem with my asterisk
Heat = #1 cause of disk failure. If they are roasting to the touch they will
fail in 2-3 months.
- Original Message -
From: Dushyanth [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: 10/05/2006 9:44 AM
Subject: [asterisk-users] Asterisk Server : IDE HDD frequent crash
Hi guys,
i just want know how do i enable CDR in asterisk. and is it possible to get the time spent on each extension for a caller? for example
time spent in a queue
+
time spent on agent exten
+
time spent on ivr
so if its possible, how?-- RegardsRizwan HishamSoftware Engineer
On 05 Oct 2006 23:06:00 +0200, Benny Amorsen [EMAIL PROTECTED] wrote:
Actually, does anyone make an IP phone which doesn't do PoE?
It looks like the Linksys phone that resembles a traditional wall mount phone. I have seen no mention in the specs that it operates on PoE. That's a shame, because
So I patch my asterisk (version 1.2.12.1) with the patch given by Moises.
http://galileo.ivsol.net/play_dtmf-1.2.12.1.patch
Thanks Moises.
When I type in show manager command PlayDTMF it is their. With the show manager
commands it is not within the list containing all the commands.
When I
So I patch my asterisk (version 1.2.12.1) with the patch given by
Moises. http://galileo.ivsol.net/play_dtmf-1.2.12.1.patch
Thanks Moises.
When I type in show manager command PlayDTMF it is their. With the show
manager commands it is not within the list containing all the commands.
When I
Eric I've had issues with Iaxy's that sound very similar, performing a
full reset following these instructions and re-provisioning them solved
any problems.
http://www.voip-info.org/wiki/view/IAXy
Bails
Erik Anderson wrote:
Greetings - I have recently purchased 2 IAXys. The documentation
Hello
Am starting on my Asterisk journey - am getting a single span Digium
card to connect Asterisk to our Alcatel 4400 EPABX and install about 100
VoIP instruments.
The Asterisk VoIP extensions and Alcatel digital extensions have to talk
to each other.
Am I right in understanding that
IN
is there a good and free asterisk gui that is not tight to a live cd?
I like [EMAIL PROTECTED] but it looks like I need to install the livecd. I
just want to run asterisk on my debian install. Is there a way to run
[EMAIL PROTECTED] on debian? or anything similar?
thanx in advance
Pat
DF == Dave Fullerton [EMAIL PROTECTED] writes:
DF Greetings I have a couple polycom phones (501 and 601) I'm messing
DF around with and I've noticed something weird. Both phones
DF synchronize their clocks to a central NTP server here on our
DF network and both phones are 11 seconds slow. All of
I cannot find this option in the snom firmware, the only thing I found is
DTMF via SIP INFO:
This sounds nice but I guess it will break stuff if you need DTMF tones to
get through the menu of a remote PBX.
Ideally * would need to interpret the SIP INFO message from the Snom as
start
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Just thought I'd let people know that I've created a new StumbleUpon
group for Asterisk sites.
If you have a site that is related to Asterisk and is not listed, feel
free to add it.
Alternatively, if you're new to Asterisk and want to find out
On 10/6/06, K Y Iyer [EMAIL PROTECTED] wrote:
Is that broadly correct?
Yes
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Title: RE: [asterisk-users] Asterisk Configuration Complete Newbie question
Thanks very much - let me see how far I can take it now.
Best wishes
Iyer
-Original Message-
From: [EMAIL PROTECTED] on behalf of Lacy Moore - Aspendora
Sent: Fri 10/6/2006 03:37 PM
To: Asterisk Users
On Thu, Oct 05, 2006 at 04:07:14PM +0200, Michael Neuhauser wrote:
I've created and attached a one line patch (for 1.4 branch, r44464) that
should give you the info you need (sort of). But be aware that I haven't
tested it on 1.4 (only on 1.2, but things are different there). Only use
this
On Thu, Oct 05, 2006 at 07:22:16PM -0700, Mike Morris wrote:
I'm preparing for my first asterisk install, and would like to ask a
hardware question confirm my understanding of some basics:
* The Q: I'm looking for 2 FXO ports to have asterisk answer 2
incoming lines. There
Hi,
can two * boxes use the same realtime database?
I know they can in terms of connecting to the same db, but it is my
understanding that the peers are created realtime as and when it
registers, in other words even of the two boxes share the same db, the
peer will only exist on the one it
Hi all
i have Asterisk server
I have IP authentication from provider
when everi order some DID from him, he will forward to my Asterisk
where i register the DID and works fine
Now i have given access to one more office
so i want to forward some of the DID from my asterisks to other Server
how
Hi
can some one clarify
does the aterisks act like a SER
Ram
___
--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
Another way would be to set the dtmf option to speed dial and then add a
speed dial number 1: *1
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
No - at least not that I've been able to figure out. These phone's were made to be used with Cisco's Call manager software (Skinny?) and the SIP firmware doesn't seem to allow this. Softkey buttons (like hold, transfer, conference), seem to be static and you can't change them. You could always use
On 06/10/06, ram [EMAIL PROTECTED] wrote:
Hi
can some one clarify
does the aterisks act like a SER
http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+not-proxy
--
Peter Bowyer
Email: [EMAIL PROTECTED]
___
--Bandwidth and Colocation
Look long and hard before purchase of a TDM400 It doesn't work with many
motherboards, and Digium's anser is try another Motherboard
Seriously consider the Sangoma A200
5 year Warranty and works with all motherboards.
John Npvack
Mike Morris wrote:
I'm preparing for my first asterisk install,
2006/10/4, Benko [EMAIL PROTECTED]:
Hello!
How can i change the verbose logging level to a file in 1.4?
In 1.2 i was used to set the verbose level via asterisk -Rx 'set
verbose 5' but in 1.4 it is always reset to OFF again, so (nearly)
nothing is logged to /var/lib/asterisk/verbose:
seems the
Thanks. My asterisk servers are in California, USA and the service provider is SBC (ATT).
Asterisk 1.2.8zaptel 1.2.6Hardware: digium TDM2422P
I get the following error messages in /var/log/asterisk/messages:
Oct 3 00:34:18 WARNING[16716] chan_zap.c: Ignoring signalling
Oct 3 00:34:18
http://www.voipsupply.com/home.php
On 10/4/06, Devraj Mukherjee [EMAIL PROTECTED] wrote:
Nokia E series with proper firmware upgrade :)
On 10/5/06, Steve Glaus [EMAIL PROTECTED] wrote:
bilal ghayyad wrote:
Hi List;
I would like to know where I can find the IP Phones
that can be used
Grandsream IP phone Budge Tone 1001, 102
Softphone X-Lite
Ekiga (Ubuntu)
Etc
Jose Diaz
Forrest Beck wrote:
http://www.voipsupply.com/home.php
On 10/4/06, Devraj Mukherjee [EMAIL PROTECTED] wrote:
Nokia E series with proper firmware upgrade :)
On 10/5/06, Steve Glaus [EMAIL PROTECTED]
Hi Ram -
so i want to forward some of the DID from my asterisks to other Server
how can i do that, and i need to give them access to calling out also
You need to connect your asterisk machine together. The most common
ways to do this are either with IAX or SIP. To do this with IAX, you
On Thu, Oct 05, 2006 at 11:41:32PM -0700, Sam Norris wrote:
Heat = #1 cause of disk failure. If they are roasting to the touch they
will fail in 2-3 months.
One word: smartd.
I didn't know it existed, and I'm amazed I didn't. Everyone on this
list should be running smartd, and know what it's
On Fri, Oct 06, 2006 at 08:59:41AM +0200, Jan du Toit wrote:
PS: This reply will probably go under a new thread with the same
subject. I receive the digest mode of the mails on this list, and
replying to it breaks the thread. How can I avoid this in the future?
Switch out of digest mode.
I partitioned/formatted a new WD2500 with NTFS on a WinXP machine,
filled it with data (mostly 10MB FLAC and SHN soundfiles). Then
transferred it to an AAH Asterisk server box with a Digium TDM400P
(1FXO/1FXS) and an Audigy2 soundcard. I installed it as hdb, booting off
hda (no other
Hi Ed -
5. Digium TDM22 (TDM400P)
6. Analog phone plugged in port 3
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
Zaptel.conf :
loadzone=us
Todd,
Appreciate you have submitted to a non-commercial forum. One cannot but note
though that most of what you require is probably already available
off-the-shelf in commercially available packages and does not need to be
reinvented.
If you wish to know more of one such package, please
Did you set immediate=no in zapata.conf?
Francesco
Eddie Johnson Jr wrote:
Hello,
I have the following setup:
1. Ubuntu Dapper Server 6.06 plus latest patches
2. Asterisk 1.2.11
3. libpri 1.2.3
4. Zaptel 1.2.8
5. Digium TDM22 (TDM400P)
6. Analog
Also http://www.enterux.com/ in Mumbai, India - very, very helpful
people, indeed
HTH
Best wishes
Iyer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jose diaz
Sent: Friday, October 06, 2006 6:51 PM
To: Asterisk Users Mailing List - Non-Commercial
On 10/6/06, Gareth Owen [EMAIL PROTECTED] wrote:
Morten,Hmm, I haven't tried Asterisk 1.4 - I guess I should upgrade my system to see what is going on.Can you post the INVITE message that is being rejected?
This INVITE results in a 488 from the phone:
INVITE sip:[EMAIL PROTECTED] SIP/2.0Via:
Hi Vicente,
I solved my problem and now the PBX I set up can make and receive calls
without any problem using Telecom (Italy) PRI lines.
My zapata.conf is:
context = telco_zap
group = 1
immediate = no
internationalprefix = 00
language = us
nationalprefix = 0
pridialplan = unknown
Hi list,
I must be in tutorial writing mode this week, as I have prepared another
tutorial on how to avoid queue_log file rotation on AAH/TrixBox and other
systems. This is done automatically but it's quite an annoyance because it
interferes with queue_log analyzers like QueueMetrics and
I just upgraded an old Asterisk 1.0.xx to 1.2 but there are some changes
in the trunk definitions of sip.conf
All my trunks stopped working.
Is the sintax someting like this?
register=200:1000:[EMAIL PROTECTED]:5060/200
this is to user 200 (why do we need to put it 3 times???)
with password
Patrick Aljord wrote:
is there a good and free asterisk gui that is not tight to a live cd?
I like [EMAIL PROTECTED] but it looks like I need to install the livecd. I
just want to run asterisk on my debian install. Is there a way to run
[EMAIL PROTECTED] on debian? or anything similar?
You
Hi list,
I must be in tutorial writing mode this week, as I have prepared another
tutorial on how to avoid queue_log file rotation on AAH/TrixBox and other
systems. This is done automatically but it's quite an annoyance because it
interferes with queue_log analyzers like QueueMetrics and ends up
I am a little stumped on this one and it may be because my brain is
ready for the weekend. I am trying to set an extension for forwarding
all calls to voicemail. So if a user set's their phone to forward all
calls to extension 2000 it will drop the caller in the user's
voicemail box.
I
Thank you for your response. They are all connected to the LAN, and
when they, out of the blue, go dead is that they loose their dial tone and so
forth. Somethiing need to be changed in the Config, but I am affraid that
if I start making changes, I can screw things even worst.
Ed
Thanks for your response.
No, there;s no firewall and they are all correctly connected to the
LAN. They work just fine, and then, one or two days later and out of the
blue, they start having problems.
Ed
___
--Bandwidth and Colocation provided by
Nevermind. Just decided to use:
exten = _22XXX,1,Voicemail(u${EXTEN:[EMAIL PROTECTED])
On 10/6/06, Forrest Beck [EMAIL PROTECTED] wrote:
I am a little stumped on this one and it may be because my brain is
ready for the weekend. I am trying to set an extension for forwarding
all calls to
Yes, I have and I received the following:
In zapata.conf your first two channels should be fxs_ks because the first
two modules are FXO mdoules. Your last two channels should be fxo_ks because
the second two modules are FXS modules.
For the TDM400P(TDM 22) the FXS modules work with the phone.
Yes, I did. Still nothing.
Ed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Francesco
Francesconi
Sent: Friday, October 06, 2006 10:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No Dialtone
Did you
Hi Forrest -
I am trying to set an extension for forwarding
all calls to voicemail. So if a user set's their phone to forward all
calls to extension 2000 it will drop the caller in the user's
voicemail box.
exten = 2000,1,Voicemail([EMAIL PROTECTED])
this of course gives me a error that
Any more suggestions,
Call Digium. They will get you to the point where the hardware will
work. If it won't work (and there's nothing wrong with your system),
they should exchange for a unit that will work.
___
--Bandwidth and Colocation provided by
Have you ever tried allow=alawulaw in the same line? just a tip...On 10/6/06, Morten Isaksen [EMAIL PROTECTED]
wrote:
On 10/6/06, Gareth Owen [EMAIL PROTECTED]
wrote:
Morten,Hmm, I haven't tried Asterisk 1.4 - I guess I should upgrade my system to see what is going on.Can you post the INVITE
Ed,Do the phones lose their registration? If you run sip show peers when the phones are not working, do they show as being registered or not?AlexOn 10/6/06,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Thanks for your response.
No, there;s no firewall and they are all correctly connected to
Remco Barendse wrote:
I cannot find this option in the snom firmware, the only thing I found is
DTMF via SIP INFO:
This sounds nice but I guess it will break stuff if you need DTMF tones to
get through the menu of a remote PBX.
I'm pretty sure that when you AREN'T sending the DTMF inband,
That's not a forward condition. As far as I know, you can't forward calls
between Asterisk servers. A forward must complete on the Asterisk server the
original call was serviced by.
Doug.
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Friday, October 06, 2006
On Oct 6, 2006, at 12:07 AM, Christian Stredicke wrote:
Here comes the advertisement for snom phones: http://www.snom.com.
CS
And here the one for cisco phones: http://www.cisco.com/en/US/
products/sw/voicesw/products_category_buyers_guide.html#number_1
--
Michiel
I have followed this configuration to the letter but still no joy. Do I have
to load some modules at start up like gtalk.so or jabber.so? Should my user
show up as available on other peoples buddy list after asterisk starts?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi,
Im doing some research for Disk on a Module (DOM)with asterisk realtime. To have no moving parts for a special project, I know I can use 3.5 or 2.5 HDDs but DOMs sound interesting.
Does someone have working experience with this?
Basically the Asterisk Realtime will be stored in MySQL and the
Upgraded to 1.2.12.1 and the problem went away.. must have been an IAX bug.
On 10/5/06, Matt [EMAIL PROTECTED] wrote:
Interesting.. this is almost the same issue I'm having, except that I
am sip before iax.. where as this guy is iax then sip.
Yes I would be interested in testing out your product.
Does anyone have any other recommendations. A softphone would work for
me. I would like something that had a chat feature like eyebeam does.
I found another product called SNAP that will pop a web page, but it can
only pass cid info not
Thanks for the reply...zapta.comf[channels]group = 1language=encontext=incomingsignalling=fxs_ksswitchtype=nationalusecallerid=yeshidecallerid=nocallwaiting=yesmusiconhold=defaultusecallingpres=yescallwaitingcallerid=yes
I stumble on this URL that is Chat Line script written by Steven L.
Edwards called 'Match Chat' here:
http://bugs.digium.com/file_download.php?file_id=11080type=bug
But I can't seem to find any additional info on Author or Applications -
I was wondering if you might know more about either?
I'd like to know if anyone has a suggested fix for this...
You have a 'cluster' of Asterisk servers that use DUNDi etc for registration
redundancy, finding other phones etc. You have a separate Asterisk box for
voicemail. For voicemail deposit/retrieval you trunk the call over to the
voicemail
Erick Perez wrote:
Hi,
Im doing some research for Disk on a Module (DOM) with asterisk
realtime. To have no moving parts for a special project, I know I can
use 3.5 or 2.5 HDDs but DOMs sound interesting.
Does someone have working experience with this?
Basically the Asterisk Realtime will be
On Fri, Oct 06, 2006 at 02:42:41PM +0200, Benko wrote:
2006/10/4, Benko [EMAIL PROTECTED]:
Hello!
How can i change the verbose logging level to a file in 1.4?
In 1.2 i was used to set the verbose level via asterisk -Rx 'set
verbose 5' but in 1.4 it is always reset to OFF again, so (nearly)
what about the interval of the registration? is 2 minutes too often? Dovid B [EMAIL PROTECTED] wrote: Timed out from what I have seen comes from either a poor internet connection or a problem with your ITSP.- Original Message - From: stan ford To:
Jeremy McNamara wrote:
Erick Perez wrote:
Hi,
Im doing some research for Disk on a Module (DOM) with asterisk
realtime. To have no moving parts for a special project, I know I can
use 3.5 or 2.5 HDDs but DOMs sound interesting.
Does someone have working experience with this?
Basically the
For us the voicemail server doesn't have to know what phones are
registered where. We have an externnotify script that drops x number of
msgx.txt files into the respective voicemail folders after any call that
goes through the voicemail server. x in this would be the number of
messages.
Since
On 2006-10-06 06:31:48 -0700, Jay R. Ashworth [EMAIL PROTECTED] said:
On Thu, Oct 05, 2006 at 11:41:32PM -0700, Sam Norris wrote:
Heat = #1 cause of disk failure. If they are roasting to the touch they
will fail in 2-3 months.
One word: smartd.
I didn't know it existed, and I'm amazed I
Hello,
I am trying to use Asterisk pull its configuration (Sip.conf,
Extension.conf) from the Postgresql (ARA - Realtime). I am missing
documentation regarding setting this up (connectivity portion)
For example, in file extconfig.conf I need to add:
sipusers = pgsql,asterisk,account
sippeers =
Kristian Kielhofner wrote:
Erick,
Or Just use AstLinux which kind of does what Jeremy described :)
http://www.astlinux.org
P.S. - I am the creator of AstLinux
--
Kristian Kielhofner
Sorry to reply to my own post, but there seems to have been some
confusion in what I said here.
You can also change some settings in the zapta and zaptel
config.. to reduce
echo and interference on the line..
This is the most important thing here - what does your zapata.conf look
like?
zapta.comf
switchtype=national
This is not necessary in your case. It pertains to PRI lines, and
I was experimenting with FastAGI in Asterisk 1.4 and wrote some code around it.
I was using the AGISTATUS variable to determine if I had been able to connect
to the fast agi server, and act accordingly.
1.2 appears to be different. It has no such AGISTATUS variable, but more
importantly, it
There is a patch that allows a jump to N + 101.
Thanks,
Steve
-Original Message-
From: Douglas Garstang [mailto:[EMAIL PROTECTED]
Sent: Friday, October 06, 2006 4:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] AGI() in 1.2 and 1.4
I
JdT == Jan du Toit [EMAIL PROTECTED] writes:
JdT PS: This reply will probably go under a new thread with the same
JdT subject. I receive the digest mode of the mails on this list, and
JdT replying to it breaks the thread. How can I avoid this in the
JdT future? Thanks.
Switch to a newsreader
Does anyone have a way to send the DID in place of the CID number. I
want pop a web page with the DID in the URL but all the software I have
seen only supports putting the CID info in the URL. If I could swap the
two I could just use the programs as is. The two programs I have looked
at so far
Is there something wrong with the chanisavail() application in 1.2.12.1?
My dialplan has:
[syst_Route]
exten = _[*0123456789].,1,NoOp(*** Originated call ${CALLERID} - ${EXTEN})
exten = _[*0123456789].,n,NoOp(FOO1)
exten = _[*0123456789].,n,ChanIsAvail(SIP/${EXTEN})
exten =
Michael,You should be able to just do this: Set(CALLERID(num)=${DNID})... Though the VoIP-Info page is very vague about the DNID variable. You might try it out though.Best of luck!Alex
On 10/6/06, Michael Sampson [EMAIL PROTECTED] wrote:
Does anyone have a way to send the DID in place of the CID
from http://www.asteriskguru.com/tutorials/chanisavail.html
If there is no available channel the ChanIsAvail application will
continue with the execution of the extension with priority n+101
Douglas Garstang wrote:
Is there something wrong with the chanisavail() application in 1.2.12.1?
My
That's not how it appears to have worked before. Previously, I was able to call
it and then simply check the value of the ${AVAILCHAN} variable at n+1. The
docs imply that jumping to n+101 only occurs if j is supplied, and I'm not
passing a 'j'.
*CLI show application chanisavail
-= Info
James Harper schrieb:
I tried one of these and pretty much got it working under visdn. If
you
do decide to try one, make sure you get the HFC version. Earlier
ones
used another chipset and definitely weren't supported using open
sourced
drivers.
Please post back if you do get one and
The bad news is that the 1.4.1 beta firmware won't help solve your problem, the
problem is being caused by the multiple ptime directives in the INVITE
message.
According to RFC2327 ptime is a media-level description and hence applies to
all the codecs in the m=audio line, thus it is only valid
anyone have experience with IntuitiveVoice's Asterisk system?
Do you Yahoo!?
Get on board. You're invited to try the new Yahoo! Mail.___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
Hi ho,
is there anyone out here that is making use of the regcontext and
regexten settings in sip.conf? I've tried this on two Asterisk boxes
(1.2.10 and 1.2.12.1) and in both cases I don't see the Noop priority 1
being created upon SIP client registration, show dialplan xxx reveals
no
Hello Comunity,
How can I get Asterisk realtime working with Postgres? (without ODBC)?
Thanks
John
/doc/realtime.txt in Version 1.4 Beta2
Currently there are three realtime database drivers:
* ODBC: Support for UnixODBC, integrated into Asterisk
The UnixODBC subsystem supports many
Hi,
I'm a novice in asterisk.
I'm just want to know if we can develop a Call centre application on an asterisk ?
And if ok, have you some url link to help me or simple a open source application doing the job ?
Thank you a lot.
Imed
Découvrez un nouveau moyen de poser toutes vos
Anyone know why I get HTTP Connection Closed on the display of a
7960 running a SIP image?
Only seems to happen when registering against my Asterisk box from
the WAN. I have 1:1 NAT happening on my firewall. Phones function
perfectly otherwise. TFTP working fine across the firewall as
This happens if you have a logo_url configured for your phone and the
phone can't access it. I'm guessing you don't allow 80 through the
firewall to the server that's serving the image.
--
Aaron Daniel
On Fri, October 6, 2006 20:13, Robert Goodyear wrote:
Anyone know why I get HTTP Connection
Hi all, I have a client whose business is currently running on [EMAIL PROTECTED] 2.6 with Cablevision' s (CV) Optimum Voice (OV) and 3 lines. There are going to be 4 additional trunks needed and I'd like to move/migrate them off of OV, to a better more flexible/open/supportive VSP. OV does not
previuos post mangled.
Hi all,
I have a client whose business is currently running on
[EMAIL PROTECTED] 2.6 with Cablevision' s (CV) Optimum Voice (OV)
and 3 lines. There are going to be 4 additional trunks
needed and I'd like to move/migrate them off of OV, to
a better more
Yes, you can easily use asterisk for a call center, start looking here
http://www.voip-info.org/wiki/view/Asterisk+call+queues
M
Imed Imed wrote:
Hi,
I'm a novice in asterisk.
I'm just want to know if we can develop a Call centre
application on an asterisk ?
Hi,
I am wondering ifastcc has ever worked for someone because it always return 0 for answeredtime! I tracked every bit of informaion on google and wiki and finally found out that its because of asterisk returning to dial plan after executing Dial, so
astcc.agi runs through the end without
91 matches
Mail list logo
