Fine, did you read the question well and understand about what I am asking?
I know well what Verbose do and what Goto do, and my question is not related to
what they are doing because I used Goto 100 times or more. I have been working
on Asterisk more than 5 years and installed alot of sites.
its name.
Just google it or lookup the latest modules available.
Regards,
Sammy
On Fri, Jul 6, 2012 at 3:20 AM, bilal ghayyad bilmar...@yahoo.com
wrote:
Hi All;
If I set a context other than the default context, then
I do not see a generation for a configuration
Dear Warren;
I did not understand the example below well.
What the Verbose will do? It will write in the CDR or the database? Really this
did not understand.
Also did not understand this lineL same = n,Goto(${EXTEN},from-internal,1) How
it will work? Can u plz explain?
Regards
Bilal
Thanks Tim.
One of the problem that I am facing is the complicated generated configuration
for the FreePBX, is it the same thing in the Elastix?
To understand this complicated generated commands, is there a documentation to
explain this for FreePBX or Elastix?
One of my friend told me that
Hello;
Is it possible if I have already asterisk installed on Fedora machine to
install the GUI asterisk now without doing a fresh installation using the
Asterisk Now CD?
Which version of the GUI that should be selected to work with the asterisk
version? For example, if I have asterisk 1.8
OK, is there asterisk-gui that differs that freepbx? Or Freepbx is a GUI for
asterisk?
In other words, if I have asterisk and I need to add for it a GUI, is there
asterisk-gui which is differs than freepbx or it is the same?
Regards
Bilal
-
Hello;
Is it possible if I have
Hi All;
Based on what I have to use Trixbox or FreePBX?
Can someone advise?
Regards
Bilal
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-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of bilal ghayyad
Sent: Thursday, July 05, 2012 6:26 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] FreePBX: How to hangup if the
caller did
Dears;
Thanks for all the replies and help.
First of all, I am not looking to have the custom context only for outbound, I
need this also to separate the extensions into partitions, so I can have same
extensions in different contexts, also extensions in context A can not call
extensions in
Hi All;
If I set a context other than the default context, then I do not see a
generation for a configuration in the extensions_additional.conf for this
context, but always the generation for the configuration is for the default
context (from-internal).
Normally, I have to put some Phones in
Dears;
In FreePBX, when I select voicemail for the extension, and if the caller sent
for the voicemail, and he leaved (or did not leave) a voice message, and did
not press #, so the channel will stay open and this is not good specially if
the call was coming from outside via the analoge lines
Dear;
What is the setting to be done on freepbx to let the voicemail go for hangup
after while (or after leaving the message) even if the caller did not dial #.
It is very important for me to be sure of the hangup status.
Regards
Bilal
--
Dear Warren;
I am thinking what if I need to have a special configuration for a phone,
something other than what the GUI is generating. In other words, what if the
GUI generated a configuration that still I need something else, I feel I have
to be able to put these configurations that I would
Hi All;
Using the FreePBX, after I added the extension from the GUI, I discover that it
is automatically added in the extensions_additional.conf in the context
[ext-local] and [from-did-direct-ivr]
How I can change these context name? I need to determine this. How?
Regards
Bilal
--
Dears;
One of the problems I faced with Polycom is the voice volume and ring volume,
it is low.
When it rings, even if it is maximum volume, still it is weak.
When I talk and I set the volume to the maximum, I still feel the voice volume
is low and would if to increase it.
The volume is
Hi All;
Any one used Digium IP Phones D40?
I need to know if they are stable with good voice quality? Comparing to Polycom
330, which is better? Let us talk frankly although I know that we have to
support Digium.
Regards
Bilal
--
Dears;
I need to order Digium card and not able to know which one is the best quality?
Is it that of AEX with the end E or EF or P or B?
I saw those card that its slot is small (I think those that end by EF), are
they the best card?
Really I am caring to have a card that has echo cancelation
at this
time. Other web gui's might work, but I am not
familiar with them.
FreePBX's sentiment on the subject is shared here:
http://www.freepbx.org/trac/wiki/AsteriskRealtime
-John
On 05/24/2012 05:46 PM, bilal ghayyad wrote:
Thanks for all for the help and kindly reply.
One last point
Thanks for all for the help and kindly reply.
One last point that will help me alot:
I am thinking to have 4 Servers running Asterisk and 2 Servers to be for
database. The load to be distributed on the 4 Asterisk Servers with ability to
be redundant (using any redundancy technique). The 4
Hi All;
I need to use Asterisk for 20 000 users, so which asterisk version to be used?
Is there asterisk version that supports 20,000 users on one hardware machine?
Can I use one strong hardware server i7 with 64 GB RAM and fast hard desk to
handle 20 000 users, and concurrent calls 2000? Or I
, for a client who wants to achieve numbers you have
just said. Let's
see how it works in few months.
Leandro
2012/5/23 bilal ghayyad bilmar...@yahoo.com
Hi All;
I need to use Asterisk for 20 000 users, so which
asterisk version to be
used? Is there asterisk version that supports
Hi All;
I did not install AsteriskNow, but I am thinking to install it.
If I installed it, I can see the call details records for the extensions? I can
request the CDR to be between specific dates and to specific extension?
Regards
Bilal
--
Dears;
How I can increase the voice volume in the analogue line (at dahdi port)
without getting a problems in the entered digits (when using Background
function)?
I got to know previously that it is possible to increase the gain at hardware
and not software, this is better to avoid getting a
Dear;
I am talking about something else.
When I said increasing the volume from the hardware, I was mean something else.
Before when we were using Zaptel, we were able to do this (increasing the
volume from the hardware) in the /etc/modprobe.conf but currently we are using
dahdi and I do not
I am sure there should be another place .. if I increased it from
chan_dahdi.conf, the voice quality is bad and the calls will disconnecting
while we are talking ..
Increasing voice volume from chan_dahdi means increasing it at software level,
I am sure there is a place to increase it at
, bilal ghayyad
wrote:
What is happening with me that when I used fedora core
16, I compiled
and installed dahdi 2.6 and then compiled and installed
asterisk 1.4
and it did not create chan_dahdi. I tried to select it
by running make
menuselect and I discover that it is not possible
Hi All;
First of all, I am trying to install vicidial and actually vicidial requires
asterisk 1.4 and can not work with asterisk 1.8, in addition there is a special
version of asterisk 1.4 that is required for vicidial which is
asterisk-1.4.39.1-vici.
The problem that it look like there is a
Hi All;
I installed Asterisk 1.8.11-cert1, and it look like the default is ulaw for the
sound files. How I can fix this?
Athough file beep.gsm is existed under path (/var/lib/asterisk/sounds/en), but
when I used the Record function, it gave me the following (so I am sure there
is something
Dear;
The output of the ./configure that is related to dahdi is:
checking for DAHDI_RESET_COUNTERS in dahdi/user.h... yes
checking dahdi/tonezone.h usability... yes
checking dahdi/tonezone.h presence... yes
checking for dahdi/tonezone.h... yes
And the dependecies of the chan_dahdi as I saw in
Dear;
Well, I did make menuselect and I really found the XXX and did not get the
ability to select the channel. So what could be the reason?
From the other side, I find the following when I type for lspci
02:01.0 Ethernet controller: Digium, Inc. Wildcard TDM410 4-port analog card
(rev 11)
Dears;
I see this at the /var/log/asterisk/messages:
[Apr 20 01:49:48] ERROR[1657] codec_dahdi.c: Failed to open
/dev/dahdi/transcode: No such file or directory
Again, I am installing asterisk and dahdi at Ubuntu (uname -a
Linux House 3.0.0-17-server #30-Ubuntu SMP Thu Mar 8 22:15:30 UTC 2012
Dear Warren;
Yes, first thing I do is the make all and make install for dahdi, then I do
./configure and make and make install for asterisk. But I do not find the
chan_dahdi under the /usr/lib/asterisk/modules. WHY?
If I used asterisk 1.8, then I do not have any problem.
What I am missing?
Dear Warren;
Yes I am compiling and installing dahdi first and then I start by asterisk
1.4.39 but I do not find chan_dahdi under /usr/lib/asterisk/modules, but if I
used asterisk 1.8, it is working fine.
From the other side: I tried asterisk 1.4.44 and same thing (I am not able to
see the
Hi All;
Is it normal if I used asterisk 1.4 and dahdi, then I will not find chan_dahdi
under /usr/lib/asterisk/modules? And I will not be able to type dahdi commands
(dahdi restart for example) in the asterisk CLI?
Actually what I found only the following:
app_dahdibarge.so app_dahdiras.so
Hi All;
It look like DAHDI versions that before 2.5 have a problem to be compiled on
ubuntu, can someone check below and advise me how to fix this?
The output of the uname -a is:
Linux House 3.0.0-17-server #30-Ubuntu SMP Thu Mar 8 22:15:30 UTC 2012 x86_64
x86_64 x86_64 GNU/Linux
I am trying
Dears;
In asterisk 1.8, it is not more possible to use DeadAGI?
Also, I found the below commands in the a2billing and I would to ask why it set
the sequence 1 for the Hangup()? Maybe because it is related to the NoOp? How?
[a2billing-callingcard]
exten = _X.,1,NoOp(A2Billing Start)
exten =
Dears;
I am looking to get a telephony card that has GSM slots (ability to place my
GSM card into it) in addition to analoge FXS and FXO.
There is a card that I found it but really I do not know how much it is
reliable: http://www.atcom.cn/AX2G4A.html
Did anyone tried atcom?
Is there a
Hi All;
I need to use IVR functionalities in Asterisk, I am asking if there is a ready
made thing for some IVR functionalities (like saying the numbers, the date the
currency ... etc)?
Regards
Bilal
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Hi All;
Is it possible to restrict the authentication to be based on the username and
password and to be allowed for IPs within the LAN (for example, 192.168.10.x)?
I do not need it to be based on the IP only and do not need it to be based on
the username and password only, but I need it to
Hi All;
Is there a collaboration contact center (hope to be open source) Integrated
with Asterisk (hope with vicidial), so the agent will be able to receive chat
or emails sessions and deal with the customer. If the agent in a call with the
customer, then he will not get chat session. Is there
Hi All;
If we need the admin to have the ability to hear what the agent is talking
without the agent or the customer feel, how this can be done? Is it using
MeetMe or something else? How? When the admin start hearing, there will be a
peep (because I do not need this, it should be silent
Hi All;
Is there an admin tool (web based) to check the voicemail and manipulate it
(delete all the voicemail under extension, showing how many voicemails for the
extension, ... etc)?
Can AsteriskNow do this? Or any other recommended tool that it is very good in
this?
Regards
Bilal
--
Hi All;
I know that I can use the AGI to call (run) a script (php or python or any
other kind of scripts), but the question is:
If I have information that I need to build a decision in the extensions.conf
based on it, and these informations can be obtained using this script, so how I
will
Hi All;
Really I need to know why when using the h in the exten =, then we use
DeaAGI with it?
I am using vicidial and I see this line alot, so I need to know how it work
(when it will be executed):
exten =
Hi All;
If I need to build IVR using Asterisk (so I will read and write to database),
until now from my reading, I can understand that the best way is to use AGI to
call external script like php which will manipulate every thing, correct?
Well, the returned values from this script that I can
Hi All;
Which asterisk version that support the ability to have the configuration in
the database?
Regards
Bilal
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Hi All;
I heard from some friends that there are a dsl router that has Linux OS and it
has asterisk on it, so the ip phone can register on this router, also if the
router has FXS or FXO ports then it can be used to place calls through them.
Is it really? Where I can these routers? Did anyone
Dear Binni;
My asterisk version is:
Connected to Asterisk 1.4.39.1-vici RPM by dem...@goautodial.com
So it is only by 1.4.19?
By the way, the version I am using has been installed using goautodial.
Regards
Bilal
Hi, I've played around with using a database
Hi All;
Because vicidial is working with asterisk 1.4, so I would like to know in case
of using asterisk 1.4, can I have the configuration to be in the database? As I
know that version 1.8 is supporting configuration to be via Database instead of
conf files, but what about 1.4?
From the other
Hi All;
Is there a telephony card that contains analoge ports and E1s at the same time?
Regards
Bilal
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Dear All;
I am afraid from IRQ misses: 1
The ISDN E1 was working fine on the machine, the electrical disconnected and
then the Red Allarm. I checked the dahdi and I found that I have to reinstall
dahdi again and I did. But still not becoming UP.
The output of the cat /proc/dahdi/1 is
Dear All;
Anyone used GoAutoDialer or ViciDial or ViciDial group and can advise if they
are good and stable with asterisk. Appreciate if any advise.
I am using asterisk and need to use outbound dialer and really caring for the
below 3 things to be possible to be applied (all togethor to be
Hi All;
I am looking for a good Outbound Dialer and to be practical with possibility to
do modification on it, the outbound dialer should send the calls to the agent
when the agent is logged in as long the agent is belong to the queue (or let us
say the skill group of this campaign).
Any one
Hi All;
I need to use a gateway that converts from SIM to E1 to I can send and receive
calls via the GSM, so did any one use a good gateway for this and reliable and
stable and costly effective, so he can advise us to use it?
Also, it will be a separate product if we need also to use it for
Dear List;
I have internal Menu that I use it when doing outside call (it ask some
questions, and then it ask to dial the number that need to call it), I can
access this menu by dialing 9, but after the Background stay play the wave
file, then whatever I dial (to select) and whatever the
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Tuesday, November 29, 2011, 7:10 AM
Hello Bilal,
You can check on http://www.kannel.org/
Cheers,
Sendil
On Tue, Nov 29, 2011 at 4:17 PM, bilal ghayyad bilmar...@yahoo.com
wrote:
Hi All
Hi All;
When using the IP Phone and will be prompted to enter a digits (for example,
using the Background function), so when dialing the digits, I do not see these
digits on the LCD of the Phone, how can I make it appear?
Actually I need it because I have a Background funtion that ask the
Hi All;
I am facing a problem that when the call send to the queue, it is sending to
the agent while the agent already has a call!! And it keeps sending the calls
until the agent hangup the call the first call, then the calls go to the next
agent and so on. And while the agent is having the
Hi All;
When the call coming via the E1 dahdi and I handle the call (as first step) by
exten = 5631040,1,Goto(OrangeCMG,s,1) it is not working and the call will be
disconnected instead of queued.
But, when I handle the call (as first step) by playing any sound file and then
send for the
Dear Warren;
Thanks a lot for the reply and kindly help.
I ran the below commands and now I have the database asteriskcdrdb and the
table cdr, also I did the configuration in the cdr_mysql.conf but until now I
do not see entries in the database (still the table is free).
The asterisk version
Dear;
It look like AGI is not suiting this, maybe I need AMI?
Because no need to do a call to collect the data, I need to collect data
without doing a call ... so what do u suggest?
What I need, is to know how many concurrent calls in the queue, and how many
waiting and how many agents in the
Dear;
In case I need to retreive the real time data (for example, how many calls
currently in the queue, and how many calls currently waiting in the queue, how
many agents currently are logged in ... etc).
How to get this?
Is it using the AGI? From where I can get information about this?
Dear Raj;
Thanks a lot.
Actually I need to do a dash board for reporting, so I beleive the only way is
to use the AGI, correct? But where I can find documents or link that can help
me to do this?
About ur sentence:
some ready-made packages (both FOSS and proprietary) that will display this
Thanks a lot.
Really I am trying to know how to do AGI to get the information from asterisk
(for example, how to talk with asterisk to know the concurrent calls, or the
number of agents in the queue, ... etc)? Where I can find this?
Regards
Bilal
Actually I need to do a
Dear wcselby;
Thanks a lot for your reply.
For below script, I have some questions if you can help me:
1) I am looking to have reports for the call center, so I need to determine how
many calls in the queue, and how many agents logged and when the agent logged
in and when logged out ... etc.
Hi All;
As I am using the ${CALLERID(num)} to be part of the filename that I am
recording it, I am facing the following problem:
If the incoming call (via PSTN) reached for an extension (which is the
reception), and then the extension transferred the call to the proper person,
and we need to
Hi All;
Is it possible to be part of the voicemail to play a wave message as following:
The person you are calling is not available, press 0 if you need to call the
operator or 1 to leave voice message?
I know that I can do this as part of the extensions.conf, but I am looking if
it possible
Hi All;
Is it possible to store a variable at context and using it in another context
or in the MACRO? For example, how I can store the ${CALLERID(num)} in a
variable and use it in another context or in a MACRO?
Regards
Bilal
--
at 1:59 PM, bilal ghayyad bilmar...@yahoo.com
wrote:
Dear;
By the way, the asterisk version is: 1.8.4.2
Yes I tried Set(CALLERID(num)=5631040) as shown
in the below dialing, and
no success. Also I tried Set(CALLERID(num)=1040) and I
tried
Set(CALLERID(num)=065631040) as the city
Dear All;
make progdocs, we use it to create documentation, correct?
Well, how I can use this documentation if I need to search for a topic or
settings? Any example?
Regards
Bilal
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Bialal,
what hw you are using and what is the h/w files. ?
On Tue, Oct 18, 2011 at 5:23 PM, bilal ghayyad bilmar...@yahoo.com
wrote:
Dears;
We contacted the Telecom provider and they confirmed
multiple times that
the DID service is enabled, but again still the caller
id
need to be updated? Also, should I use Set(CALLERID(num)=1040) or
Set(CALLERID(num)=1040)?
What other factors I am missing? What I have to check?
Regards
Bilal
---
On Tuesday 18 October 2011, bilal ghayyad wrote:
We contacted the Telecom provider and they confirmed
multiple times
Hi All;
I was having asterisk 1.8.5 and I installed 1.8.7, I copied the
extensions.conf, sip.conf, voicemail.conf that I was using them in 1.8.5 to the
new asterisk 1.8.7.
Every thing fine, but now the voicemail is not working at all !! Even does not
display on the consol any thing that it
Dear;
Are the below are used any more in asterisk 1.8 version, and for what we use
them:
;dial trunk
exten = _X.,1,Dial(SIP/trunk/${EXTEN})
;exten h must be in same context!
exten = h,1,noop(extended CDR)
exten = h,n,set(CDR(hangupcause)=${HANGUPCAUSE}) ; hangupcause
exten =
Dear all;
I have three agents and I need the calls to be always send for agent1 and if he
is busy then to be sent for agent2 and if he is busy then to be sent for agent3
and if all busy then to stay in the waiting until one of those three agents is
available. How?
Do I have to set the strategy
Dear Tariq;
About elastix.org, this can be use with Asterisk or it is coming as a complete
IP Telephony, Call Center, IVR and Reporting?
Because, I do not need to install another IP Telephony on the server which
already has asterisk which is an IP Telephony, this will cause a problem in the
Dear Dale;
Penalty is priority, or what exactly?
Also, how I can set the penalty of the member?
Regards
Bilal
--
Dear all;
I have three agents and I need the calls to be always
send for agent1
and if he is busy then to be sent for
Hi All;
In the zaptel, we were increasing the gain of the voice volume at the hardware
level from the /etc/zaptel and /etc/modprob.conf files, but now we are using
DAHDI, so where to do the same thing?
I am looking actually to increase the volume at hardware level and not software
to avoid
the same result that is means that you don't
have permission from
your provider
kind regards
2011/9/26 C F shma...@gmail.com
Confirm with your provider that allow you to set
caller id on outbound.
On Sun, Sep 25, 2011 at 1:59 PM, bilal ghayyad bilmar...@yahoo.com
wrote
the file.
On Sat, Sep 24, 2011 at 2:08 AM, bilal ghayyad bilmar...@yahoo.com
wrote:
Hi All;
I noticed in the queues.conf the configuration for
recording the calls in
the queuing, and regarding to the filename (or any
other parameter), it is
written that I can determine
Hi All;
I do not know if the SEP_Mac_address.cnf.xml of the cisco file is also has
effecting on the DID and Caller ID to appear at the destination, because I
found the following:
localCfwdEnabletrue/localCfwdEnable
semiAttendedTransfertrue/semiAttendedTransfer
= _90Z,105,Hangup()
---
Set(CALLERID(num)=5631040)
add this before the Dial command.
On Sat, Sep 24, 2011 at 4:03 PM, bilal ghayyad bilmar...@yahoo.com
wrote:
Hi All;
The DID range that we took from the telecom starts
from 1030 and end by 1059, now whenever we
Hi All;
The DID range that we took from the telecom starts from 1030 and end by 1059,
now whenever we place a call, the destination see the number 5631030. I gave
the phone extension 1040, and when I call, still the destination see the number
is 5631030?
Kindly find below the configuration of
Hi All;
I noticed in the queues.conf the configuration for recording the calls in the
queuing, and regarding to the filename (or any other parameter), it is written
that I can determine the filename using the command:
Set(MONITOR_FILENAME=foo)
But it should be called from the dialing plan,
Hi All;
I configured some queues, and I configured the dialed numbers for login and
logout for the agents.
Two agents are logged in, the first two calls are received at the agents and
they answered and hangup. Again, the two agents are idle and ready to receive
calls. The third and call goes
Dear Tareq;
I am not using mysql, the configuration on the text configuratoin files and the
logs are existed under the directory (/var/log/asterisk).
Well, to use mysql: then it means the configuration will be also in the
database or I can use mysql only for reporting?
What is the Flash
Hi All;
Asterisk version is: 1.8.5.0
But I see at the consol the following warning and really I did google but did
not understand if it is bug or related to settings:
[Sep 13 15:04:56] WARNING[2209]: chan_sip.c:19667 handle_response_invite: just
did sched_add waitid(3429468) for
Hi All;
Anyone advise for a free (open source) reporting to be used for asterisk call
center?
Regards
Bilal
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Hi All;
How can I get a SIP trace to troubleshoot a one way of communications? I need
to see what is happenning in the packets to know the reason of the problem.
Thanks
Regards
Bilal
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OK, I can buy echo canceller from Digium and how will be installed in the
digium card? Or it is a hardware?
Currently I am reading a message at the consol that Unable to enable the echo
canceller .. does this means that Digium card that I have is not supporting?
This is the output of the
Thanks for the help and reply.
And this can be done only by setting the callerid=5100 in the sip.conf? Or I
have to do any thing else?
Regards
Bilal
--
I need that if five IP Phones make outside calls, then
destination
should see only 56725111...
You can set the
The current dahdi version is:
PBX-FF*CLI dahdi show version
DAHDI Version: 2.4.1.2 Echo Canceller:
Well, the output of the dahdi_cfg as shown below, it declares there is invalid
argument. But, really I tried to change the configuration in the systems.conf
from fxoks=1-16 to fxsks=1-16 but did
Hi All;
The main number is 56725000 and we have DIDs from 5000 to 5999. Now, I need
that if five IP Phones make outside calls, then destination should see only
56725111 so I beleive it is related to the DID 5111 but I do not know what I
have to do a settings for this DID and where, so I can
Hi All;
To overcome the echo problem, what mainly I have to do in the configuration
other than the following line in the system.conf under dahdi directory?
echocanceller=mg2,1-16
1) How can I know if the digium card supporting echo cancellator?
2) If I am getting a message in the consol that
Hi All;
Suddenly, we restarted the Asterisk machine and the echo appeared. The lines
are analoge.
At the consol, I see this message:
[Aug 10 14:36:05] WARNING[3789]: chan_dahdi.c:4831 dahdi_enable_ec: Unable to
enable echo cancellation on channel 1 (No such device)
[Aug 10 14:36:07]
Hi All;
The asterisk version is 1.8.4.2
Why codec translation from and to gsm is not possible? I think it was possible
in previous versions.
I am missing something to have this codec translation possibility?
Please advise.
Regards
Bilal
--
Dear Paul;
I got to know the reason for the problem, it is becaused I have to use
SIP/bilal instead of SIP/599 as user configured as bilal in the sip.conf and
not 599.
But I am still having some points that are really not clear and causing a
problem:
1) If I need to login via the agent ID,
What I am looking for is:
If my IP Phone is related to a pickup group #1 and a call is ringing at pickup
group #2, so I can pickup the call that is ringing at group #2 and I do not
know its extensions?
In other words, how can I pickup any phone that is ringing without knowing its
extension,
Dear Carlos;
And when the Asterisk SCF will be available? With which version?
Regards
Bilal
-
Hi All;
Actually what I am looking into is a method to have
multiple Asterisk Boxes to be working togethor as one entity
so a distributing for the load and for the tasks can be
Really I am thinking in something and I donot know if it can be used or not:
Is it possible to use the Database (to be located in a server) so whenever the
agent is login via any server, this will be logged in the database, so all the
queues in all the boxes can check with this database to see
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