Meyerriecks
Sent: Saturday, 31 May 2014 3:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Flashing Red Lights in TE420 (5th Gen) Card
On Fri, May 30, 2014 at 4:07 AM, Lee, John (Sydney) john@compuware.com
wrote:
Even without plugging
I have the following software installed in a Centos Box with a TE420 (5th Gen)
card.
. Centos 6.5 64-bit
. Asterisk 1.4.22
. dahdi-linux-complete-2.9.1.1+2.9.1
. libpri-1.4.14.
Even without plugging in the ISDN into span 1, all 4 spans are flashing red.
Plugging an E1 into span 1 makes no
Hi,
I have noticed it for a while but I just thought about confirming this with the
Asterisk community.
As the compilation of DAHDI will need to reference Kernel-devel, does it mean
that after DAHDI is installed, we should not yum update kernel because it will
affect the operation of DAHDI?
wanting to upgrade ...
2014-04-15 10:37, Lee, John (Sydney) skrev:
Hello,
I have been running Asterisk for the past 5+ years on RedHat and I never
upgraded it before.
All my Asterisk software is of the following release:
1) Asterisk 1.4.21.2
2) Libpri-1.4.4
3) Zaptel-1.4.11
I would like
Hello,
I have been running Asterisk for the past 5+ years on RedHat and I never
upgraded it before.
All my Asterisk software is of the following release:
1) Asterisk 1.4.21.2
2) Libpri-1.4.4
3) Zaptel-1.4.11
I would like to move the OS to CentOS and then I thought I can at the same time
ponder
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Wiater
Sent: Saturday, 26 May 2012 5:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Function not Registered??
On 5/25/2012 3:18 AM, Lee, John (Sydney) said:
-- Executing [*1223*1**1900
Hi all,
I am running the same Asterisk 1.4.21.2 with the same configuration on all the
servers in the region.
I got this function called func_devstate which I use to control the lights of
the Polycom phones.
This module works well for all the Asterisk servers except this one.
To get it
Thanks Sam, John and Justin for your wonderful advice.
Yes, it was the sip.conf parameter reinvite= which was causing the
problem.
Setting it to NO will fix it.
Thanks all in asterisk-users mailing list.
The contents of this e-mail are intended for the named addressee only. It
contains
I have been deploying Asterisk (open source PABX) in the company which I
work.
So far, all the Asterisk servers do not really talk to each other.
Recently, I am experimenting to dial from one Asterisk server to another
through the WAN and I encountered a no-audio problem although the
callee's
I was trying to do a SIP call between two Asterisk servers (1.4.21.2)
1) On the caller server, I coded the following in extensions.conf
Dial(SIP/1166:password@asterisk-callee);
2) On the callee server, I coded the following in sip.conf
[1166]
type=friend; Friends place calls
I was trying to do a SIP call between two Asterisk servers (1.4.21.2)
1) On the caller server, I coded the following in extensions.conf
Dial(SIP/1166:password@asterisk-callee);
2) On the callee server, I coded the following in sip.conf
[1166]
type=friend; Friends place calls
I was trying to do a SIP call between two Asterisk servers (1.4.21.2)
1) On the caller server, I coded the following in extensions.conf
Dial(SIP/1166:password@asterisk-callee);
2) On the callee server, I coded the following in sip.conf
[1166]
type=friend; Friends
chan_sip does not support specification of the password to be used for
authentication in the dial string itself;
your :password suffix is just being sent to the target system and it
is trying to find a matching extension in the dialplan (and failing).
Thanks Kevin. This is what I reckon from
We are setting up an office in Malaysia.
We contacted Telekom Malaysia and are surprised to be told that ISDN-30
is no longer available.
They are yet to give us information of the replacement technology.
Does anyone have any experience and information with this?
Thanks in advance.
--
???
Hello Lee,
Telekom Malaysia provide PRI lines. We've been actively using their services
for the past few years with success. Let me know if you need contacts.
Regards,
Arstan
On Thu, Jan 20, 2011 at 9:56 AM, Lee, John (Sydney) john@compuware.com
wrote:
We are setting up an office
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Use modprobe to find E1/T1 jumper
setting
onPRI card
On 09/29/2010 02:52 AM, Lee, John (Sydney) wrote:
Do you mean that if I could define 30 channels in span 1 for
example, then span 1 is set to E1
In Asterisk, the funny thing is if a certain component is not installed
properly or the config file has a typo or something, this will be shown
up as a non-existent command in Asterisk command line interface.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
:30 +1000, Lee, John (Sydney) wrote
Does anyone know if I could use modprobe command to find out rather than set
the jumper on a Digium PRI card?
I want to find out the jumper settings on the card without opening the box
which will cause down time.
Thanks.
--
Carlos Chavez
Director
Does anyone know if I could use modprobe command to find out rather than
set the jumper on a Digium PRI card?
I want to find out the jumper settings on the card without opening the
box which will cause down time.
Thanks.
--
The very obvious thing to check is the permission of the
mac-addr-directory.cfg.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hin lee
Sent: Thursday, 18 March 2010 4:56 PM
To: Asterisk Users
Yes, this is still one of the unsolved mysteries I wanted to find out
about Polycom provisioning despite using it for a few years now. I used
vsftpd and initially used boot server opt 66 and type string but could
not get it to work.
I asked our guru in DTW and he told me to use 129 and lo and
and DHCP option number, it would be a good idea all else
being
equal. Maybe 160 would give the same trouble as option 66 :-)
Thanks for your post.
-Karl
- Original Message -
From: Lee, John (Sydney) john@compuware.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
-boun...@lists.digium.com] On Behalf Of Lee,
John
(Sydney)
Sent: Thursday, February 11, 2010 1:21 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] app_dial.c: Unable to create channel of type
'Zap'(cause 34 - Circuit/channel congestion)
Just to share some experience with everyone
] app_dial.c: Unable to create channel
oftype
'Zap' (cause 34 - Circuit/channel congestion)
On Thu, Feb 11, 2010 at 06:20:54PM +1100, Lee, John (Sydney) wrote:
Just to share some experience with everyone about what happened
today to
our Asterisk 1.4 box with Digium TE412P card.
We had
Just to share some experience with everyone about what happened today to
our Asterisk 1.4 box with Digium TE412P card.
We had an unscheduled power outage which shut down the Asterisk box.
When the power went up, Asterisk came back up okay but the ports on the
card were all red. Zttool show red
.
Stuart
From: Lee, John (Sydney) john@compuware.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wed, 27 January, 2010 8:02:14
Subject: Re: [asterisk-users] Polycom phone DND state
I am using
In your dialplan, you should put in...sth like
exten = 1001,hint,Custom:virtext1001
In your script, you should put in...sth like
Set(DEVSTATE(Custom:virtext001=INUSE);
Set(DEVSTATE(Custom:virtext001=NOTINUSE);
In the phone directory.xml, define an entry with ct=1001 and turn bw on.
Reboot phone
I am using 1.4.21.2 and DND is definitely working.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Saturday, 23 January 2010 2:50 AM
To: 'Asterisk Users Mailing List - Non-Commercial
when use the VoiceMail , all the directions all english. i want to
know is there some Chinese version of sounds available now?
or should i record it myself?
http://www.voip-info.org/wiki/view/Asterisk+sound+files+international
Look under Chinese (Mandarin)
--
Bon journo Aldo.
I am having several issues with my first SP 650.
* Assembly: 2345-12600-001 Rev.G
I have deployed more than 200 IP650 with the same assembly as yours and
so far there are no problems.
The first thing I have noticed is that I was not able to upgrade the
unit's
Has anyone experienced this problem before?
I am running Asterisk 1.4.21.2
If I run:
MixMonitor(..)
Dial(SIP/...)
Both parties cannot hear each other.
As soon as I comment out MixMonitor, the audio can be heard.
I saw this issue on https://issues.asterisk.org/view.php?id=16256
It seems to
I don't think this can be done.
In your scenario, B is effectively the host and if B drops the line, both A and
C will be dropped off as well.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
1) I have not seen a blue light (usually red/yellow) before on a Digium card
and so don't really know what it means.
2) Try to see if you can see any messages coming up from the Asterisk box
itself (not thru putty or other remote console). You should see a steady
stream of error messages
the CRC4 and it worked straight away. Can you recommend a CRC
type at all or would it be best to leave it as nothing?
David of Brisbane.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Lee, John (Sydney
I found that it was a bit incomplete for China.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Wednesday, 23 September 2009 3:35 PM
To: asterisk-users@lists.digium.com
Subject: Re:
BTW, I have been using the n-way conference feature from Polycom.
By n-way, they mean only 4 parties (including the host) and the
interface is quite neat because you can manage the conference from the
display and you can mute, far-mute, hold and resume each parties.
To use this Polycom nway
The URL is a good start but for some large countries which I have worked
for, the list misses some important information like inter-city,
inter-state, inter-city mobile and local mobile and IDD.
To me, nothing can replace local intelligence.
-Original Message-
From:
A user embedded an * in a Read command and it causes my AEL script to
fail.
Does anyone know how I could code to detect it?
Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix,
I have a cron job that restarts Asterisk every night.
This is supposed to be an old Asterisk best practice for 1.2.* but I think it
does not harm.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
does Digium provide a service support for a compatibility question
about
their PRI hardware ?
Before you open a call with them, you will have to register your Digium
card by entering the serial number. The serial number is printed on a
sticker which is attached to the card. There is no way
I think you have to write your own agent login and logout so that you
will not have this problem.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shanavaz E
A
Sent: Wednesday, 2 September 2009
Just a quick guess - is it because you did not program your Polycom digit plan
properly in sip.cfg?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi
Sent: Tuesday, 1 September 2009
Please find attached my Asterisk sip.conf .
Can you please let me know what modifications are needed ?
Yes, I am referring to the Polycom sip.cfg and not the sip.cfg in
Asterisk.
Somethere down in sip.cfg, there is a line that looks like this:
digitmap dialplan.digitmap=#700| ...
I'd contact Digium - they're really good with providing support - just
add the following line and dial it:
Thanks Matt for your suggestion.
We despatched a new TE412P card to replace the existing card but the
same problem occurred. So, I think it is not the Digium card problem.
At the same
1) Can ACD (Automatic Call Distribution) service work with asterisk, and how
to set up ACD in asterisk ?
You can (and it is better to) write your own code in Asterisk.
2) How call barging can set up in asterisk ?
There is a zap barge cmd - not sure if this is what you want.
3) How call
It also means that unless your target cchannel is in gsm format
How can I check what format my channels are using?
___
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register
Is this the one you are talking about?
Do you mean that if I play MOH using any of the formats below, then
there will be no CPUs wasted for translation purposes?
*CLI core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your
I was copying tracks from CD into mp3 files so that I could use it in
Asterisk 1.4.21.2 MOH. (BTW, I have already secured proper license to
play MOH to callers.)
I used MS Media Player version 11 and rip it at 128kbps (smallest) but
whenever I listen to MOH, I saw the following message on the
Yep, agreed.
Convert the file to the native codec(s) in which it will be played.
Alex, could you please elaborate on this? I am no audio guy.
On Media player, I can rip it into mp3 or wav or windows media audio.
Which one should I use?
___
--
Does anyone know how to find the serial number of Digium card without
opening the machine?
I was trying to call for support at Digium and they asked me for the
serial number.
___
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:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie: How to find the serial number
ofDigium card?
On Sunday 16 August 2009 19:59:50 Lee, John (Sydney) wrote:
Does anyone know how to find the serial number of Digium card without
opening
I have this DELL PE2950 running Asterisk 1.4.21.2 on RHEL 5 with no
problems since Dec last year. We are using Digium TE412P to connect to
an E1 ISDN line. Since Dec last year, we did not add or delete any
software or hardware. We also did not do any yum update.
The linux kernel is
I have this DELL PE2950 running Asterisk 1.4.21.2 on RHEL 5 with no
problems since Dec last year. We are using Digium TE412P to connect to
an E1 ISDN line. Since Dec last year, we did not add or delete any
software or hardware. We also did not do any yum update.
The linux kernel is
I am running Asterisk 1.4.21.2
For reception, I defined a simple queue with one SIP phone as the only
member.
When I receive an incoming call, I test QUEUE_WAITING_COUNT to see if it
is 0.
If it is 0, then I will playback a message to tell the caller to be
patient and then do a
I saw a few posts of this problem before I could not figure out the
reason I am getting it.
I am running RHEL 5, Asterisk 1.4.21.2, zaptel 1.4.11 and libpri 1.4.4
Basically, if I dial into a queue and hang up the phone, Asterisk did
not detect the hangup request and Asterisk will only hang up
Solution: http://bugs.digium.com/view.php?id=12655nbn=10
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Lee, John (Sydney)
Sent: Wednesday, 22 April 2009 3:56 PM
To: Asterisk Users Mailing List - Non
Daily Asterisk restart
Daily log rotation
Voicemail clean up for people leaving an organization.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James
Mutuku
Sent: Wednesday, 22 April 2009 3:15 PM
Daily Asterisk restart
Do you think its mandatory in production env?
It could be a pre-1.6 advice but I still stick to it.
I did it to all my production Asterisk servers.
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Thanks guys.
It was the If vs if that was causing the problem.
This is probably due to my good coding practice of other languages in
the past :-)
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Watkins,
I just want to confirm but it seems that if-then-else is not permitted
in case structure.
It was not really documented but it seems to be the case.
Can anyone confirm?
switch(${DIALSTATUS})
{
case NOANSWER:
{
// if-then-else not permitted
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'
We then Chmodded everything under /dev/zap/ , rebooted and almost fell off
our chairs when it worked!
By right, if the problem is due to this error, you should see a permission
error message in
I did not really spend too much time on looking at call forwarding and
wonder if someone could help me.
It seems that for setting call forwarding on the Polycom phone itself,
only forward all calls will work. The other call forward function
like forward if no-answer for n rings or forward if
What is IDAP-T1? How different is it from normal T1?
Any chance I can get it to work with Digium 412P and Asterisk 1.4.* ?
If yes, what would zaptel.cof look like?
Any difference from normal T1 config?
Thanks.
___
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My working meetme.conf is like below.
[general]
[rooms]
conf = 101,,
conf = 102,,
Your original email says your meetme.conf is:
[rooms]
conf = 101;
If you don’t want to use passwords, I think it is better to use:
[general]
[rooms]
conf = 101
Hope this helps!
Of course you should be using AEL.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Alan Lord (News)
Sent: Tuesday, 10 February 2009 6:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
www.voip-info.org
[...]
So, the easiest way that people could contribute to improving Asterisk
documentation right now would appear to be by improving articles on
www.voip-info.org...
Absolutely.
What I tend to do is the make contributions to a particular page
whenever I encountered a
There's nothing special about analogue phones in China, they are fully
interchangable with analogue phones elsewhere... Perhaps you have a
configuration problem, or, hardware problem on the Rhino Channel Bank,
perhaps the ports are wired the wrong way and the phones care, perhaps
the
I've not used Rhino kit, but, that sounds like a firmware bug that
they
have a workaround for... With any luck it's very infrequent and
they'll
be releasing a fix once they've worked out the cause... Sorry I can't
help,
might be best to ask Rhino about the details of the problem...
The
I am testing analog phone and fax machine plugged into Rhino Channel
Bank which is connected to TE412P card. This site is in China.
I am running RHEL 5, Asterisk 1.4.21.2, Zaptel 1.4.11 and libpri 1.4.4
I ran into a problem which is analog phone can hear dial tone and can
make outgoing calls.
As the subject says, I need to implement on my call center the Agent
functionality, son the agents could logon and logoff to the queue
How can I do this configuration? Or where can I read some info about
it
Here is a few links I used when I developed mine.
Calling all Polycom gurus:
I am using Polycom IP601 phones with Asterisk 1.4.21.2
In all Polycom phones, I set the following in sip.cfg.
dialplan dialplan.impossibleMatchHandling=2
/dialplan
(I leave the digitmap unchanged because I thought setting
impossibleMatchHandling will ignore the
The reason is your audio file is too high quality.
Asterisk can only play back audio file of 4000Hz.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ??
Sent: Tuesday, 11 November 2008 5:35 PM
To: asterisk-users
Subject: [asterisk-users] music
I am testing Asterisk 1.4.22, zaptel 1.4.10.1 and libpri 1.4.4 on
RHEL5
on DELL PE2950 and using ISDN-10.
What device?
I am using TE412P.
No message on the console of the machine?
Yes, nothing at all.
The machine just froze and had to be rebooted.
This probably means one of two
I am testing Asterisk 1.4.22, zaptel 1.4.10.1 and libpri 1.4.4 on RHEL5
on DELL PE2950 and using ISDN-10.
I thought about cutting over to production tonight when I noticed a
serious problem.
SIP calls are fine but if I dialed to outside (Dial(Zap/g1)) a few times
or someone called in a few times,
I did not know what I did but I bumped into something in the log that
says:
[Oct 16 ...] ERROR[24536] res_config_mysql.c: MySQL RealTime: Ping
failed
(2006). Trying an explicit reconnect.
[Oct 16 ...] DEBUG[24536] res_config_mysql.c: MySQL RealTime: Server
Error
(2006): MySQL server has gone
Also i would suggest enabling full log, as it's one place you can see
everything. Then use grep to search for realtime messages. Your
logger.conf should already have commented line:
full = notice,warning,error,debug,verbose
Yes, I did that.
# tail -fn0 /var/log/asterisk/full | grep -F
Hi Atis,
queue_log = mysql,asteriskcdrdb,queue_log
that is engine,database,table
If it's wrong, you should see some warnings when asterisk is starting
up.
Thanks for the suggestion. I did not put in queue_log for table and
it has just taken the default which is queue_log.
In the console
You might want to double check the socket path. Some distributions
use
/var/run/mysqld/mysqld.sock as the socket file.
Thanks for the suggestion Tilghman.
I am using Redhat and the socket file is indeed
/var/run/mysql/mysqld.sock.
Actually, if you specify the wrong socket file, you will see
http://ftp.iq-labs.net/queue_log-
1.4/asterisk_queue_log_realtime_1.4.19.patch
This uses standardized realtime/mysql library from asterisk addons.
For it to support SQL inserts in 1.4, you would also need to apply
both patches from (1 for asterisk, another for asterisk-addons)
To: Lee, John (Sydney)
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] realtime queue_log to mySQL backport to
1.4
Hi John,
On Mon, Oct 13, 2008 at 9:51 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
http://ftp.iq-labs.net/queue_log-
1.4
(NULL), callid, queuename, agent, event);
[...]
+ }
}
-Original Message-
From: Atis Lezdins [mailto:[EMAIL PROTECTED]
Sent: Monday, 13 October 2008 8:02 PM
To: Lee, John (Sydney)
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
Sorry to post a C compile error on this mailing list but this is
Asterisk related.
Basically, I was following
http://www.plack.net/index.php/2007/01/07/asterisk_modification_for_queu
e_logging
to patch logger.c and Makefile in Asterisk 1.4.* in order to write
queue_log to mySQL database.
When I
This looks really old and weird. I could suggest using realtime
queue_log backport from 1.6 which i'm currently using.
That's good info, Atis.
I will definitely give it a go.
winmail.dat___
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Yes, unfortunately, VOIP wiki did not mention about installing
mysql-client which it should have been.
Without mysql-client, you cannot change passwords, grants, etc.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stefan Schmidt
Sent:
On Wed, 2008-09-10 at 18:10 +1000, Lee, John (Sydney) wrote:
I am struggling to find out how to code hint in AEL2.
I did hint(Custom:light1) and it keeps complaining about the :
(colon).
It works fine for SIP device like hint(SIP/439).
Anyone who has tried it before?
Yes, a while
context BLF {
hint(Sip/1000) 1000 = NoOp();
};
Works for me
Thanks Eric.
I did not experience any problem in hint with SIP. The problem is if you use
it with Custom.
winmail.dat___
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I am struggling to find out how to code hint in AEL2.
I did hint(Custom:light1) and it keeps complaining about the : (colon).
It works fine for SIP device like hint(SIP/439).
Anyone who has tried it before?
___
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: Thursday, 11 September 2008 2:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie AEL2: Syntax for Hint
On Wed, 2008-09-10 at 18:10 +1000, Lee, John (Sydney) wrote:
I am struggling to find out how to code hint in AEL2.
I did hint(Custom:light1
A cheaper alternative would be the voip wiki.
http://www.voip-info.org/tiki-
index.php?page=Asterisk%20config%20extensions.conf
Unfortunately, as advised by other asterisk users,
http://www.voip-info.org is sometimes really not that up-to-date.
However, that does not mean that we should give
I believe that this is what I need to enable more than one buddy icon?
Can you please point me in the right direction. Only the polycom
screen, I can only see 1 buddy icon despite having 2 speed dial
entries.
I have been able to successfully turned on presence (which is the term
used
It's not perfect, because it
doesn't display DND or queue login/pause status, but it's better than
nothing.
James, on a different note, is it true that at this stage, we can never
get any queue login status/light on Polycom phone?
I posted a query a few days ago but I have got 0 reply.
Any
Sorry, needed to add one more note. To clarify, my agent phones have a
speed dial assigned for their login, and another to pause/unpause. I
could then use DEVSTATE to enable or disable the light next to their
speed dial button based on their status. I can't use it to update
anything on the
I have been coding my own IVR for ACD (aka queue) using Polycom phones
using AEL2. In particular, I have coded my own AgentCallbackLogin
because a) cmd AgentCallbackLogin() is buggy and will not be supported
by dev anymore b) I can put in features like hotdesking and additional
validation like
Just out of curiosity, where do you get this AddQueueMember syntax
from?
Here:
http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.c
om
/books/9780596510480.pdf
page: 367
Oh so the VOIP Wiki is out of date!
Now, where should we go to for reliable Asterisk info then?
I need login Agent(Member) in asterisk.
use this option:
for example:
AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13)
Just out of curiosity, where do you get this AddQueueMember syntax from?
http://www.voip-info.org/wiki/view/Asterisk+cmd+AddQueueMember
Description:
I have made class for MOH uploaded a mp3 file to the folder.
Now I am using this class for music on hold during dialing.
Now when call has been established, I put the other end on hold.
So from that end I should listen uploaded file.
But I am not getting audio.
From memory, you need to
Hello all!
Is there a way to (mis)use asterisk itself as a softphone? Can
I make a call
from within the CLI? Can asterisk from itself produce a ringtone? I
Or can bind a system-command to incoming calls?
Any help is sincerely appreciated!
You can install a browser softphone on the same
Better still - is it possible to SSH (or some sort of connection
method)
from a remote PC to the Asterisk server and make a call using CLI?
Sure, you can use the CLI 'console dial' command.
Do you mean that I will be able to hear the call from my PC if I do
'console dial' on the remote
Doesn't Queuemetrics run on a license basis?
Anything else that's probably open source and free?
Does anyone have any comments/experience about using asteriskguru queue
statistics?
http://www.asteriskguru.com/tutorials/installation_guide.html
___
2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s
holdtime), W:0, C:134, A:48, SL:88.8% within 120s
Members:
Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet
Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet
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