Anyway to disable the join conference notification on the party that is
joining, but not the parties already in the conference?
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Hi all,
What is maximum number of three party conferences can a quadcore 3GHz
system can handle? All the parties a setup with G.711 codec.
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To
something you have to test and see. Using VICIDIAL in
performance testing mode I have gotten to over 100 conferences on a
similarly equipped server with a very rapid call turnover rate.
MATT---
On 5/15/08, Wai Wu [EMAIL PROTECTED] wrote:
Hi all,
What is maximum number of three party conferences
Hi list,
My long duration calls are being timeout by my SIP VoIP provider for
failure of receiving re-INVITE within their timeout limit. Is there a
way to config Asterisk to automatically send a re-INVITE message every
10 to 15 minutes? I looked into the sip.conf file and couldn't find such
a
Hi list,
I just installed 64 bit Linux, and ready to install Asterisk through
source on it. Are there any settings have to change to build 64 bit
Asterisk? Thnx a million.
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Hope you don't mind I jump in here. I am interested in DECT's handover
of live calls. My question is, does the IP address on the phone change
when moving from on access point to another?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Reeves
Sent:
Thanks. It make perfect sense. I was just curious why the manager app is
needed. Since the phone can see 4 AP at the same time, when it wants a call to
be handed over to a different AP, couldn't it just send a re-invite to Asterisk
and call it a day?
Wai,
The IP address is really on the
I have been following this discussion. You do have a point. However, the
way * works right now. If a channel does not require trans-coding to get
into a conference, coder usage is counted. So I really do not know what
difference putting the transcoding in meetme is going to make. I mean,
how could
But his preference of G729 is to save bandwidth.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton
Sent: Wednesday, October 03, 2007 8:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] meetme
Hi list,
Is there a limit on the length of an extension? I have an 18 byte long
extension, when issuing goto, Asterisk comes back with invalid
extension on the console. Anyone had this experience before?
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Hi list,
Has anyone use app_conference? I want to hear what your opinions are. Thnx.
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calling? Seems best to me to spy on an extension. YOu also
can do a show channels to see who is talking to whom.
on Wednesday 09/26/2007 Wai Wu([EMAIL PROTECTED]) wrote The parameter to
Chanspy should be the whole or part of the channel name.
I do not understand what you mean by sip trunk
Have you tried to load the driver with ec disable? Last time (long time
ago) when I was working on a quad card, we weren't able to get ec to
work with hardware ec on.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Alexander
Sent: Thursday,
Hi all,
Anyone know where the asterisk log file is stored? I have some failed
calls into my Asterisk box, and I just want to find out why those calls
failed. Thnx.
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Very nasty indeed. Through my experience with PRI, the TelCo switchs are
not that present to deal with. Your method will work, kind of. However,
if the TelCo decides to send you a call during that split second of
idle, how are you going to handle it. The best way is still to call your
TelCo to
Do your phones have the 172.17.x.x as the proxy address?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian M.
Arlinghaus
Sent: Wednesday, September 26, 2007 4:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
But you may want to read your asterisk.conf file to make sure the path in
which your system store it.
You will see something like this
astlogdir = /var/log/asterisk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Wednesday, September 26
The parameter to Chanspy should be the whole or part of the channel name. I do
not understand what you mean by sip trunk. It make perfect sense that you can
hear both streams of voice when you use the phone's extension as Asterisk
usually uses SIP/extension+xxx as the channel name of the call.
2007, Wai Wu wrote:
Hi everyone,
I am running into wall today with simultaneous call limits. I have two
Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a
lot of sip calls from one machine to the other by issuing AMI Originate
commands to one machine. The machine
to?
Thanks,
James Texter
On Fri, 2007-09-21 at 08:51 -0400, Wai Wu wrote:
I am not so sure if the interrupts has any thing to do with it.
I run some more test just now and I am getting these error on the
console of the call receiving machine. All it does is wait for 45
seconds. I think
Hi everyone,
I am running into wall today with simultaneous call limits. I have two
Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a
lot of sip calls from one machine to the other by issuing AMI Originate
commands to one machine. The machine that makes calls plays a
are to
them. They're not exactly what I was looking for, but maybe that will
help. All TCP connections require the Kernel to page the information
but I can't seem to find out how to access that limit if any.
On 9/20/07, Wai Wu [EMAIL PROTECTED] wrote:
Hi everyone,
I am
:
On Tue, Sep 18, 2007 at 04:22:29PM -0400, Alex Balashov wrote:
On Tue, 18 Sep 2007, Wai Wu wrote:
Any one know how to increase the Linux limit? I am hiting a wall on
200 calls playing files at the same time. From Asterisk console, I am
getting messages like
Sip_request_call: Unable to build
: [asterisk-users] Linux limits
You have to increase the amount of available file descriptors per
process:
http://hausheer.osola.com/docs/11%C2%A0%C2%A0
On Tue, 18 Sep 2007, Wai Wu wrote:
Hi all,
Any one know how to increase the Linux limit? I am hiting a wall on
200 calls playing files
Hi all,
Any one know how to increase the Linux limit? I am hiting a wall on 200
calls playing files at the same time. From Asterisk console, I am
getting messages like
Sip_request_call: Unable to build sip pvt data for asterisk1/700
Too many open files
Is this a limit of my Linux box? I only
Does anyone have this experience? My TCP connection the Asterisk Manager
Interface is chopped off after 15 minutes of operation.
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Don't know about IAX. As for SIP, You will know what ip address and port
the audios should be transmitted to by looking at the sdp session. Just
goto the * console and enable sip debug.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bilal
ghayyad
Sent:
Hi all,
Just ran into some issue with the originate AMI command. It seems that
there is a limit of around 120 calls I can place with the originate
command simutanously. By that I mean sending Asterisk a lot of originate
command very fast. Anyone know if there is a limitation? Thnx.
Of Atis
Sent: Monday, September 10, 2007 5:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Manager API - Originate command
On 9/11/07, Wai Wu [EMAIL PROTECTED] wrote:
Just ran into some issue with the originate AMI command. It seems
PGP SIGNED MESSAGE-
Hash: SHA1
Wai Wu wrote:
Hi all,
Just ran into some issue with the originate AMI command. It seems that
there is a limit of around 120 calls I can place with the originate
command simutanously. By that I mean sending Asterisk a lot of
originate command very fast
Just checked. I do have Async set to yes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Monday, September 10, 2007 7:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Manager API
Can someone share pointers to Asterisk's Dialogic support? Which boards
are supported, driver status, and etc.
Thnx
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] On Behalf Of Armin
Schindler
Sent: Tuesday, August 21, 2007 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Dialogic support
On Tue, 21 Aug 2007, Wai Wu wrote:
Can someone share pointers to Asterisk's Dialogic support? Which
: [asterisk-users] Question on the Monitor command on AMI
Try MixMonitor()
l.
In data Thu, 09 Aug 2007 00:24:47 +0200, Wai Wu [EMAIL PROTECTED] ha
scritto:
Hi all,
Is there a way to have this command to record a mixed file instead of
one for in and one for out? I have set the Mix parameter to 1
Hi all,
Is there a way to have this command to record a mixed file instead of
one for in and one for out? I have set the Mix parameter to 1, but it is
still generating two files. I would prefer it to have the in and out
files mixed. Thnx.
___
I don't see the point of the service provided by GrandCentral. Party A
calls party B through GrandCentral. Party B know party A's number and
calls party A back, now party A can call party B directly, and party A
has party B's directly number.
-Original Message-
From: [EMAIL PROTECTED]
Anyone?
-Original Message-
From: [EMAIL PROTECTED] on behalf of Wai Wu
Sent: Wed 7/4/2007 12:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Need advice to get wcte11xp and wcfxo to load
I have a X100P and a TE110P in my Asterisk box. I can
I have a X100P and a TE110P in my Asterisk box. I can either get the
X100P or the TE110P to work, but never both. Here's my zaptel.conf
span=1,0,0,d4,ami
em=1-24
fxsls=25
When I load wcte11xp and wcfxo, I will get this error.
[EMAIL PROTECTED] etc]# modprobe wcte11xp
ZT_CHANCONFIG failed on
Hi all,
I need the zap channels going, but got the following error. What do I
need to change in my configuration? Thnx.
chan_zap.c: In function `zap_send_keypad_facility_exec':
chan_zap.c:2309: warning: implicit declaration of function
`pri_keypad_facility'
chan_zap.c: In function
Subject: Re: [asterisk-users] Help. Cannot compile version 1.4.6 with
the following error
Wai Wu wrote:
Hi all,
I need the zap channels going, but got the following error. What do I
need to change in my configuration? Thnx.
chan_zap.c: In function `zap_send_keypad_facility_exec':
chan_zap.c
compile version 1.4.6 with
the following error
If you don't need libpri, you could just remove it. The problem is that
you already had it installed, and it was too old for newer versions of
Asterisk to use.
- Wai Wu [EMAIL PROTECTED] wrote:
Thnx. It is working now. I though that I didn't have
Hi all,
I setup two * boxes with two sip phones (one to each * box). I can make
calls from one sip phone to the other via IAX both ways. However, the
dtmf tones are just oneway. I use rfc2833 for dtmfmode in the sip.conf
on both * boxes, and gsm for IAX. What do I need to achieve twoway dtmf
Hi all,
Anyone know how to dynamically create meetme conference rooms on the
flight? I remembered a while ago there was a switch that tell meetme to
create the conference room is the room is not defined in the
meetme.conf. It doen't seem to be working for me anymore.
Thnx
] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Thursday, 8 March 2007 4:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Coaching in asterisk
Is there a way to setup a conference where party A can coach another
Party B, at
the same time
Two things.
1) This is a bug(feature) of standard analog switchs which only clear the talk
path when both sides of the call are terminated.
2) You should post this in the asterisk development list.
-Original Message-
From: [EMAIL PROTECTED] on behalf of Patrick Fortin
Sent: Fri
] RE: Coaching in asterisk
Wai Wu wrote:
Ouch, I just have to move to 1.4. Is 1.4 stable at all under heavy
load?
You're more courageous than I am.
-Stephen-
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on an extenal application controlling Asterisk through AMI.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Friday, March 09, 2007 12:22 PM
To: Asterisk Users Mailing
Is there a way to setup a conference where party A can coach another Party B,
at the same time, all other parties cannot hear party A? In order words, partis
A and B can hear every one, and party A can only be heard by party B.
Thnx
attachment:
] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Thursday, 8 March 2007 4:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Coaching in asterisk
Is there a way to setup a conference where party A can coach another
Party B, at
the same time
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 am use Fedora 3, and run into a 1.4 compile issue.
When 'make install' I got this message.
[EMAIL PROTECTED] asterisk-1.4.1]# make install
make: expand.c:489: allocated_variable_append: Assertion
`current_variable_set_list-next != 0' failed.
make: *** [utils] Aborted
[EMAIL PROTECTED]
Found out I need make version 3.8 or later
-Original Message-
From: [EMAIL PROTECTED] on behalf of Wai Wu
Sent: Thu 3/8/2007 5:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] 1.4 compile issue
I am use Fedora 3, and run into a 1.4 compile
Anyone?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Monday, March 05, 2007 10:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] TC400B
Anyone tried the digium TC400B transcoding card? What
Anyone know the gsm encoding mip requirement from g711? Or number of
channels can be transcoded from g711 to gsm at a time.
Thnx
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Anyone tried the digium TC400B transcoding card? What are your opinions?
Thnx
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I assume in this case, you are making the out of the PRI line. Well, that's
exactly how PRI works. What you should do is look at the progress code and
determine what the call status are (busy, disconnected number, moved number,
etc) and play a proper message for the customer. As for the second
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Anyone know how to
direct sip calls in a dial plan to a specific proxy if * is registered with more
than one proxy?
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You don't want to do that. The max I tried was 2.
From: [EMAIL PROTECTED] on behalf of Ard
Sent: Mon 6/5/2006 5:29 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] How many TE405 ...
Hi,
Is it possible to use 4 TE405 boards in one server ?
I can either get the TE100P working or the 3 X100P clones working, but
never both. I have the TE100P connected to a channel bank, and X100P
clones to lines from the phone company.
This is my zaptel.conf
span=1,1,0,d4,ami
fxsks=1-24
loadzone=us
fxols=25-27
loadzone=us
I then do
[EMAIL
and using either SPA-3000's, a TDM400, or a Mediatrix 1204.
Kerry Garrison
Publisher - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Tuesday, May 02
PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, May 02, 2006 10:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Need help configuring TE100P and 3 X100P
clonewith MD3200 chipset
On Tue, 2 May 2006, Wai Wu wrote:
[EMAIL PROTECTED] root
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] PRIs from two different telco
Wai Wu wrote:
One question thought, does
the hardware echo cancellation work much better than software?
I bought a Digium TE411P hoping the hardware echo canceler would improve
my
Hi,
I am getting this message on the * console on my first pri span. Pri
show span show it is down, and I can't make any calls from the span.
Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
Apr 27 07:40:23
some other type of signalling like EM.
On Thu, 2006-04-27 at 14:58, Wai Wu wrote:
Hi,
I am getting this message on the * console on my first pri span. Pri
show span show it is down, and I can't make any calls from the span.
Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI
taking an intterupt.
You might want to try to put the te411p card on a different cpu, or if
its probably an ide card doing it, try playing with hdparm (make your
drivers slower) or disable that card, and take a new one.
On Thu, 2006-04-27 at 14:58, Wai Wu wrote:
Hi,
I am getting this message
My TE411p does not seem to like to have two PRIs from different telcos
(span 1 and span 2). I can get one working, but not the other. However,
if I use vpmsupport=0 when loading the wct4xxp module, they both work.
But here is the problem, vpmsupport=0 disables the on board echo
cancellation. Any
] On Behalf Of C F
Sent: Thursday, April 27, 2006 12:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] PRIs from two different telco
You should really take this up with Digium support, and don't forget
to share your experience.
On 4/27/06, Wai Wu
, Wai Wu [EMAIL PROTECTED] wrote:
I just tried it. Same problem, one of the two spans is not working. If I
load wct4xxp with vpmsupport=0, then both spans working.
BTW, here is zaptel.conf
span=1,1,0,esf,b8zs
span=2,2,0,esf,b8zs
bchan=1-23
dchan=24
bchan=25-47
dchan
] PRIs from two different telco
On Thursday 27 April 2006 12:18, Wai Wu wrote:
My TE411p does not seem to like to have two PRIs from different telcos
(span 1 and span 2). I can get one working, but not the other. However,
if I use vpmsupport=0 when loading the wct4xxp module, they both work
PROTECTED] on behalf of Wai Wu
Sent: Thu 4/27/2006 3:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] PRIs from two different telco
Yes. It is always the same pri regardless port on the TE411P. If I disable the
hardware echo canceling
Time to report back. We took out the daughter board (no more hardware echo
canceling) on the TE411P, and the problem is gone. Guess we have to RMA the
card.
From: [EMAIL PROTECTED] on behalf of Wai Wu
Sent: Thu 4/27/2006 3:30 PM
To: Asterisk Users Mailing List
If I download zaptel-1.2.5, do I still have to apply the
zaptel-1.2.5-patch?
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Hi,
What is the version number of the lastest stable release, and how to get
it through CVS or wget? Thnx.
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Hi all,
I am running 1.2.7.1 asterisk on FC3. Every thing works except dtmf
detection on my Zap lines. I am using a TE411P with isdn NI2. Thnx.
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Hi all,
Is there a way to record a call conversation starting in the middle of
the call? I know I can recording whole conversation with mixmonitor, but
I prefer only recording certain part of the conversation. Thnx.
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Subject: Re: [Asterisk-Users] Call recording
On Thu, Apr 20, 2006 at 07:41:48PM -0400, Wai Wu spake thusly:
Hi all,
Is there a way to record a call conversation starting in the middle of
the call? I know I can recording whole conversation
Hi,
Anyone know how to compile asterisk for a hyperthreaded processor? Thnx
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Hi,
Does cvs checkout asterisk gets the later version of asterisk? I tried
cvs checkout -r v1-2-7 asterisk, and didn't work for me. The only
thing works is cvs checkout -r v1-2 asterisk. What exactly is version
tag for version 1.2.7? Thnx
___
).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth
Sent: Friday, April 14, 2006 2:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call center running
Asterisk-soundquality-critical!
Wai Wu wrote:
I did
I did not install soxmix in my linux box. If you having issues with
mixmonitor, you can put both legs of the call into a conference and
record the conference
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth
Sent: Thursday, April 13, 2006 1:20 PM
I just check the source code, Monitor uses ast_writestream and it
eventurally goes down to au_write, g723_write, etc. They don't commit to
the disk. So, in effect, if you have a lot of ram, the audio should stay
in ram until it gets swap out or the file is closed.
-Original Message-
I would say what is going to prevent content providers like google and yahoo
becoming telcos. Now they too would have their peer arrangement.
From: [EMAIL PROTECTED] on behalf of Bob's Leaky News Service
Sent: Thu 4/13/2006 8:26 PM
To: Asterisk Users Mailing
I think this belongs to the development mail-list.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver
Sent: Wednesday, April 12, 2006 12:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
running Asterisk - sound
quality-critical!
Hi,
how do you record calls? Monitor app. or MixMonitor or something else?
How does your storage backend looks like?
What kind of channels do you use? Do you record IAX2 channels?
Regards,
Tamas
Wai Wu wrote:
You got to be kidding about 53 calls being
Except that mixmonitor still has a bug in it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Wednesday, April 12, 2006 11:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call center
Asterisk-sound
quality-critical!
Wai Wu wrote:
Except that mixmonitor still has a bug in it.
What kind of bug? Issue number?
FYI: yesterday one issue has been fixed :D
http://bugs.digium.com/view.php?id=6457
Did you mean that type of bug? If something else, please let us know...
T
You got to be kidding about 53 calls being recorded at sametime is an issue. I
have done at least twice as many on my dual xeon 3.4Ghz system and had no
problem as clients like to record every call that goes through the system. Then
again, in my system, the in and out channels are mixed first
I am surprised that his was able to make outbound calls.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Friday, April 07, 2006 9:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Inbound PRI
Title: [Asterisk-Users] Anyone have a definitive list of Manager eventsper category?
hm, I have to try that. I am using for third party control
so the need to know all the events.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josh
McAllisterSent: Tuesday, April 04, 2006
Title: [Asterisk-Users] Anyone have a definitive list of Manager eventsper category?
Hi,
After
apply patch and make clean; make install. Do I have to do a make sample to have
new asterisk running?
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Isn't aheeva a commercial product? Whoever wants to find out how it is should
ask aheeva for referrals, and I recommand him personally pay visits to their
customers on their expenses if he is a prospect.
From: [EMAIL PROTECTED] on behalf of Kevin P. Fleming
a bug where
MixMonitor stops recording at random (see http://bugs.digium.com/view.php?id=6457).
There are a couple of working patches for it. Thanks.
On 4/3/06, Wai Wu
[EMAIL PROTECTED] wrote:
Hi
all,I am setting up a script to record all the call. There are two app
for recording. "Mo
I don't think you can selectively receive events. I am also write an app using
heavy manager actions, and I put the filters on my app. So far, I have not seen
traffic from these events do a dent to my application/network performance.
From: [EMAIL PROTECTED] on
Interesting about your tellco. A the tellco I have dealt with sent the
DISCONNECT message when a non-operational number is called. The usual
messages will come in the this order
1. proceeding
2. one or more progressing
3. disconnect with the cause value(if number is non-operational, or the
Hi all,
I am setting up a script to record all the call. There are two app for
recording. Monitor and Mixmonitor, one mixing the audio on the fly and one
mixing it at the end but also allow a option not to mixing the audio at all. If
mixing the audio on the fly is not that taxing on the CPU,
Hi,
Option 'e' is for
selecting an empty conference to join. My question is. How do I know what the
conference number is for the next party to join? Does it set it to a
variable?
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If your T1 is ISDN, you should automatically have it. If
your T1 is RobBit, you have to check with the CO to see is the ANI/DNIS service
is turned on (it is seperate service for RobBit T1), and the T1 is usually set
to wink start.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi,
Does Asterisk have
builtin (T1 or E1) span monitoring? If a span goes down, will asterisk know
about it. Personally, I would like to have a event generated through the Manager
API interface.
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List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Span monitoring
Wai Wu wrote:
Does Asterisk have builtin (T1 or E1) span monitoring? If a span goes
down, will asterisk know about it. Personally, I would like to have a
event generated through the Manager API interface.
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