We're using an SC420 and using BRI with a quadbri Junganns card, with IAX
softphones and one hardphone.
Working well except that we sometimes get dropped connections between IAX and
the server with a max retries exceed message, which comes from the chan_iax
driver code. The BRI side of things
On Fri, 17 Jun 2005, Paul Redstone wrote:
We're using an SC420 and using BRI with a quadbri Junganns card, with IAX
softphones and one hardphone.
Working well except that we sometimes get dropped connections between IAX and
the server with a max retries exceed message, which comes from
We upgraded to cvs-head a couple of days ago, but haven't changed the
config files at all.
Prior to the upgrade, all inbound calls to our call queues were
recorded, and the filename was like agent-6043-1109793719-24472.gsm.
After the upgrade, the filename is now 1109793719-24472.gsm
What do
Hi,
My telco provider is SBC. I think that they use FSK to transmit caller ID.
How can I set-up Asterisk so that I can see caller ID on incoming calls.
Regards,
Stojan Sljivic
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Calin Serbanescu
Hi all
Am experiencing a very weird behaviour with sip info DTMF . I have an *
box which is a satellite hop away. When I make a call from a Grandstream
call set to send DTMF thru sip info, am able to navigate through the *
menus very well. meaning that the DTMF is being recieved very well.
In article [EMAIL PROTECTED],
Kevin Bockman [EMAIL PROTECTED] wrote:
Yes, meetme requires a clock source. You could try ztdummy. I tried
using an FXO card as a clock source and observed that SIP calls connected
to the conference seemed to get out of sync. Basically, after perhaps 20
In article [EMAIL PROTECTED],
qrss [EMAIL PROTECTED] wrote:
Yes, meetme requires a clock source. You could try ztdummy. I tried
using an FXO card as a clock source and observed that SIP calls connected
to the conference seemed to get out of sync. Basically, after perhaps 20
minutes or so in
We upgraded to cvs-head a couple of days ago, and now get a whole slew
of warnings in the error log:
interface.c: Junk at the beginning of frame xx
has anyone else seen this, or do I need to incur the wrath of developers
and post this to the -dev list ;)
Julian
Title: Message
Hi,
Has
anyone experienced the same problem.
My
telco provider is SBC.
Regards,Stojan
Sljivic
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stojan
Sljivic - GDSSent: Tuesday, June 14, 2005 14:48To:
'Asterisk Users
Agreed,
I will post some pics early next week:)
Wojtek
- Original Message -
From: Nicols Gudio [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, June 16, 2005 8:27 PM
Subject: [Asterisk-Users] Viva Madrid!
enough said
--
Nicols Gudio
Buenos Aires - Argentina
Hi;
Connected ata186 sip version 3.1.1 to IVR system...able to
call / receive calls from ata but does not accept any dtmf
When internal extension is dialed, it is not recognized and
continues IVR musicDoes antbody has any idea how to make dtmf
configuration from ata186 sip version?...it
[outgoing]
ignorepat = 9
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5061)
exten = _9.,3,Playback(nomoreline)
exten = _9.,4,Hangup
Chris
- Original Message -
From: Martin Roy [EMAIL PROTECTED]
To:
Hello,
When dialing somewhere and the other side is down, Asterisk waits until
dial timeout before sending CHANUNAVAIL. I think that if after several
seconds there are not any reply (I mean at the IP level) we could
consider that the link is just down and handle the situation.
Is it
Robert Rozman wrote:
Is framing and coding (ami,ccs) right for Italy ?
They are dummy settings with bristuff. The example config will surely do :)
--
Emanuele
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
The call generator is very not user friendly now and undocumented, i
recommend not to use it and use some simple script somewhere for now.
(you could find some scripts somewhere on astertest.com).
I will put fixing that callgenerator on the (big) todo list.
Zoa,
Matt wrote:
Has anyone gotten
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Yves
Sent: Friday, June 17, 2005 2:09 AM
To: Asterisk - Users
Subject: [Asterisk-Users] Dial timeout when server down
Hello,
When dialing somewhere and the other side is down, Asterisk
waits
Hi, I have the SuSe9.2 installed in a box with a QuadBri.
I have followed all the instructions i have found and this is my best
result... only one error compiling zaptel :( .. y have the kernel
sources an already made the links to its
drwxr-xr-x 8 root root 328 Jun 16 20:20 .
drwxr-xr-x 12
Americo
60+60 isn't a VoIP term directly but a generic one within the telephony industry
if it were 60+30 it would mean the following
You are billed for 60 second as soon as the call is answered, even if
you only stay on the line for 7 seconds
The +30 then referes to the onward billing cycle,
Hi,
I've bought a Siemens GSM-modem based on the Siemens TC35-module.
I studied the operation manual of the modem and found, that for
transferring voice via the RS232 wire, the module supports
RS232-mulitplexing and wires the voice data on a separate
channel (whatever this means on RS232?).
When using the dial command and the D option to send DTMF digits when
the channel is answered, is there a way to allow for some dead air,
and then send more DTMF digits? I would like to automate a call, and
it requires entry of a few short dtmf digits all a couple seconds
apart from each
First, let me apologize for the multiple posts - my procmail recipe had a
bug that hid most mail form the list for a day.
The inheritance of includes creates a problem for me. I want to group the
extensions, not put them all in default to control access to features. So
[office] extensions should
I have a question that I've so far been unable to find the answer to:
Using an E1 interface for my PSTN connection I want to setup 5 SIP
phones in a call group (with a unique number for inbound calls) but only
allow the call group to receive a maximum of 3 calls at any one time.
Does Asterisk
If I have multiple Sipura 3000 device how can I dial out properly? I
can receive call without any problem and that's working really well.
Caller ID is shown and when someone call he get's the welcome message
the same way I have it configure with the X100P card. I don't seem to
have
Hi,
what new Dell servers are compatible and KNOWN to work with Digium TDM
cards? I've looked at Digium's compatibility list
at http://www.digium.com/index.php?menu=compatibility. Does this mean
that other Dell servers like SC1420, SC1425, 800, 1800 are working just
fine with TDM cards?
Can
Hello, we would like to hook up analog modems behind an Asterisk server,
and we're very interested in the experiences that others have made when
attempting that. We assume that there are no inherent problems with
modems in respect to the Asterisk software, but it appears that the
FXO/FXS
Hi,
-Original Message-
Currently, we only transmit at 1200bps, is this rate problematic with
Digium cards? Up to what data transmission rate are Digium
cards known
to work reliable? We do not think we'll ever go beyond
9600bps, can we
do this with a let's say TDM400P?
On a
Hello, we would like to hook up analog modems behind an Asterisk server,
and we're very interested in the experiences that others have made when
attempting that. We assume that there are no inherent problems with
modems in respect to the Asterisk software, but it appears that the
FXO/FXS
Is there a way to define multiple contexts for agents/queues such
that in a multi-tenant environment, there could be two different,
say, Agents 1000?
I'm setting up a multi-tenant configuration and I'm giving each
tenant a web-based interface to define their own agents and I
wouldn't
Btw here is an article on google maps that
I wrote about the other day.
http://www.smh.com.au/news/Technology/Map-hacks-make-data-come-alive/2005/06/16/1118869033845.html
This is one of the best examples for www.craigslist.com I have ever seen http://housingmaps.com/
Cheers,
Here we have PowerEdge 2850's doing the donky work with a Wildcard
TE405P in each.
I have seen no operational issues at all with the system or the cards.
We are running CentOS 3 as the operating system and the stable
version of asterisk
The only niggle is that when the cards are modprobed on
Can show me some link for the Last FaxApp sources (working with last
spandsp - i think pre18)
This link is not working. ftp://ftp.soft-switch.org/pub/spandsp.
ftp://ftp.soft-switch.org/pub/spandsp
The Domain soft-switch.org ftp://ftp.soft-switch.org/pub/spandsp is
not resovable.
And what is
Thanks for the reply. I'm more interested in lower series then 2850,
like PE SC1420, PE 800. I don't
need so much power. ;-)
-David
Here we have PowerEdge 2850's doing the donky work with a Wildcard
TE405P in each.
I have seen no operational issues at all with the system or the cards.
We are
But when BT-100 calls 7960 the following is happening:
-- Executing Dial(SIP/3710-8f2b, SIP/1707|15) in new stack
-- Called 1707
-- SIP/1707-e96a is ringing
-- SIP/1707-e96a answered SIP/3710-8f2b
-- Attempting native bridge of SIP/3710-8f2b and SIP/1707-e96a
May 4
Hi,
Is it possible to use an Asterisk box as billing gateway in a PSTN network? (Asterisk box somewhat connected to PSTN switch)?
In case the answer to the above question is yes, how to proceed?
Thanks,
Simon
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
I saw this in the list awhile back, it helped me setup my sipura 3000s
to act as trunks
Setup the PSTN side of the Sipura 3000 as a trunk within Asterisk
In AMP add an extension (e.g. 200) to correspond to Line 1 on the SPA,
ensure that port is 5060 and context is from-internal. It should be
named
I'm using autodial in conjuction with TxFax to send faxes on demand.
An home made application generates the call file and puts it in the
outgoing spool, the file is like this:
Channel:Zap/g1/1232314324
MaxRetries:0
RetryTime:60
WaitTime:20
Context:faxout
Extension:s
Hi!
I have configured a SIP trunk with a dialing rule.
The trunk behaves normally for incoming calls but when in used for
outgoing call a strange thing happens.
When I place a call I cannot hear the tone confirming that the remote
phone is ringing. I simply hear the voice as soon as the party
I bought a Dell SC1425 and installed a T1/E1 card from Digium and I tried to
configure it using [EMAIL PROTECTED] scripts and did not work, so I went the
long way and
configure with zaptel's instructions and voila! It works like a charm.
Oswaldo
-Original Message-
From: [EMAIL
If you set up only internal extesions in the default context, then include
default in [building1] and [office] all of those extensions can call
internally. I set up several standard features into the default context which
everyone can access.
If you want to control feature access, say, for
Hi
I am trying to achive this for a specific need of a customer.
He has a DID pointed to an Asterisk server, I need to provide him dialtone
when the calls hits the server. How can I achieve this?
Let's say something like this:
Exten = s,1,Answer
Exten = s,2, Provide Dial tone
Exten = s,3, Dial
Dear All,
I'm running version 0.7.1 of Asterisk server for our global help desk.
We have put together a comprehensive reporting package for static's from
the CDR.
I'm not able to calculate the time a call is in the queue before it goes
to an agent?
I would appreciate help with working this
Check out DISA.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Oswaldo Arratia
|Sent: Friday, June 17, 2005 7:51 AM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: [Asterisk-Users] 2nd
Hi,
I am programing one aplications to hear my e-mail on PBX asterisk. I
want to have music on hold when the my aplications get the e-mail the
mail server real.
I am doing un IVR to this aplicationes.
How I can do that?
When is the routines on the source asterisk to music on hold?
What files
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Anton Krall
Sent: Wednesday, June 15, 2005 3:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] WiFi IP Phones
Guys.
I know there are wifi sip phones
it sounds like you need to investigate the application called DISA..
it might be what you are looking for
if not, what dial tone is your client expecting? (internal, external - other??)
David
On 17/06/05, Oswaldo Arratia [EMAIL PROTECTED] wrote:
Hi
I am trying to achive this for a specific
Time waiting for an agent is one of the fields recorded in the queue_log
see the following
http://voip-info.org/tiki-index.php?page=Asterisk%20log%20queue_log
Regards,
Mac.
-Original Message-
From: Shad Mortazavi [mailto:[EMAIL PROTECTED]
Sent: 17 June 2005 15:54
To:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Dean Collins
Sent: Thursday, June 16, 2005 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] WiFi IP Phones
Ahmm Andrew, are you sure they are
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Huddleston, Robert
Sent: Tuesday, June 14, 2005 3:49 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Anyone
The libsupertone library installation happened because the libxml2 library
was not installed. After it was installed the problem was solved.
Now I am experiencing the problem following:
Do you have any tip???
Thanks.
[chan_unicall.so] = (Unified call processing (UniCall))
== Parsing
Given that they are radio transmitters, there is always the risk that
they can cause a spark and ignite something. Additionally, reports have
happened of the battery itself getting shorted when removed and causing
everything from bullets to other explosive situations to occur. When
you short the
We have been running Asterisk for about a month now
and one of the things I miss the most is the ability to se whos online
and available and whos not. Surely, theres the manager interface,
but unless youd want to install extra software on each client computer,
this is not a good option.
Manuel Casal ha scritto:
I made the make menuconfig and make dep in the kernel sources.
i do not remember well how i solved that problem but i'm sure that make
dep will issue you a warning and stop.
run make to start the kernel build process and then stop it after few
seconds. it will
Hi all, DID is Direct Inward Dial where the last 3 or 4 digits of the
dialed number are passed through and are used/translated to call a
specific extension. (See Centrex)
DISA is Direct System Access where incoming line(s)are auto-answered and
receive internal dial tone, the caller then has
Hi Bjorn,
Maybe it could be done as some form of
check against call forward to voicemail etc.
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bjorn
Sent: Friday, 17 June 2005 11:51
AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Hi,
This error I got it just when I gonfigure zaptel support isdneuro 31
channels.
But if I configure zaptel to support T1 and just 24 channels I have no
problem.
#
# Zaptel Configuration File
#
# This file is parsed by
cheapest in the world.
Ha Ha Ha Ha
How is your electricity sold? Hour, Watt, or by unit (KW/h)?
And as for cell phones being cheap, you have a receiver pays setup! How good
is that, then you have so many competing Telcos that sometimes you just
cannot call the house across the street
On 6/16/05, Tony Hoyle [EMAIL PROTECTED] wrote:
Terry H. Gilsenan wrote:
And as for cell phones being cheap, you have a receiver pays setup! How good
is that, then you have so many competing Telcos that sometimes you just
I believe that's unique to the US, the idea of paying for actually
Title: Untitled Document
Hi All,
I have configured Line1 (2011)and
Line2(2012)in SipuraSPA-2000 (latest Firmware)to use
G729. In sip.conf I have set disallow=all, allow=g729
IfLine1 is in use by an agent, then Line2 won't
work and viceversa (Inbound Calls Only).I have 40 license for G729.
I hava tdm400p card on [EMAIL PROTECTED] box, this card works fine for weeks,
today i did a CVS update to the latest head files and the card is not working.
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart
For anyone else who might run into this, I got around the transferring to
voicemail problem by putting a canreinvite=no line into the section for each
caller's SIP address in sip.conf. Not ideal, but it works.
I also had to add a dtmfmode=inband for my Mediatrix 1204 addresses to be
able to
The Sipura SPA2000 only supports one G729 call at a time. Same with the
Linksys PAP2.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Fri, 17 Jun 2005, David wrote:
Hi All,
I have configured Line1 (2011) and Line2 (2012) in Sipura SPA-2000 (latest
If you tell an american 'this is it', s/he will think of ways to change it and
make
it another way, thats why we have a '96 telecommunications act, and
why having cell phones in the states are the cheapest in the world.
Oh yeah. Americans are always faster, better, bigger. We cheese eating
Title: Untitled Document
Hi,
The Sipura SPA-2000 can only support one G729
call
Regards
Erick
- Original Message -
From:
David
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Friday, June 17, 2005 11:33
AM
Subject: [Asterisk-Users] G729
Marco Parmeggiani escribi:
Manuel Casal ha scritto:
I made the make menuconfig and make dep in the kernel sources.
i do not remember well how i solved that problem but i'm sure that
make dep will issue you a warning and stop.
run make to start the kernel build process and then stop it
Manuel Casal ha scritto:
make[1]: Entering directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp'
make[1]: *** No rule to make target `modules'. Stop.
make[1]: Leaving directory `/usr/src/linux-2.6.8-24.16-obj/i386/smp'
make: *** [linux26] Error 2
Hi ALL.
I have a problem with TxFax application. (RxFax is working properly)
Txfax does not work when sending tiff files bigger than approx. 18Kb. If tiff
file smaller than 18Kb everything is OK. Tested in LAN with Panasonic UF-E1 fax
machine (tiff files was created by the rxfax from that
What card do you have ? Is there are jumper setup that you can specify
E1 or T1 ?
E1 cards a shipped set up as T1 by default
regards
m.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Do you have analog TDM in it?
-David
Oswaldo Arratia wrote:
I bought a Dell SC1425 and installed a T1/E1 card from Digium and I tried to
configure it using [EMAIL PROTECTED] scripts and did not work, so I went the
long way and
configure with zaptel's instructions and voila! It works like a
Hello,
I am not too familiar with the settings in our PIX (learning though).
Here is the only access-list setting that we have in place for Asterisk:
access-list acl-prod permit udp any host EXTERNAL_*_IP_HERE eq 5060
In rtp.conf we are allowing ports 1 - 2.
We are not using SIP Fixup
I don't get a queue_log file?
At what stage was this introduced?
Thanks
Shad
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
I'm getting exactly the same behavior as was posted about in
http://lists.digium.com/pipermail/asterisk-users/2004-March/040380.html
I've upgraded (both ends) to CVS stable (CVS-v1-0-06/17/05-13:15:49).
Jun 17 13:46:17 NOTICE[15942]: chan_iax2.c:4053 authenticate: No way to
send secret to peer
Maybe, but that would not
have been a reliable way of handling it, as not all users would necessarily use
voicemail. Besides, I would think that this feature is supported by several SIP
devices (it has to do with messaging), so it would be better If Asterisk
supported this feature by
[EMAIL PROTECTED] ~]# modprobe zaptel[EMAIL PROTECTED]
~]# modprobe wcfxoZT_CHANCONFIG failed on channel 1: No such device or
address (6)FATAL: Error running install command for wcfxo
[EMAIL PROTECTED] ~]# ztcfg -vvv
Zaptel
Configuration==Channel map:Channel 01: FXS
I only get the first 1cm of the page in tiff.
Already tried to change the version of libtiff (.71), spandsp (pre18),
asterisk (CVS) and nothing!
The quality of image on that small band (1cm) is perfect.
Miguel
___
Asterisk-Users mailing list
Bjorn,
Take a look at the Asterisk Management Portal at http://sourceforge.net/projects/amportal
It has a flash-based panel that will give you what you are looking for.
Bryan M. Johns
One Ring Networks
300 West Wieuca Road, NE
Building One
Suite 205
Atlanta, GA 30342
404.303.9900 x: 104
Bjorn,
Take a look at the Asterisk Management Portal at http://sourceforge.net/projects/amportal
It has a flash-based panel that will give you what you are looking for.
Bryan M. Johns
One Ring Networks
300 West Wieuca Road, NE
Building One
Suite 205
Atlanta, GA 30342
404.303.9900 x: 104
Quick question about call routing. We're currently setting up our
system so that any phone calls made from our system over a t1 line to
another legacy system go through a dedicated t1 server. Is there any
method of checking to see if a number dialed exists on the system? Any
help would be
Take a look at the Asterisk Management Portal at
http://sourceforge.net/projects/amportal
It has a flash-based panel that will give you what you are looking for.
No need to install AMP to get this, just install FOP : http://www.asternic.org/
hth
Hi
I have Asterisk up and running perfect with a Digium TDM400P card and 4
FXS ports. There are 4 ATT 4-Line 954 phones hooked to the system Each
of the 4 lines is hooked to each phone. The problem is when you are on
line 1 (or any line) and someone else picks up line 1 they can here the
On Fri, 2005-06-17 at 12:33 -0400, David wrote:
Hi All,
I have configured Line1 (2011) and Line2 (2012) in Sipura SPA-2000
(latest Firmware) to use G729. In sip.conf I have set disallow=all,
allow=g729
Please take the time to read the Sipura documentation where it states
that
I have tdm400p card on [EMAIL PROTECTED] box, this card works fine for weeks,
today i did a CVS update to the latest head files and the card is not working.
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS
On Jun 17, 2005, at 7:56 AM, Daryl G. Jurbala wrote:
You find me a reliable Teir 1 ISP T1 in New Hope, PA for $300 to $400
and I'll give you the amount I save over the next quarter. NPA-NXX is
215-862. Good luck.
That sounds almost like Xeno's Paradox there... if you gave away the savings you
Title: MGCP files for Polycom
Does anybody know were I can download the MGCP files for the Polycom IP500?
Thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
It looks from here like you've rebooted the system after checkout, but your
system was not configured to load zaptel drivers at boot time.
Have you forgot to do 'make config' while in /usr/src/zaptel ?
Hope this helps
- Original Message -
From: David Romero [EMAIL PROTECTED]
To:
-- Started music on hold, class 'default', on
SIP/193.111.200.67-0815c790
-- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1'
-- Called Agent/1001
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/101|20|tr) in new
stack
-- Called 101
-- Agent/1001
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Saturday, 18 June 2005 2:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bill seconds
cheapest in the world.
Ha Ha Ha Ha
How
You find me a reliable Teir 1 ISP T1 in New Hope, PA
for $300 to $400
and I'll give you the amount I save over the next quarter. NPA-NXX is
215-862. Good luck.
Ive got a Full T1 from a rather
large Mid-Atlantic CLEC for $291. Ive got about dozen of them
Hi noticing the cvs updates of late, I'm wondering if there is support
for fifo/shell commands in the extended dialplan language? can it
fully replace agi scripts? Looks really interesting...
___
Asterisk-Users mailing list
Good Day
Has anybody here looked closely at the call cost calculation in ASTCC?
Can you duplicate the way the cost of a call is calculated? I believe
that there is an error in the code. I have fixed it, I think and
submitted a patch but we need user comments. I would appreciate if
We have been running Asterisk for about a month now and one of the
things I miss the most is the ability to se whos online and
available and whos not. Surely, theres the manager interface, but
unless youd want to install extra software on each client computer,
this is not a good option.
did you udate first?
- Original Message -
From:
David Romero
To: Asterisk-Users@lists.digium.com
Sent: Friday, June 17, 2005 9:36 AM
Subject: [Asterisk-Users] tdm400p not
working after cvs-head update
I hava tdm400p card on [EMAIL PROTECTED] box, this card
t in your dial statement?
- Original Message -
From: Neil Bullock [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, June 17, 2005 3:48 PM
Subject: [Asterisk-Users] callqueues confused :(
-- Started music on
Hi,all,
Now,I am working at make an realtime monitorfor the call center based on asterisk.
and ,I had search the archive and wiki.Through the return info from the management API,
I canget the waiting calls ,abandoned calls ,hold time, etc,but I don't know how to get
the number of incoming calls.
Greetings! to all
Now, with some hard time and help from many genurous people's in the list, I
have come to this point with my TDM20B card my teliax's IAX2 account.
I hope someone may help me with this issue mentioned below. I have already
selected my codec as gms in my iax.conf as well as in
If I set up an * server will I still be able to use my local Anchorage
phone number through my * box?
Thanks for any help,
Jon
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
95 matches
Mail list logo
