FOP works depending on how your agents signin.
What are you using? Agentcallback or agentlogin?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Hall, Eric M.
|Sent: Lunes, 01 de Agosto de 2005 03:38 p.m.
|To: Asterisk Users Mailing List -
This is configured on your features.conf file. In there you can see what
keys to use to do blind and attended transfers, be sure those lines are not
commented out.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|[EMAIL PROTECTED]
|Sent: Lunes, 01
On Mon, 1 Aug 2005, Phoneguy wrote:
There are 2 methods blind and announced here you go:
Blind:Call someone, or receive a call. Hit 'Trnf'
The screen displays TRANSFER TO? and you hear a dial tone.
The other end can still hear you, so don't say anything nasty.
Dial the number and hit
Hi:
How do I check it?
Thanks
--- Darren Wiebe [EMAIL PROTECTED] wrote:
Check and make sure that astcc-config.conf is owned
by the same process
that owns apache. Usually the problem is that
astcc-admin cannot write
to the file due to permission problems.
Darren Wiebe
[EMAIL
Hi, I have the following configuration, which doesn't seem to work, any
help much appreciated
Linux 2.6.11 used to run asterisk
CVS version of zaptel
X101P
So far, so easy. However, whenever I turn the machine on with the card
in, I get
FXO PCI Master abort errors.
Depending on the way it
On 2 Aug 2005, at 01:10, Bill Wesson wrote:Hello list,This sounds interesting. Has anyone looked at the source code of these phoneclients. I would be reluctant to download and install software that could bea trojan software.Thanks,Bill WessonIf anyone is interested I'm (slowly) developing a GPL'd
hello,
What version of the linux are you using?
Do you disable hyper-threading, APIC, etc??
Thanks
CCF
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Miller
Sent: Saturday, July 23, 2005 03:33
To: Asterisk Users Mailing List - Non-Commercial
Maik Schmitt schrieb:
one of our customers which wants a soft transfer between his old pbx to
asterisk and sip. the setup is as follows:
telco ---pri--- asterisk ---pri--- legacy pbx
everything is fine exept that when dialling from the legacy pbx it takes
about 3 seconds before the asterisk
Hi!
I am trying to get my ISDN phone to work with my
asterisk box.
Now my asterisk wont start
Current situation:
I have a cable from my Billion ISDN (Bipac V1.0) to
my old NT1.
The cable is crossed like this:
1
2
3 - 4
4 - 3
5 - 6
6 - 5
7
8
Then I have a cable from
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit
Sent: Monday, August 01, 2005 10:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] what is the problem with gmail and the list.
I have not been
Hi,
I am using * behind a PBX for internal use (not yet ready to replace it
fully). Unfortunately PBx does not support Caller-ID, (or people here do
not know how to configure it) When a call comes, I want to get a
telephone no from customer, record a message and sent it as a mail, so
that we can
Hello Everybody,
I am a new user in this group.
I have installed asterisk on my test linux machine and setup the call from one asterisk user to another asterisk user successfully. It is working great.
Now i want to setup the call from one asterisk user to any PSTN user in the world or vice
Hi,
I came across this
nice looking web SIP dialer. However I cannot find how I can download it. Anyone
know how...??
http://www.geocities.com/babarnazmi/
SIP (Session
Initiation Protocol) based PC2Phone Dialer
Hello,
Here are my observations / Report on
what I see about the new Digium
TE110P Card.
- Recently we switched to the new TE110P
card in replacement of the old
E110P Interface.
- Unlike previous times with the old
E110P, this time we are seeing
some Random Problems with the new
I really think this matter deserves
attention. I have been asked many timesabout it.
Regards,
Victor.
Hello, I can understand why asterisk is
designed to not to allow two UAs with the same usr/pwd, http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html,
but I have to
If anyone is interested I'm (slowly) developing a GPL'd Java
applet that
works as an IAX softphone.
I should have a test version out at the end of the week for a
limited number of testers.
Tim.
http://www.westhawk.co.uk/
Hello Tim,
We'd be interested to
I have the following asterisk configuration (sip.conf) :
[General]
externip=82.79.81.3
localnet=192.168.10.0
localmask=255.255.255.0
[Phone1]
type=friend
host=dynamic
nat=yes
qualify=yes
context=sip
callerid=Phone1 1
disallow=all
allow=gsm
[Phone2]
type=friend
host=dynamic
qualify=yes
I am interested in how much CPU and RAM asterisk requires for call handling.
1. What is the minimum CPU required for asterisk to manage 60 concurrent calls
without transcoding.
2. Handle calls on a 75% no transcoding, 25% transcoding
3. How many calls can it connect per second ie from one
Aug 2 04:03:46 ERROR[1552]: Signalling requested is FXO Kewlstart but
line is in PRI Signalling signalling
This is your problem, probably in one configuration file you have fxo
kewlstart as signalling, and in the other one you have pri signalling.
Greetz,
Zoa
---
http://www.asteriskguru.com
How does TDM work, how do you connect to it?
I have the impression it can't be routed like ethernet, but a cable from your
switch has to be plugged into the providers equipment.
I have seen the Asterisk info about TDMoE - does this mean that the Asterisk
card will modulate the signal on the
Hello,
Here are my observations / Report on
what I see about the new Digium
TE110P Card.
- Recently we switched to the new TE110P
card in replacement of the old
E110P Interface.
- Unlike previous times with the old
E110P, this time we are seeing
some Random Problems with the new
Hello, long time lurker, first time writer
We have the following set up
ITSP
|
|
Internet
|
|
Cisco 2600
|
|
SwitchAsterisk Server running 1.0.9(has public ip)
|
|
Cisco 515e Pix Firewall running Pix OS 5.3(run's a class c 1-to-1 nat and pat)
|
|
Grandstream GXP-2000(run latest fw from
Do you know witch do I need to use?
-Opprinnelig melding-
Fra: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne av Zoa
Sendt: 2. august 2005 12:31
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [Asterisk-Users] Config HFC-card in asterisk.(Config the phoneand
Hello !
I got a dual E1 card from Dialogic (D300/SC-2E1 old card with ISA)
at my desk.
Is there a channel driver available for this kind of card ?
Best regards
Johann
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hello everyone, I have just received 3 brand new Polycom SoundPoint IP
600 from voisupply.com and I have the exact same problem on all of
them. When I receive a call, the phone is ringing correctly but when I
answer it, it takes exactly 10 seconds before I can hear the caller. I
also have
Hi,
I'm wondering if someone tried to use Asterisk and ISDN as an ISDN
simulator for Cisco lab.
Searched the net but I didn't find anything.
Thank you,
Mihai
Start your day with Yahoo! - make it your home page
Hello list,
we are experiencing a strange problem here. We offer SIP-Accounts to our
customers. Our asterisk connects via SIP to a remote PSTN gateway of a
TelCo.
In some outgoing calls (SIP to PSTN) the called party suddenly hears a
very loud beeping noise that goes on until the call is
hi,
I am going to open up a call center starting with 5 and expanding to 20
seats in 3 months. I have decided to use asterisk. I don't think I need
FXO or any other card from digium.
If you have any document regarding setting up a call center with
asterisk then please let me know.
What
Hello,
I have an Asterisk box with a TE410P connected to a PRI line and agents with
X-Lite installed on the same LAN as the Asterisk server. Sometimes, when I
make outbound calls it hangs up as soon as other party tries to picks up the
call. Does someone ever experienced this situation? On
What kind of call center: inbound, outbound or both?
how many lines per agent will you have?
what kind of trunks will you be using?
do you need to tie into an existing database?
do you want screen-pops?
MATT---
-Original Message-
From: Zeeshan [mailto:[EMAIL PROTECTED]
Sent: Tuesday,
The SC420 does not support APIC (at least not with current BIOS). The
Digium TDM400B card seems to always seek the same IRQ so it locks one down.
The clone X100P cards often get a shared IRQ, even if vacant ones are
available. I've found both the Digium card and the clones very frustrating
I've tried getting Festival working with Asterisk.
Here is what debugging from asterisk says:
-- Executing Festival(SIP/VoIP-e576, please record your message) in new
stack
== Parsing '/usr/local/etc/asterisk/festival.conf': Found
Aug 2 07:14:49 WARNING[49829]: app_festival.c:444
Zeeshan schrieb:
hi,
I am going to open up a call center starting with 5 and expanding to 20
seats in 3 months. I have decided to use asterisk. I don't think I need
FXO or any other card from digium.
If you have any document regarding setting up a call center with
asterisk then please let me
Thanks to everyone who responded. I have a pretty good idea now what we would
need!
___
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Perhaps a sip debug peer will shed some light? Never had an issue
like this myself and I am installing 25 more today so I hope not;-)
On Aug 2, 2005, at 7:10 AM, Rich Adamson wrote:
Hello everyone, I have just received 3 brand new Polycom
SoundPoint IP
600 from voisupply.com and I have
Hi
I'm trying to implement a Callback mechanism whereby I generate a Call
file and connect an arbitrary extension with my cellphone (via a SIP
Channel).
If I create a .Call file that connects the channel
SIP/[EMAIL PROTECTED] with a local extension/context I get some
weird issues with DTMF
hi list,
I'm running a newly installed [EMAIL PROTECTED] an i registered two soft
phone. both soft phone are registered
8901/8901x.x.x.xD 255.255.255.255 50710Unmonitored
8900/8900y.y.y.y D 255.255.255.255 6281
Unmonitored
but when I call
If i were to use Asterisk with, say 1000 IAXy's, is there a way to
provision them from a central spot? It would be very improper to have
to let each end-user do this by him/herself...
The manual and documentation for the IAXy is very limited, and i can't
find anything about this kind of things...
Let's start basic, we know that both PCs that are running the soft
phones can see the aah server, but can both PCs see each other? Can
they ping each other? (ie, they are not across a NAT router or
something like that?)
G
Mark Anthony C. Delfin wrote:
hi list,
I'm running a newly
hi
Solution of your problem is in this article which i am pasting from an
online document
A SIP phone usually registers with a SIP proxy. This message comes from the
inside of the NAT to the server on the outside. Now, there's an open
connection in the NAT device. As soon as there's no more
Hi!
I have set up my ISDN phone to connect to a ISDN card
in my Asterisk. It actually works now J
I use [EMAIL PROTECTED] 1.3
I can call inn and there is no problem talking on the
phone.
When I lift the headset of the phone I automatically
calls the extentions set in incoming
I have added the following to a macro that is used for all
extensions so a user can access voicemailmain by pressing * during the
voicemail prompt
; check voicemail
exten = a,1,voicemailmain(${macro_exten})
exten = a,2,hangup
The behavior is a little weird, the * key is not
On Tuesday 02 August 2005 06:35, Obelix wrote:
I have seen the Asterisk info about TDMoE - does this mean that the
Asterisk card will modulate the signal on the Ethernet cable to allow it
plug directly into a proper TDM connection?
TDMoE is just a method of taking the 8000Hz, 8-bit ulaw/alaw
On Tuesday 02 August 2005 06:16, Obelix wrote:
I am interested in how much CPU and RAM asterisk requires for call
handling.
I *really* dislike these kinds of questions.
Grab some hardware and try it. It is the *ONLY* way you will know for sure.
Grab a single processor Pentium 4 or Celeron
Try this line in /var/lib/astcc and see if it helps.
chown apache * and
chown -R apache * from /var/www/cgi-bin.
If it does not help, do an ls -l in /var/www and see who owns it.
Then repeat the command above with the correct user.
Darren Wiebe
[EMAIL PROTECTED]
chawki hammoud wrote:
You aren't dealing with analog phones, and you aren't transmitting DTMF
signals.. the functional difference between analog and digital systems
kindof precludes what you are looking to do.. meanwhile, once the entire
number has been dialed, the outgoing call should be started almost
Using Asterisk Management Portal with Broadvoice. It used to work just
fine; calls would come in and be answered with no trouble at all. A
few weeks ago with no configuration changes at all Asterisk stopped
picking up calls and started giving a busy signal whenever someone
calls.I've tried
And as much as you dislike these kinds of questions; its unfortunate
that the community doesn't have any good answers to them available--they
should be. It would be great if we could get some independent
verification of digium's claims/figures.
voip-info:
Hello everyone,
I have an IAX server ([EMAIL PROTECTED]) with a FXO card.
I have a trunk connected to a voip provide, asteriskout.
When I call my server from a public phone, I want to route this call
to the asteriskOUT trunk so that I can make long distance calls.
How can I setup a secret
Many people seem to want this feature. I think they are just
confused. I've never actually heard of a good reason to let multiple
devices register with the same username/secret. Most of the time they
want a call to ring on multiple devices and they are trying to make
a device == extension,
Hello everyone, I have just received 3
brand new Polycom SoundPoint
IP 600 from voipsupply.com and I have the exact same problem on
all of them. When I receive a call, the phone is ringing correctly but when I
answer it, it takes exactly 10 seconds before I can hear the caller. I also
have
You may have bought the Chinese Versions and hence the problem in slow
response.
Have you tried the US versions available from http://www.iareaphone.com
?
-S
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Redstone
Sent: Tuesday, August 02, 2005
Hello ALL
SS7 for asterisk release http://www.footnotess7.com/. but i not yet account to download.
any body have SS7. could you like send to me.
thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Graham,
Digium IAX2 FXS unit called IAXY is just no good. I would say that it is
garbage.
Try the IAX2 ATA ( AG168 sold as Netweb ATA-100) with a life line port
made by Atcom and available from http://www.iareaphone.com
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
With due respect to Digium and Mark Spencer and the greatest protocol he
defined, I have used IAXY and I regret to say that IAXY at $99 is plain
garbage compared to the $49 ATA made by ATCOM.
Try the ATCOM AG168 sold as ATA-100 by iareaphone.com. This has an
additional lifeline port and gives
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Adrien Laurent
Sent: 02 August 2005 14:56
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] How to create a secret code to use
[EMAIL PROTECTED] server's long distance plan from a
Use Googleextensively andthe WIKIsitehere http://www.voip-info.org/wiki-Asterisk,
till you become familiar with the architecture of Asterisk. probably for a
couple of months.
You can come back here if you still have any questions
at that time and all the member here would be happy to
Adrien Laurent wrote:
Hello everyone,
I have an IAX server ([EMAIL PROTECTED]) with a FXO card.
I have a trunk connected to a voip provide, asteriskout.
When I call my server from a public phone, I want to route this call
to the asteriskOUT trunk so that I can make long distance calls.
Your
Does anyone know if the 3rd patch listed on this bug fixes the seg fault
problem related to voicemail?
http://bugs.digium.com/view.php?id=4800
--
respectfully, Joseph
___
Asterisk-Users mailing list
mattf wrote:
What kind of call center: inbound, outbound or both?
It will be inbound 90%+ as I only need 2 seats for outgoing.
how many lines per agent will you have?
one line per agent.
what kind of trunks will you be using?
Don't know yet. I am open for options and basically I
Page 4 documents the Authenticate Feature. I'm a Newbie, so I can't give you
much more help beyond that, but it should point you the right direction.
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-pdf/vm1.pdf
Subject: [Asterisk-Users] How to create a secret
If you don't mind, can you follow up with this on the list. I'm
interested in learning how the different agent logins affect FOP.
Thanks,
Waldo
On Aug 2, 2005, at 1:46 AM, Nicolás Gudiño wrote:
Looking for a good web app that will show agents that are login to
queue. I tried the operator
Hello,
I am sure this has been answered so many times as it is one of the most
fundamental features of Asterisk.
Here is my scenario,
I have setup my asterisk server with a TDM400p which have one FXO and FXS
card.
My SIP server is up and its working fine only in SIP network ( I used ser)
For my
As promised, we just released the first version of the asteriskguru
Queue Statistics.
Screenshots and download at:
http://www.asteriskguru.com/tools/queue_stats.php
---
Small description:
The Asteriskguru queue statistics, is a PHP based program, which gives
anyone who uses queueing in
I'm interested in test the client. Please contact me
out of list [EMAIL PROTECTED]
Thanks in advance.
--- Vlasis Hatzistavrou - asterisk mailing list
account [EMAIL PROTECTED] wrote:
If anyone is interested I'm (slowly) developing a
GPL'd Java applet that
works as an IAX softphone.
On Tuesday 02 August 2005 09:55, Adam Dobrin wrote:
And as much as you dislike these kinds of questions; its unfortunate
that the community doesn't have any good answers to them available--they
should be. It would be great if we could get some independent
verification of digium's
Regardless of what has (or has not) been implemented in asterisk, there
is a very valid business reason for wanting an extension number to ring
on multiple phones and to determine the status of an extension from
multiple phones. Business have needed (and implemented) that for years.
Having such an
Hi there
I am wondering if anyone out there has used app_conference?
I am currently using meetme. My main problem with meetme is that one cannot
specify to stop sending voice packets to a participant when they are speaking.
This results in doubling the bandwidth for the participant
Hi there
I am currently using Asterisk with Meetme on a 2.4 linux
kernel. I am using Ztdummy with usb-uhci driver for timing. It seems to work
ok, although I havent tried it with more than 5 users. However, I am now
looking to move into a production environment and some people have said
How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?
___
Asterisk-Users mailing list
On Tue, 2 Aug 2005, Frank Sautter wrote:
Maik Schmitt schrieb:
one of our customers which wants a soft transfer between his old pbx to
asterisk and sip. the setup is as follows:
telco ---pri--- asterisk ---pri--- legacy pbx
everything is fine exept that when dialling from the legacy pbx
(Please forgive my sending this again, but the list seems to have been
acting up a little the last few days, and I didn't see it appear when
the list started working again)
We have a Shortel system at out main site. We're putting Asterisk
servers at several smaller remote sites. I know I'll
On 8/2/05, Victor Alvarez [EMAIL PROTECTED] wrote:
Hello,
I can understand why asterisk is designed to not to allow two UAs with
the same usr/pwd,
http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html,
but I have to find a solution for this.
My first option
I have been using Sixtel from the beginning of the year and service was
getting worse and worse. Yesterday I tried to access the website to get the
CDR and I got an error saying that the domain no longer exists. I checked the
whois and it says that the domain is on hold. Have they finally
Using the Xten X-Lite client (free) I am able to connect to a local
[EMAIL PROTECTED] server and when trying to connect to the remote server
(a mirror of the local) I am unable to connect.
The first server is a local lan, the remote is using microsofts pptp
vpn client to connect.
Looking at the
Hi all -
This isn't really directly Asterisk related, but has anyone successfully
set up a Polycom phone to register two lines on two different Asterisk
boxes? I can get the first line to register, but the second one does not.
I can still place calls from that second line, which indicates to me
Rich Adamson wrote:
Regardless of what has (or has not) been implemented in asterisk, there
is a very valid business reason for wanting an extension number to ring
on multiple phones and to determine the status of an extension from
multiple phones. Business have needed (and implemented) that
Hi,
We would like to hook a asterisk bases system into a
Artisoft Televantage based system.
Does anyone have a idea of how to do this? Basically we need to trunk H323
lines. Is there anywhere a howto, or any documentation relating to this?
Please let me know
Thanks
Sascha
Hello,
Does anyone know how Asterisk manages calls on a system? More
specifically, does it spawn a thread off of the asterisk program... are
they separate processes? We're trying to see what kind of system load
the PBX will create when calls are put through.
Thanks,
Tim
At 09:58 AM 8/2/2005, you wrote:
Adrien Laurent wrote:
Hello everyone,
I have an IAX server ([EMAIL PROTECTED]) with a FXO card.
I have a trunk connected to a voip provide, asteriskout.
When I call my server from a public phone, I want to route this call
to the asteriskOUT trunk so that I
I ring multiple phones ALL THE TIME without needing duplicate
username/secrets. The following line wrapped, but you can still see
what's happening. When someone dials extension 3400 the devices with
SIP the three SIP usernames (we set them to MAC-[a|b|c|d] where the
letter indicates which
Hello,
You have several choices if you are doing almost all inbound, here's a
summary:
- Native Asterisk Agents and Queues (easy to setup but no screen pops
native. need add-ons for that, some are commercial)
- There are several companies that sell add-ons for Asterisk
queues/agents to
It might be helpful if you posted your setup, and relative sections of your
extensions.conf etc.
Is this a new install? are you using VoIP extensions, FX, or what? Is the busy
signal when you call from one extension to the other, when you dial-out? or all
of the above?
Subject:
Please send this information to me also.
On Thu July 28 2005 01:03, Michael D Schelin wrote:
Hello everybody, for all of you that have searched for a real fax
solution, look no further. We now have T38 faxing. Please contact me for
more information.
Thanks
Michael D. Schelin
ShellTel
I found your original message:
Can somebody please help here. At least respond and call me a moron.
I have tried everything. I finally gave up and installed [EMAIL PROTECTED]
from the iso and I am back to the exact same problem. Everything seems
to work but my extensions are all busy. I used
Tim King wrote:
How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?
Once is enough. Perhaps you did not provide enough information for
anyone to help
-Original Message-From: Tim King
[mailto:[EMAIL PROTECTED]Sent: Tuesday, August 02, 2005 11:29
AMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: [Asterisk-Users] WHat does it
take
How many times do you ask for help here before getting a
respone? Every single
How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to
pay
someone to help here. Anybody got a clue?
go to the asterisk console, and do show dialplan, make sure things there
look as you expect from your
Hello,
We are using AstTAPI to make call´s from Outlook and it woks perfect. Our issue
is that it doesn´t work so well with our CRM. Although the TAPI line
initialization is successful, AstTAPI doesn´t report the Phone Number and TAPI
doesn´t know about our extension. Anybody knows how to
i think may be you should read this:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Macro
On 8/2/05, Damon Estep [EMAIL PROTECTED] wrote:
I have added the following to a macro that is used for all extensions so a
user can access voicemailmain by pressing * during the
I'm interested in test the client. Please contact me
out of list [EMAIL PROTECTED]
Gracias
FG
2005/8/2, Gonzalez Mata David [EMAIL PROTECTED]:
I'm interested in test the client. Please contact me
out of list [EMAIL PROTECTED]
Thanks in advance.
--- Vlasis Hatzistavrou - asterisk mailing
As a follow up to this thread..
We have been getting a number of cards now that have a PCI device id of
d161:0410. These all appear to be rev 2 digium cards and will not work
with version 1.0.7 (and previous) drivers. They require 1.0.9 drivers which
have this (and a few other) PCI device ids
Adam,
I thought Andrew Kohlsmith gave the individual good
advice without intentionally malaciously spitting in
the guys face.
For the question, 'Whats the ' Minimum CPU required
for 60 calls?
I think a Pentium 3, high end, which is cheap right
now, should do fine, but you will need either 3
Bruce Komito wrote:
I've been using realtime to store my voicemail configuration in a mysql
table for several months now, and have had no problems...until today. A
few weeks ago, I upgraded to the latest CVS and today I noticed voicemail
is not updating the password when the user changes it
Hi list,
we want to connect asterisk to an traditionnal PBX (EADS 6550/Matra).
People from telco told that they can't connect two PBX's using E1/T1 or
only with QSig signaling.
I wanted to use EuroISDN. In this case, it was me told that VN6-VN7
would be used. The PBX has a spare ADQ card
I have been playing
with a 480i with the new firmware 1.2.0.162I hope to get some form
of
paging intercom
function to work. In the wiki someone post that ALERT_INFO type of paging
might
be in this version
of firmware but I have been unable to find anything on this
yet.
I have tried
On Tuesday 02 August 2005 11:28, Tim King wrote:
How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?
Try giving us some kind of data to work from. This is
Carlos Chavez wrote:
I have been using Sixtel from the beginning of the year and service was
getting worse and worse. Yesterday I tried to access the website to get the
CDR and I got an error saying that the domain no longer exists. I checked the
whois and it says that the domain is on
Hello
I have a Premisys Slimline Channel Bank connected to a
Digium TE110P. I am not able to call the FXS extensions or get dialtone on
them. The channel bank is connected via a T1 crossover to the cable and lights
show green. I really need to get this functioning by end of day. If
Since my previous response was incorrect, I will go ahead and give the
instructions. I pulled these from the forum at
http://sourceforge.net/projects/asteriskathome/
Instructions for getting DISA to work with AAH (or AMP).
As Adrien pointed out, you should have Digital Receptionist set up or
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