what we needed and I want to ask if I understand the naming correctly:
FXS = pstn-signals for calling someone (towards central pbx/server) and
knowing that someone is calling you
FXO = ...?
FXS has a phone plugged in it
FXO hgas a phone line plugged in it
I have been searching for the necessary components for my setup from sometime back;
yet to install Asterisk and will be installing softphones on Linux Server and on all windows PCs(most of them are Windows Xp,others are Windows 2000 professional,Windows 98); but could not decide which softphone
so i was looking at the internet and i read a lot, the cheapest are the
Grandstream BudgetTone
but some reviews of this list says they are not so good ... so i found
Many people hate these phones, yet I've found my 3 BT100 to be
excellent for a network of friends and associates (not everyday
Is it possible to bypass incoming ring on asterisk so that incoming
calls come to asterisk box will be directed straight into did?
Try setting callerid=no on the FXO channel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi,
try xlite if you have enough bandwitdh for G711 codec requirement..
try firefly if you want to use G729 codec freely (linked via dll)..
both of them are the best freeware softphone for windows.
Best regards,
Stevanus
infra struct wrote:
I have been searching for the necessary
Hi,
Were looking for an experienced Asterisk engineer/programmer to
configure and install Asterisk systems.
This is a full time position and the person will be based in Asia. Share options are available and we are open to
negotiate.
Minimum of 1 year experience installing and
Bugger, thanks for replying and telling me, might send a request through
to Grandstream and see when they intend on releasing it.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Svensson
Sent: Thursday, 9 June 2005 3:54 PM
To: Asterisk
Hi List,
I have a test asterisk box with a TDM400P with 4 FXO modules plugged in.
Yesterday I could use the box without any issues - no problems.
This morning, the sound on the box was absolutely horrible. After some
fiddling about, I have rebooted the box, and now asterisk refuses to start!
Hi,
I've tried your suggestion but the result is still the same...
Have another suggestion?
Best regards,
Stevanus
Wilson Pickett wrote:
Is it possible to bypass incoming ring on asterisk so that incoming
calls come to asterisk box will be directed straight into did?
Hi Michel,
Michel Brabants wrote:
I didn't see a bri-adapter on the digium-site, only pri it seems. Any
recommendation on that or is there a bri-adapter from digium. I'm also
open to other vendors. I saw that there are others which ar ecompatible
with asterisk, but I don't have a lot of time
Nobody ? :-(
-b
- Original Message -
From:
barney
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, June 08, 2005 11:39
AM
Subject: [Asterisk-Users] performance of
* in several scenarios
Hi,
Is here someone who could
hi all
i have a notice message that comming frequently says
that pbc.c:1329 pbx_extention_helper; cannot find
extention context 'default'
anyone know this warning and how to solve because its
realy anoying
thks
roy
__
Do You Yahoo!?
Tired of
Hi,
This is to let you all know that I have it working now. Thanks to
Titus who supplied his list of a working combination (
http://amatisoft.homelinux.com/demo/index.html ) and some other tips.
For archive and history purposes I will post my combination which may
help others who will run into
Jean-Michel Hiver a écrit :
Hi List,
I have a test asterisk box with a TDM400P with 4 FXO modules plugged in.
Yesterday I could use the box without any issues - no problems.
This morning, the sound on the box was absolutely horrible. After some
fiddling about, I have rebooted the box, and
I've made a backport of this patch for asterisk stable. You can get it
here: http://www.maxosystem.net/asterisk . The page is in Spanish, but
you just need to download and apply the patch to chan_zap.c. It also
works with bristuff patch applied.
Julian J. M.
On 6/9/05, Neil and Fiona [EMAIL
Hi,
I'm pulling my hair out, cause cannot connect to EuroISDN BRI in Italy with
octobri card from Beronet. I use bristuff and have following zaptel.conf...
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
#
# First come the span definitions, in the format
# span=span num,timing,line
This is mostly a testing/bug fix release. Hopefully by the next
version I will have some real documentation up on the site. Since
it's primarily a platform rather than an end user system, without
documentation it's not nearly as useful as it could be.
http://asterisk.ochsnet.com
Chris
You're connected to a p2mp bri, switch to bri_cpe_p2mp
Matteo.
Il giorno mer, 08-06-2005 alle 19:54 +0200, Robert Rozman ha scritto:
Hi,
I'm pulling my hair out, cause cannot connect to EuroISDN BRI in Italy with
octobri card from Beronet. I use bristuff and have following zaptel.conf...
Hi all,
We are successfuly running an Asterisk server with standard SIP hard
phones and it is working well. We are looking to deploy some soft
phones on our Linux desktops. There seems to be several floating
about. Anyone out there with some good/bad experiences with particular
Linux softphones.
Hi,
I recently got a TDM04B and after installing and getting asterisk up and
running I connected a handset to one of the ports. Unfortunately I don't
get a dial tone when I lift the handset.
What could be the cause of this?
Could someone point me in the direction of a proper config for a
Hi,
try xlite, it has linux version..
Best regards,
Stevanus
Eric Bishop wrote:
Hi all,
We are successfuly running an Asterisk server with standard SIP hard
phones and it is working well. We are looking to deploy some soft
phones on our Linux desktops. There seems to be several floating
Hi,
-Original Message-
We are successfuly running an Asterisk server with standard SIP hard
phones and it is working well. We are looking to deploy some soft
phones on our Linux desktops. There seems to be several floating
about. Anyone out there with some good/bad experiences with
On Wed, 2005-06-08 at 23:39 +0100, David John Walsh wrote:
Angus
Jumping in with both feet
a BT socket with a capacitor in is commonly refered to as a Master
socket, and are very cheap even without wholesale. It gets its name
from being the socket that BT installed into the house for
On Thursday 09 June 2005 00:52, James Bean wrote:
span=1,1,0,ccs,hdb3
The same thing happens.
Did you rerun ztcfg? I have heard rumour (but not seen it myself) that you
need to fully reset (power off/on, not just reboot) to get the card to accept
a new clocking method.
You may consider
Is it true that a FXO port will NOT provide a dial tone?
On Wed Jun 08, 2005 at 08:23:54PM +0300, [EMAIL PROTECTED] wrote:
Hi,
I recently got a TDM04B and after installing and getting asterisk up and
running I connected a handset to one of the ports. Unfortunately I don't
get a dial tone
On Thursday 09 June 2005 06:32, [EMAIL PROTECTED] wrote:
Is it true that a FXO port will NOT provide a dial tone?
FXO means it connects to a central Office -- it accepts dialtone and ring (it
acts as a telephone)
FXS means it connects to a Station (telephone) -- it provides dialtone and
ring
Voilà.
Now i know where is the problem.
I use 2 ISDN channels with a with a fritz! card and the junghanns capi
drivers.
The problem appears with SIP to ISDN calls.
The SIP 180 ringing message doesn't appear because the ISDN PBX sends
the ALERT message in-band (channel B), and not in the D
Inbound is the problem - I am regisitering to the host and receiving faxes.
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Robert Goodyear
Sent: Wednesday, June 08, 2005
I'm *positive* that Digium (or your reseller) will swap this out for no extra
charge unless there's a restocking fee for having to order it in.
Actually, last time I looked there wxas a difference in price. Weren't
the FXO a little more expensive?
___
We are setting up asterisk with Teliax and having trouble getting the
incoming call to work
all the time, the outgoing does not seem to have a
problem.
Here's what I've been using for the last several months:
[teliax]; for incoming calls
context=teliax-incoming
type=user
auth=md5
Hi all, first post. My company's office in the UK is soon going to get a
Cisco VoIP solution system. What I am interested in, and couldn't find
googling, is if it is possible to connect an Asterisk solution to the
Cisco system and have all the nice advantages of it (mainly calling the
i am a newbie, but have you tried
genzaptelconf -s -d
Dan
On 6/8/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
All,
I have an [EMAIL PROTECTED] installation with a TDM40B card. I can make
internal
IP calls with no problems, but when I try to dial out I get a message that
All
Hi,
when i use the *8 for the call pickup the call i fetch is directly
connected and i can't see the callers number.
What i want is that the call in the first rings at my phone and in the
second i can see the callers number.
I am using a polycom 500 ip phone. Is this a special polycom
On Thu, 9 Jun 2005, Andrew Kohlsmith wrote:
I also check if I'm loosing interrupts and everything seems ok. Also I
pull out the TDM400 from the box.
This tells me it's got nothing to do with the TDM400 or lost interrupts.
It could be that the user-land side (i.e. Asterisk as opposed to
I have had good success with my efforts to send faxes over voip using ulaw,
surprisingly, and I want to move it from testing to reality. I have an
account with Teliax, who have been very good. For voice I use g729 and ulaw,
but for faxing I can only allow ulaw. However, Teliax only sets the
[EMAIL PROTECTED] wrote:
Hi,
I recently got a TDM04B and after installing and getting asterisk up
and running I connected a handset to one of the ports. Unfortunately
I don't get a dial tone when I lift the handset.
This board is FXO which you plug incoming phone lines into it. So plugging
Thanks Julian,
I tried installing cvs-head today but it crashed on compile and the
machine rebooted when I did make clean.
I'll try the patch and see how I go.
On Thu, 2005-06-09 at 08:55 +0100, Julian J. M. wrote:
I've made a backport of this patch for asterisk stable. You can get it
here:
On 6/7/05, Michael Graves [EMAIL PROTECTED] wrote:
On Mon, 6 Jun 2005 11:17:20 -0600, Colin Anderson wrote:
http://www.pcmag.com/article2/0,1759,1812887,00.asp
Specifically, his assertion that ISP's would sniff traffic and block, say,
the SIP port. You could play wack-a-mole with port
I recently got a TDM04B and after installing and getting asterisk up and
running I connected a handset to one of the ports. Unfortunately I don't
get a dial tone when I lift the handset.
What could be the cause of this?
Could someone point me in the direction of a proper config for a
Thanks. Seems I misunderstood quite a bit.
On Thu Jun 09, 2005 at 07:43:11AM -0600, Rich Adamson wrote:
I recently got a TDM04B and after installing and getting asterisk up and
running I connected a handset to one of the ports. Unfortunately I don't
get a dial tone when I lift the handset.
Hi,
I've looked pretty wide on Google for this so I don't think it's been
asked before. Has anyone had experience integrating Asterisk with a 3COM
NBX system? The only way that I can see that looks possible is via the
3Com NBX ConneXtions H.323 product and then one of the H.323 Asterisk
I got these errors and my hardware is working so I do not think they
are an issue
Hint: insmod errors
Removing zaptel module: zaptel: Device or resource busy
What about the ztdummy module?
Dan
On 6/8/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Dean,
Here are the results of the
If they are truly two separate accounts with two register
statements (and two userid/passwords), I would guess that two
different incoming outgoing contexts would work with iax.
They are two separate accounts, but the teliax server will always
authenticate itself with the username teliax,
Getting the ztdummy module out of the mix corrected the problems I was
having. Try the following:
genzaptelconf -s -d
After that has completed, with lines attached, I did a
rebuild_zaptel
Your mileage may vary, but this seemed to do the trick for me.
- Original Message -
From: Dan
I have just signed up with Voxee.com and have attached my Asterisk
server to dial them via IAX2.
Below is the start of the log which dials the number and promply
hangs up when the call is answered, with the logs saying that the
channel is not compatiable.
I have traced this down to
If they are truly two separate accounts with two register
statements (and two userid/passwords), I would guess that two
different incoming outgoing contexts would work with iax.
They are two separate accounts, but the teliax server will always
authenticate itself with the username
Did you go to the web page that is listed at the bottom of every
message?
Look at the bottom of that web page for the address of the list admin.
By the way, that admin address is pretty much standard for ALL mailing
lists.
On Wed, Jun 08, 2005 at 07:37:35PM -0700, David Koski said:
Would an
Voxee will not accept any calls that are not in G729.
You need G729 codec on your Asterisk.
Period.
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Thursday, June 09, 2005 10:08 AM
To: Asterisk Users Mailing List -
On 6/9/05, Phuong Nguyen [EMAIL PROTECTED] wrote:
Hi all,
Does Asterisk support multi thread? I mean:
Is it possible to do one of the 2 following scenarios:
1. Play a low background music when the user record his/her voice
I don't know why you would want to do that, but here is a hack.
Try this... get rid of the friend and use the peer and
user defs.
Then pay close attention to which parameters apply to those two defs.
Change the [teliax] to something different, like [teliax-in]
and [teliax-out].
All that will do is separate the configs for incoming and outgoing, I
Kanuri,
That must be why, on their setup page (in the members section of
their site), they list ulaw, alaw, g.729, gsm and ilbc as "Supported
Codecs" ;-)
We've used them with ulaw, recently :-)
Kanuri, Seshu (Company IT) wrote:
Voxee will not accept any calls that are not in G729.
Hi everybody,
Just to clarify, voxee supports the codecs G729, ulaw, alaw, gsm,
ilbc. You will need to force select a particulat codec out of those if
you want to use it.
We purchased the g729 codecs directly from Digium so it should have no
problems working with your * installation if you also
Eric Bishop wrote:
We are successfuly running an Asterisk server with standard SIP hard
phones and it is working well. We are looking to deploy some soft
phones on our Linux desktops. There seems to be several floating
about. Anyone out there with some good/bad experiences with particular
Hello,
A long Google search didn't turn any clear answer. Does somebody use
Asterisk in combination with Lingo?
Thank you,
Bas
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
Martin Croome wrote:
Hi,
I've looked pretty wide on Google for this so I don't think it's been
asked before. Has anyone had experience integrating Asterisk with a 3COM
NBX system? The only way that I can see that looks possible is via the
3Com NBX ConneXtions H.323 product and then one of the
I guess that's Early Media Connect, i.e., if the phone supports that
(not all do), the channels get bridged just after dial completed, (SIP
183), and what you hear is the remote ring tones (from your telco),
not locally generated (as if it received SIP 180 Ringing).
What IP phones are you using?
I've just checked the download page, and the latest firmware available
is 1.0.1.8. Where did you find 1.0.1.9?
This phone has some nasty bugs, one of them being that the other end
HEARS you after you press the Transfer button and you hear a dialtone.
It doesn't send any message to asterisk so
Here's what it looks like Robert
-- Executing VoiceMail(SIP/6153245827-0a2e,
[EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL
According to the fab sheet for the Dlink router they provide, it's SIP with
G711, G723, G726, G729.
Order the service, get the router, plug the WAN port into your LAN, fire up
Ethereal, power up the router, and sniff what's being passed, you might be
able to determine the user/pass, IP and codec
Thanks again Julian.
Quick update. Worked great with 1.07 (Which is good since cvs head gave
me hell today)
On Thu, 2005-06-09 at 08:55 +0100, Julian J. M. wrote:
I've made a backport of this patch for asterisk stable. You can get it
here: http://www.maxosystem.net/asterisk . The page is in
Rich and anybody else on Teliax might want to check a couple of times. I
have seen a few people having the same issue in the last couple of weeks.
We have been seeing this if we do random tests between 5-60 min.
I have tried one other thing in combination with Rich's config is to use one
of the
Hi,
I've been having some problems with my internet connection, it
cuts out for aprox 30 seconds at a time and after that i have
to do a reload in asterix for it to re-register my sip account with
broadvoice otherwise it won't accept any connection till i reload, is there
a way for it to
The telephone company in Honduras say they will only supply an E1
circuit with SS7 signaling. Has anyone else run into this?
Can anyone recommend a work-around for this problem?
Thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi Olle.
Exactly! Im using Nortel SIP server to register an * box. Could you
point me on some documentation about this new feature?
Thanks a lot.
Denis.
On 08 de jun de 2005, at 07:52, Olle E. Johansson wrote:
I guess you are registering with the Nortel SIP server? All the
incoming
Title: Message
Hi,
I have
bought two Cisco 7960 phones.
I have
tried to set-up them to work with Asterisk over Skinny protocol, but when I try
to dial the phone from Asterisk it says that all lines are
busy.
Is
there something that should be configured on the phone's side? Can someone
I have bought two Cisco 7960 phones.
I have tried to set-up them to work with Asterisk over Skinny
protocol, but when I try to dial the phone from Asterisk it says that
all lines are busy.
Is there something that should be configured on the phone's side? Can
someone help me with that?
1 2 [EMAIL PROTECTED] wrote:
Anyone know if
- it is possible to limit 1 agent per extension where
the last agent to log in overrides any previous agents
or
- a Command/application to clear all agents logged in
on extension
Does this look like it would require a custom mod to
do it?
Try this... get rid of the friend and use the peer and
user defs.
Then pay close attention to which parameters apply to those two defs.
Change the [teliax] to something different, like [teliax-in]
and [teliax-out].
All that will do is separate the configs for incoming and
If you activate (via sip.cfg) the feature Group Call Pickup, its no
surprise that asterisk doesn't know what to do with this feature
request. But it is being sent as a SIP SUBSCRIBE request, and I'm
wondering if, as asterisk stands, there is a way to take advantage of
this feature to emulate the
For redundancy I would like to write the CDRs on
tow mysql servers.
cdr_mysql.conf accept only one configuration
[global],
how to add a second host?
Thanks
Rosario
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
The problem was the dtmf, changed to rfc2833 and it works like a beauty.
setup the galaxyvoice for the incoming(free) and route calls through sunrocket and broadvoice.
thanks,
bill
On 6/8/05, Wilson Pickett [EMAIL PROTECTED] wrote:
when i try to dial a number it just dies.Meaning what? Silence?
Joseph wrote:
On Thu, 2005-06-09 at 02:24 +1000, Julien Goodwin wrote:
On 8/06/2005 11:37 PM, Sergio Chersovani wrote:
Joseph ha scritto:
When sending a call to a line defined on chan_sccp, there is an
error on the console that says:
Jun 7 08:22:29 WARNING[3924]:
I've been having some problems with my internet connection, it
cuts out for aprox 30 seconds at a time and after that i have
to do a reload in asterix for it to re-register my sip account with
broadvoice otherwise it won't accept any connection till i reload, is there
a way for it to
I have looked at the wiki and the mailing list. But
I need to find how do we setup the external IP address and the rtp ports for the
Polycom IP-500 and IP-600. There web interface has a nat setting but can't
find instructions on how to set this up. I would like to set this up via there
ftp
On Jun 9, 2005, at 7:33 AM, Chris Stinson wrote:
Here's what it looks like Robert
-- Executing VoiceMail(SIP/6153245827-0a2e,
[EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED]
[EMAIL PROTECTED][EMAIL PROTECTED][EMAIL
Rosario Pingaro wrote:
For redundancy I would like to write the CDRs on tow mysql servers.
cdr_mysql.conf accept only one configuration [global],
how to add a second host?
Thanks
Rosario
The quickest way would be to make a copy of cdr_addon_mysql and rename
the app and conf file,
Hi everyone:
I try to install astGUIclient for my call center. I'm interesting to put in
work the monitoring client, i follow step by step the installation from
scratch but when i try to run the application from my Windows XP
astGUIclient i got the follow error:
Client does not support
a simple workaround is to put a cronjob to execute
#!/bin/bash
asterisk -rx 'reload'
of course, i think that the best choice is find out why is needed to
reload, i dont think that is a normal behaviour
best regards
On 6/9/05, Armand Sulter [EMAIL PROTECTED] wrote:
Hi,
I've been having some
t downloaded the latest chan_sccp and am having problems with internal
to internal calls with callerid. I added a few debug lines to the
code to help sort it out, but here's what happens... Exten 581 calls
580. On the display 581 shows Unknown number to 580. On exten 580,
the display
On Jun 9, 2005, at 6:47 AM, Chris Mason (Lists) wrote:
Try this... get rid of the friend and use the peer and
user defs.
Then pay close attention to which parameters apply to those two defs.
Change the [teliax] to something different, like [teliax-in]
and [teliax-out].
All that will do is
Definately problems with voice quality and caller ID is not working
very well. I have e-mail a couple times and still no response from
their tech support on this. This is very concerning since I tried all
3 servers with the same results.
On 6/8/05, Julio Arruda [EMAIL PROTECTED] wrote:
Roman
Top posting to keep with the flow...
I'd have to guess at a couple of things here. Its already been stated that
asterisk _only_ uses the first of multiple dns A records when queried.
It would appear the voip.teliax.com dns A records point to 208.139.204.232
and 208.139.204.228, so one would have
I have the exact same problem.It would ideal if we could set an
astersik box with 2 E1 ports to do an IP-to-SS7 conversion. Anyone has
done this before?
C. Savinovich
At 11:08 AM 6/9/2005, you wrote:
The telephone company in Honduras say they will only supply an E1 circuit
with
Seshu,
Are you working on a VIA based motherboard?
I am working on a VIA based motherboard.
Andy Powell (author of Asterisk Live! distro) tells me that VIA is not quite
good when emulating i686 behavoir and since his distro is compiled for i686...
We are trying to confirm that but may be
On Thu, 2005-06-09 at 11:20 -0500, Moises Silva wrote:
a simple workaround is to put a cronjob to execute
#!/bin/bash
asterisk -rx 'reload'
in /etc/ppp/ip-up.local
put service asterisk reload
each time the connection is made then * will reload.
--
Dave Cotton [EMAIL PROTECTED]
Any other confirmation for this problem? My service seems to be fine
but I have not completed a long duration call yet. I had a user
complain last week about call degradation after 5-10 minutes but that
has been it. I will test some more and let you know. I am on the west
coast server.
You don't have a context called 'default'. Several parts of asterisk
will default to going to that context unless specified. Usually it will
be empty for security reasons do to that. Not certain what part from
the error message is trying to reach it, but creating a empty default
context
Well, sort of :)
We have agents using the AgentLoginCallBack functionality. The agents
log in using their agent number, with the extension automatically
entered for them.
When they log out, they again use the AgentLoginCallBack app, but using
just a # for the new extension (logs them out).
abel wrote:
Seshu,
Are you working on a VIA based motherboard?
I am working on a VIA based motherboard.
Andy Powell (author of Asterisk Live! distro) tells me that VIA is not quite
good when emulating i686 behavoir and since his distro is compiled for i686...
We are trying to confirm that but
On Thu, Jun 09, 2005 at 12:51:30AM -0300, Alejandro G wrote:
I should tell you that the TE100P is connected to another E1 board (not a
live E1) from Natural Microsystems which acts as a gateway to PSTN. This
board works as a PRI master but I don't think that this could be the problem
as long
Hello,
This issue was just handled Monday on the astguiclient-users list:
http://sourceforge.net/mailarchive/forum.php?thread_id=7448401forum_id=4358
6
You just need to use OLD_PASSWORD in the SET PASSWORD for your mysql server
to get the auth method for that account back to the pre 4.1.12
On Thursday 09 Jun 2005 17:29, abel wrote:
Seshu,
Are you working on a VIA based motherboard?
I am working on a VIA based motherboard.
Andy Powell (author of Asterisk Live! distro) tells me that VIA is not
quite good when emulating i686 behavoir and since his distro is compiled
for i686...
actually this is a freebsd box and its not behind NAT, i'm not sure if
it has to do with my internet connection anymore.
Right now it says :
bsd*CLI sip show registry
HostUsername Refresh State
sip.broadvoice.com:5060 username 3032
why not just use mysql replication to the second one?
On 6/9/05, Rosario Pingaro [EMAIL PROTECTED] wrote:
For redundancy I would like to write the CDRs on tow mysql servers.
cdr_mysql.conf accept only one configuration [global],
how to add a second host?
Thanks
Rosario
Hi, just wondering if my question is just unusual or if it is a quite
stupid one. Thought there would be someone having this kind of scenario,
but maybe I'm wrong.
btw, have a nice day
Simone
Simone wrote:
Hi all, first post. My company's office in the UK is soon going to get
a Cisco VoIP
We have a customer considering migrating from a large Nortel PBX with a
third-party voicemail system to Asterisk but one of the features they
really like is the automatic synchronization of voicemail between
Exchange and their voicemail system -- delete a message from the
voicemail system and
On Jun 9, 2005, at 8:51 AM, Rosario Pingaro wrote:
For redundancy I would like to write the CDRs on tow mysql servers.
cdr_mysql.conf accept only one configuration [global],
how to add a second host?
Might be easier to add a second host as a replica server with the mySQL Administrator.
Hi everyone:
I try to install astGUIclient for my call center. I'm interesting to
put in work the monitoring client, i follow step by step the
installation from scratch but when i try to run the application
from my Windows XP astGUIclient i got the follow error:
Client does not support
fire up Ethereal, power up the router, and sniff what's being passed,
you might be able to determine the user/pass, IP and codec
Proxy and codec is simple enough. But the authentication is via a
challenge-response on SIP so that will be a lot harder, if you have
the computing power to crack it.
-Original Message-
From: George Pajari [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 09, 2005 10:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Voicemail and MS Exchange Synchronization
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(a) Has anyone cracked this nut (or
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