still wonder why the Nortel
does not accept the Calling Party Category Request and Switch to Group
C signal. Is the Nortel PBX properly configured for México variant???
Moisés Silva
On Wed, Feb 27, 2008 at 8:56 PM, Andres Tello Abrego [EMAIL PROTECTED]
wrote:
Thanks Carlos...
Using mx
,10,4
You only get 10 digits from the phone company.
On Wed, 2008-02-27 at 18:03 -0800, Andres Tello Abrego wrote:
protocol-variant mx,20,4
___
-- Bandwidth
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have a astunicall-1.4 setup with a te110p to a nortel pbx in Mexico.
(Hate R2!).
This is what I get when trying to call to * box using testcall:
./testcall
Chan 31, class 'mfcr2', variant 'mx,20,4', end 0, caller 0, from '' to ''
Loading protocol
There should be other voices worth while...
Give other people the chance
The market is growing...
Be open :)
Roy Sigurd Karlsbakk wrote:
Hi
The latest new about the Open Source Asterisk PBX is now that mister
Spencer himself has forbidden Allison to work with any other open
source
As far I remember. I had one Linejack, ISA bus... Line jack is unable to
place calls, only to recibe calls.
The channel is a phone channel, you need to use the telephony driver
from the kernel for linejack, and configure phone.conf of asterisk and
then use it as any other channel.
Ganbaa
Mexican Spanish..
Ha, funny term...
:)
Mexican Spanish = mx from MeXico...
es = ESpain...
So, es would be.. humm. Espain Spanish?
Chuck Bunn wrote:
Hi,
What is the best way to allow a user to select the language they hear in
the dial plan? In other words I want the phone to answer Hello
Mexican Spanish..
Ha, funny term...
:)
Mexican Spanish = mx from MeXico...
es = ESpain...
So, es would be.. humm. Espain Spanish?
Chuck Bunn wrote:
Hi,
What is the best way to allow a user to select the language they hear in
the dial plan? In other words I want the phone to answer Hello
Version
smbserver*CLI show version
Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
2005-10-20 01:13:39 UTC
exten = 1100,1,Dial(Zap/5,30,Ttr)
exten = 1100,n,Dial(Zap/8Zap/5,30,Ttr)
Doesn´t dial zap/8 zap/5...
* output:
-- Nobody picked up in 3 ms
--
Buy a 3 porst fxo card and 1port fxs (green) card from digium.
Plug your fax the the fxs port.
Assign an extension to the fax at extension.conf
Create a menu.
Since the call will be bridged from fxo to fxs natively, there is very
few loss and the fax works ok.
Anyway, the diferrence between
The issue, is that when you make goto(disa_custom|s|1) the line is
already answered...
Jump directly to disa_custom|s|4 or remove from 1 to 3 steps and re-number..
Min Hwan Chang wrote:
DISA is currently disconnecting when I dial to access DISA.
Below is my extensions.conf file from
The issue, is that when you make goto(disa_custom|s|1) the line is
already answered...
Jump directly to disa_custom|s|4 or remove from 1 to 3 steps and re-number..
Min Hwan Chang wrote:
DISA is currently disconnecting when I dial to access DISA.
Below is my extensions.conf file from
youa re using -v option multiple times at startup.
That message is perfectly fine.
ali kia wrote:
hi ;
our * handle good the outgoing calls but 4 incaming calls we have this
msg :
Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2
(Ring/Answered)...
someone
Where is the source code?
It looks interesting, But I´m interested in modifying the app, so, only
with a administration password o even a file, you can configure the app...
Users are quite curious about menus and always are looking to improve
de functionality of the configured software...
I have a instalation using IAXY.
I have 2 laptops with DIAX.
An entry call from a tdm400p fxo port to diax, is all ok.
A call between diax to diax and all is ok...
An entre call from tdm400p fxo port to iaxy, is all ok
A call between iaxy to iaxy,using ulaw, is all ok.
BUT...
When calling
But those configuration doesn´t mean fault tolerancy.
When implementing a load balancing cluster, failovering is only achieved
using information replication on line...
If you don´t make some how the information persistant, then you will no
be to take over the failed server. You will only be
Better use User Mode Linux proyect (UML).
Then you can have 2 diferent machines...
You don´t have to change ports, Using non standar ports, sometimes is
dificult to configure the clients... You expect the IAX port to be at X
port, no to be at Y port, sometimes... and more if using sip devices...
check that the phone cord is correctly attached to the tdm11b.
Also verify if the channel isnt alarmed with
zap show channel N whewre N the channel number.
[EMAIL PROTECTED] wrote:
Hello
i use TDM11B,and i'm connecting the analog telephone to channel 1, and i'm
trying to use
*CLI dial 11
at
But that kind of functionality can be achived with a creative dialplan,
and queues, and I belive, is far more cheaper that having cisco 7960
phones...
And also, it doesn´t need a lot of creativity...
Add info to the CID so, it will display the department they line is
trying to get...
Race
O yes :)
So, for having a load balanced call administrarion. Using linux virtual
server proyect I have run across a few design issues.
The idea is easy. 3 servers. The Main server, to recive the calls, and
the other 2 to originate the call.
The load balancing of generating the call can be
Verify what
zap show channel N says.
I have a fxo channel alarmed, and even I unplugged and re-plugged my
phone cord, and stoped and restarted my * box, I needed to unload the
wcfxs and zaptel modules and load them again to get the alarm cleaned...
Remember to replace N for your channel
Me :)
Jeffrey Paul wrote:
After reading the wiki, it would appear that the SetLanguage
application will set the proper language variable for use by the rest of
the speaking applications, such as SayNumber. However, from my
(limited) knowledge of counting in Spanish, it's not quite as
Hummm.. thats cool..
I havent tougth about being re-provisioning the iaxy box :)...
But how do you detect the dns change? wich ddns company are u using?
Jim Van Meggelen wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Wilson Pickett
Sent: October
The way I explain the difference between fxo and fxs, is electrically...
The telco lines, PROVIDES voltage and current to makes the telephone
ring, so the FXO is a device, adapted to recive this voltage an current.
The phone, NEEDS voltage and current, to ring, so the FXS is a device,
adapted
I have blown one fxs module in that way :(
It´s a warning, and you MUST be careful of tring no to do so.
Steve Underwood wrote:
Andres Tello Abrego wrote:
The way I explain the difference between fxo and fxs, is electrically...
The telco lines, PROVIDES voltage and current to makes the telephone
could you paste the file u are trying to provide?
rich allen wrote:
iH
getting the following in the console when i go off hook from IAXy
chan_iax2.c:5713 socket_read:Rejected connect attemp from 192.168.0.5
requested/compability 0x4/0x24 incompatible with our capability 0xff3
any idea what this
:) I supposed that
iaxy are s beatiful!
rich allen wrote:
well it looks like i had a typo in the iax.conf. instead of disallow=all
i had dsallow=all which i didnt notice til i about to paste the file.
thanks for responding
- hcir
could you paste the file u are trying to provide?
rich allen
netstat -a -p -n | grep asterisk
will give you a very good idea of what ports protocol does asterisk
have opened..
Your Own ISP .com wrote:
I also would like to know the ports needed for * to run without any
issues. I am not using NAT but I do want a firewall in front of the machine.
of people.
It'll get there I'm sure.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andres Tello Abrego
Sent: October 16, 2004 7:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAXy setup
:) I supposed
Ok...
The scenario.
Requirements:
User do not want to switch to linux.
User do not want to use a softphone.
User will not be at a fixed position.
User is using windowsXP.
* PBX is at a dinamic dsl IP using an aulloxdns dinamic service...
Does each time a iaxy box boot, needs to be provided with all the
information? de ip from dhcp, and all iax related info?
Or each time the info is provided, using iax provider, then... the
information is overwritten?
___
Asterisk-Users mailing list
a) use a transparent bridge firewall
b) Use redirect with multiport of the sip ports to the * box IP.
c) And the most effective for your topology, don't use nat, use only
the routing properties of linux...
can u post ur firewall rules and routing table?
Bastian Schern wrote:
Hello *,
I try to
What about a LVS cluster with persistence?
In that way, the phone register always to the same cluster member...
And what u are balancing are the # of sip phones a server can handle...
On Thu, 29 Jul 2004, Kanwar Ranbir Sandhu wrote:
On Thu, 2004-07-29 at 00:36, Greg Hill wrote:
I
That would be great
And for that kind of massive conference attendant, a app for * that can
receive an mp3 stream and transfor it as a channel feed for a conference
room, would be the solution.
I can provide a * server for the astricon with relative high bandwith, for
a parallel conference
I have Asterisk CVS-HEAD-05/25/04-17:13:22, Copyright (C) 1999-2004 Digium.
Usign exclusively digium hardware.
3 TDM400P cards.
1 4xFXO
1 4xFXS
1 1xFX0 3xFXS
When * is attending FXO calls, bridged to FXS calls, natively ofcourse,
at a random time, the call hangus up.
Also, for example, if a
Thanks, I dont have very long cables, but I will try to use other cables.
I'm not very happy with the radioshack phones, I belive, that may be
they are the issue too..
On Wed, 28 Jul 2004, Ryan Courtnage wrote:
On July 28, 2004 08:13 pm, Andres Tello Abrego wrote:
I have Asterisk CVS-HEAD
No, u CANT dial out with a linejack...
I have 2 of them, I use them as incoming only lines, and x100p for
incoming, outgoing lines...
There is no bug about this, is a feature, that isn't present at the
linejack.
On Wed, 2 Jul 2003, Zara Trousk wrote:
I've tried all versions (stable
I tougth about using lvs and balancing incoming traffing.. but never
mosix..
On Wed, 2 Apr 2003, Liaan van der Merwe wrote:
Hallo all
Just for interest sakes..
Have any of you tried to run * on a openmosix cluster?? Could be nice for
ppl like me that dont have the resources to buy proper
I have done exactly taht usingdigium x100p cards... :)
Very nice indeed...
On Sun, 30 Mar 2003, Richard Scobie wrote:
I am hoping to implement an asterisk system between two offices as below
|---|
| PBX |-(XXX)--[asterisk]---[VPN][asterisk]--(FXO)--phone
|---|
Is a relive that digium is making the fxs cards...
But what about fxo cards?
here in mexico I have the trouble of e1 r2 modified for mexico signaling
type..
So, one solution are fxo cards, fxs was linuxjack field or cisco's ata
units...
Now, the cost is now over the fxo cards, to make
Does anyone over here use spamassasin, I belive the * list messages get
treated as spam...
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
How is the behaivor of the snom phones with asterisk?
Im thinking of getting 40 snom phones, to manage a voip solution...
How's the sound quality?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
41 matches
Mail list logo
