Hi list.
I want to know if there is any console o secretarial hardphone that
works with asterisks.
I mean a phone in witch i can see the state of the extensions, the
phone lineas, etc. Can hold o transfer easly a call, etc.
Thanks
Alvaro Parres
send a tone, becouse when i push it i
lost the dial tone.
Any idea how can i do, so * detect that tone as flash key ?
Alvaro Parres
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Hi:
Any of you know a good web admin for asterisk???
Thanks
Alvaro Parres
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http
and connect this
with a Normal PBX (Nortell), so that the actual telephone system can dial to
the Asterisk and this system can recive call form Asterisks.
Alvaro Parres
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Hi...
Some questions.
¿How do you make that some user who is in a menu, can dial any extension
that is define in other context ? Example..
[office]
100,1..
200,1..
300,1..
[menu]
s,1 - When the user is here.. can dial 200 and it
Hi...
I have the next problem.. I have a FXO card with i can make calls but i cant
recive calls.
At the consol, i get the next error:
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Zap/2-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/2-1
Hi all.
I have some questions:
1) Is there a way to get a full log of the calls (incoming and outgoing)
2) How is the intregation of Mysql and Asterisk. At witch Aplicattions.
3) And of the Extension
a) I have a Support Call Center.
Hi
I'm having a extrange problem I cant register with Iaxtel or call to digium...
But i cant make or recive IAX calls... ( I made some one with irc users )
Any idea why?
At my logs i have this from iaxtel:
NOTICE[196621]: File chan_iax2.c, Line 2832 (register_verify): No
Hi:
I have the next problem at VoiceMail application, when i call is
recive, it does not detect when the user hangup the call..
The first solution i think was to place the busydetect=yes at
zapata.conf but it cause that some calls get lost ( when some one is
talking, the * hangup the
Hi all...
How can i register wit nufone i was serching at its pages... and
I never find how to get register...
Thanks.
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Hi... some one can tell me his comments about VoicePulse Services...
for Pre-Paid long distance..
Alvaro parres
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Hi..
I'm having the next problem... with the busy detect = yes...
If i have it... The * it hang up the calls when they are
active... ( Incoming ant Outgoings calls).
If i haven't it * Doesn't detect when some one hang up and
never close the channels...
WHAT I CAN DO
Hi.. i'm having the next problem with a Asterisks Box... like every 24 hours it
give the next error when i tray to connect to it.
[EMAIL PROTECTED]:~# asterisk -r
Asterisk CVS-09/29/03-17:13:53, Copyright (C) 1999-2001 Linux Support Services, Inc.
Written by Mark Spencer [EMAIL PROTECTED]
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi some one can give me information about a good and ship ip phone IAX
or SIP
Thanks
- --
Alvaro Ivan Parres Peredo
Director de IT
[EMAIL PROTECTED]
Tel: (33) 36301294
~ (33) 36309553
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi
~ Some one know where i can find some documentation about how to
programm some
modules for asterisk.
~ Becouse i want to program a call limit per user.
- --
Alvaro Ivan Parres Peredo
Director de IT
[EMAIL PROTECTED]
Tel: (33) 36301294
~
=
| 999,3,Answer exten = 999,4,MusicOnHold(default) exten =
| 999,103,Busy
|
| See?
|
| You can limit that to just 1 user at a time or what ever you wish :
|
|
| bkw
|
| -Original Message- From:
| [EMAIL PROTECTED] [mailto:asterisk-users-
| [EMAIL PROTECTED] On Behalf Of Alvaro Parres Sent
tried RxFax... let me know if you need any extra
help with that. Sounds interesting
On 6/8/07, Alvaro Parres [EMAIL PROTECTED] wrote:
Moy:
I have working an Asterisk 1.4.4 with Unicall rn MFR2. The only
problem
i have is the RxFAX application, that broke every time... With and error
us whit the case where no caller id is send.
On 7/19/07, Carlos Chavez [EMAIL PROTECTED] wrote:
*On Thu, 19 Jul 2007 12:14:53 -0500, Alvaro Parres wrote*
Yes Moises, i was looking for it.
The main problem is only on the files for version 1.4... it give
that error when no CallerID
:
Alvaro, can you post the patch in a public place and post the URL
here? It might be a good idea to contact steve underwood to see what
he has to say about such a patch.
Regards,
On 7/18/07, Alvaro Parres [EMAIL PROTECTED] wrote:
Carlos:
Only for check do this change:
protocolvariant=mx,10,4
= R2_SIGI_12; to the
Mexico Definition we can recive restricted Caller ID calls, but no Not
Caller ID Calls.
I'm not shure why only on versions 1.4 and not on version 1.2.
Thanks.
On 7/22/07, Steve Underwood [EMAIL PROTECTED] wrote:
Alvaro Parres wrote:
Search at mfcr2.c this:
case
is to disable CallerID by setting ANI to 0. Right now I
have to use:
protocolvariant=mx,0,4
I am having the same problem on E1 service from Telmex and
Avantel.
Obviously customers want CallerID on their phones.
Thanks a lot.
On 7/23/07, Alvaro Parres [EMAIL PROTECTED] wrote:
Steve
Hello List:
I have publish a tar.gz file with an Asterisk 1.4.13 correctly patched
for compile the chan_unicall and the apps rxfax and txfax. This tar.gz file
also contain all the necesary library for work.
http://arabe.com.mx/blog/?p=10
This tar was based on the one publish by Moy at
(
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3056375835category=11908
)
I think i can connect this one to a T100P Card .. Is this
possible, and correct?.
And one more which one are good IP PHONE for *, where i can buy it?
Well Thanks.
Alvaro Parres
I have ZAP channels.. so i add the lines at zapata.conf
and it does not work. When i dial *8 it return me a busy tone.
my zapta.conf is..context=home
group=2
pickupgroup=2
signalling=fxo_ks
channel=2-3
callerid=FIJO 200
channel=3
callerid=INALAMBRICO 100
channel=2
Rich Adamson wrote:
Is it
Hi list.
I'm having the next problem.
I have a * with 1 TDM400P (4 ports) and
one X100P, with a working configuration.
Today i add one more X100P card, and i change
the config files as next:
zapatel.conf:
fxsks=1-2
fxoks=3-6
loadzone = us
Hi, i have the next problem:
I have a new Motorola cordless analog phone plug at FXS ports at my
*. But the * does not detect when i press flash?...
Any idea how can i solve this?
Is any way i can set another key to work as flash? maybe # or * ??
Thanks
Only this phone have the problem. I have other phones and they
dosent have the problem
Brian West wrote:
Do any other phones work fine? Or just this one?
bkw
On Tue, 16 Dec 2003, Alvaro Parres wrote:
Hi, i have the next problem:
I have a new Motorola cordless analog phone plug
i have check at internet, that some one use RxFax application for
recive faxes...
Where i can get this application, becouse i have the cvs of today
and it does not have application???
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Mariano:
Could you send us please the log files, and the console output... so we
can help you.
On Mon, Jun 9, 2008 at 8:01 AM, Mariano Borgognone
[EMAIL PROTECTED] wrote:
Moises, we've already set debug level at 255 on unicall.conf and at
logger.conf we've enabled full log
Hi list:
Is there any way, to set a common inter-queues leastrecent Strategy, i'm
searching a Behaviour like this:
2 Queues Q1 and Q2
2 Agentes A1 and A2
Both agents are in both queues.
First Call in the system is for Q1 and is answer by
Grate job Moy... i will test it on my PBX tomorrow...
Thanks.
On 4/20/07, Moises Silva [EMAIL PROTECTED] wrote:
Thanks a lot for the fix Humberto.
On 4/18/07, Humberto Figuera [EMAIL PROTECTED] wrote:
Hi Moises,
the Asterisk SVN-branch-1.4-r60989 make a change in the
ast_channel_alloc
Moy:
I have working an Asterisk 1.4.4 with Unicall rn MFR2. The only problem
i have is the RxFAX application, that broke every time... With and error in
the linking to the spandsp library.
If i have time this weekend i will review to fix the app,
Thanks.
On 6/4/07, Tobias Wolf [EMAIL
Hi list:
Is there any way or an idea of how to made a global queue policy. I need
to have a Global Policy or a common policy to many queues.
What i need is:
I have 20 agents they are members of 5 queues, i have a last recent
strategy for all the queues, the problem is that the
Hi list:
I'm having the next problem, it appear that the application ChanIsAvail
is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS.
I add my dialplan and the output to the cli.
THanks.
In the example i'm dialing from extension SIP/112
My DialPlan Secction:
Jared:
As you see i have the s option. That works fine on Version 1.2. Let me
see config the call limit con sip channels it works.
Thanks.
On 6/25/07, Jared Smith [EMAIL PROTECTED] wrote:
On 6/25/07, Alvaro Parres [EMAIL PROTECTED] wrote:
I'm having the next problem, it appear
Carlos:
I think you problem is on the queue systems, we have the same problem
on version 1.2.x and 1.4.x
one one of our call centers.
Try to change you agents to be dynamic, and also to change the login
method from AgentLogin to AgentCallBackLogin
Alvaro
On 6/8/07, Carlos Chavez
Hi list:
I want to modify the libmfcr2. But i can't find where is define the end
DNIS signal is define. Actually the libmfcr2 send a ONE (1) at the end of
sending all the DNIS numbers. I need to send a TWO (2), this becouse in
Mexico the normal is to send a 2 at the end, not a ONE.
In the
Could you send please your unicall.conf file
Thanks.
It appers to be a problem with de ANI digits you want to recive.
On 7/17/07, Carlos Chavez [EMAIL PROTECTED] wrote:
On Tue, 2007-07-17 at 19:30 -0500, Moises Silva wrote:
In order to help you I need testcall traces, with max level of
labortario y en el extrangero tuvimos esos
problemas, si es asi te paso un parche solo que lo encuentre para la
libmfcr.c Donde le digas como manejar la señal de private al recibir el ANI.
Saludos.
On 7/18/07, Carlos Chavez [EMAIL PROTECTED] wrote:
On Wed, 2007-07-18 at 08:10 -0500, Alvaro Parres
Hi list, i'm trying to do that iax channels can acces the pickup
feature(normaly *8 dialing).
But always the iax channel when dial *8, search for the extensión *8 on its
context.
I know i can program the *8 extension with the pickup applicatión. But its
doesn't works for me, becouse i need to
I had set it
On 3/21/07, LKS GMAIL [EMAIL PROTECTED] wrote:
Try to set the callgroup and pickupgroup up in the IAX conf.
Saludos, Lukassky.
--
*De:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *En nombre de *Alvaro Parres
*Enviado el:* miércoles, 21 de
Hi List: I have the next diagram: GSM G729 G729 IdeFisk -- Asterisk A - [INTERNET] Asterisk B - PSTN ( Via Unicall / Zap )
The user at IdeFisk Login as Agents on Asterisk B at this moment we have the next Licence Use: A) 1/1 B) 1/0 When a Call
Yes i'm recording...On 10/8/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
Alvaro Parres wrote: Hi List: I have the next diagram: GSM G729 G729IdeFisk -- Asterisk A - [INTERNET] Asterisk B - PSTN ( Via Unicall / Zap )
The user at IdeFisk Login as Agents
model of PRI is suggest for this ?
2) Some one have already do this ?
3) Is there form of correct de AMI problem ?
Well i hope that you will answered me.
Alvaro Parres
P.D. If any one from Mexica have done this before pleas contact me
(33) 35636261
??? i dont understand.
On 8/5/05, Siegel, Joerg [EMAIL PROTECTED] wrote:
Ich bin am 9.8. wieder im Hause!
Mit freundlichen Grüßen,
Jörg Siegel.
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card, is this correct
On 8/5/05, Paul Belanger [EMAIL PROTECTED] wrote:
Hello,
See comments inline
Alvaro Parres wrote:
Hi list:
I have a client that needs to connect a Asterisk PBX with a TE110P
of Digium and one Nortel Option 11.
Actually the Nortel Option 11 have
I'm using SPA 841 form SIPURA and they work very nice, and are cheep
only 80 USD...
and all the options, (transfer, DND, conference, etc) work nice.
On 8/5/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hard phones.
Varun
- Original Message -
From: Jason Walker [EMAIL PROTECTED]
We have been using SIPURA and have no problem. With the last firmware
and silence supression off.
On 8/7/05, Paul Dugas [EMAIL PROTECTED] wrote:
On Sun, August 7, 2005 1:15 pm, Thierry Wehr said:
This is not true
You have to switch to last firmware and/or disable silent suppression
I
Jonathan:
Our provider continue selleing us SPA-841, if you want the contact,
mail me outside the list.
On 8/13/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
Tom Rymes wrote:
Chris,
Maybe you could write a generic config file and post it to the wiki?
I tried to post as a
Wich kind of E1 card do you use at the NORTEL ??
it was a PRI one??? witch model ???
On 8/12/05, Mark Phillips [EMAIL PROTECTED] wrote:
Easily doable. I've done it twice now. Problem is that your users will
never know they have messages waiting.
Install a T1/E1 card into the * box and then
Hi list, i have and asterisk into a Pentium IV Server with 1GB of RAMabout 75 Polycom Phones, one E1 for incoming calls.We have program a page system with the page command and the auto answer funtionof polycom.
We have detect via diaplan if the phone isn't in call we place the call. All this via
that is a memory problem ?On 3/23/06,
BJ Weschke [EMAIL PROTECTED] wrote:
On 3/23/06, Alvaro Parres [EMAIL PROTECTED] wrote: Hi list, i have and asterisk into a Pentium IV Server with 1GB of RAM about 75 Polycom Phones, one E1 for incoming calls.
We have program a page system with the page command
that is a memory problem ?On 3/23/06,
BJ Weschke [EMAIL PROTECTED] wrote:
On 3/23/06, Alvaro Parres [EMAIL PROTECTED] wrote: Hi list, i have and asterisk into a Pentium IV Server with 1GB of RAM
about 75 Polycom Phones, one E1 for incoming calls.
We have program a page system with the page command
I have more than 100 users with out problemand i'm using the file no db.On 3/23/06, Antonio Rabena
[EMAIL PROTECTED] wrote:You can try using asterisk-addons
http://www.voip-info.org/wiki/view/Asterisk+voicemail+database orasterisk
I think here are 2 mixed subject One Substitution of the GXP 3000 Video Phone Phone with a great speaker phone. For the second Subject a think Polycom are the greatest.
On 6/13/06, Steve Underwood [EMAIL PROTECTED] wrote:
Mike Fedyk wrote: Or any polycom phone that has speakerphone like the IP501
Hi list, i have the next situation
I've a asterisk connect with a Nortel Meridian Op 11, via a PRI CARD with 5ess switch
[Asterisk] -- PRI - NET --- PRI- CPE --[Nortell]
I can call from Asterisk to Nortell with no problem, but when Nortell
place a call to me, i have the channel bridge
Hi list, i have the next situation:
B
[HT486] -- (NAT/ROUTER) INTERNET - [* server]
|
|
|
A|
[HT486] -
Both HT486 register to * server, with no problem, but when they call each other
the voice only goes from B to A but not from A to B.
My
Hi list
I have a problem on a PRI E1 card.
The connection diagramis:
[ASTERISK] -- PRI-NET -- PRI-CPE -- [NORTELL]
The problem is:
When i made a call in channels 17 to 31, there is no voice in any way...
but on channels 1 to 15 i have no problems...
Any
Hi list
I have a problem on a PRI E1 card.
The connection diagramis:
[ASTERISK] -- PRI-NET -- PRI-CPE -- [NORTELL]
The problem is:
When i made a call in channels 17 to 31, there is no voice in any way...
but on channels 1 to 15 i have no problems...
Any
Hi list
I have a problem on a PRI E1 card.
The connection diagramis:
[ASTERISK] -- PRI-NET -- PRI-CPE -- [NORTELL]
The problem is:
When i made a call in channels 17 to 31, there is no voice in any way...
but on channels 1 to 15 i have no problems...
Any
26 and Nortel uses channel 25. Thiscan be modified on at least QSIG trunks. But on EuroISDN thereshould not be a problem.
Hans[EMAIL PROTECTED] schrieb: On Wed, 2 Nov 2005, Alvaro Parres wrote:Hi listI have a problem on a PRI E1 card.
The connection diagram is:[ASTERISK] -- PRI-NET
. Asterisk uses channel 26 and Nortel uses channel 25. Thiscan be modified on at least QSIG trunks. But on EuroISDN thereshould not be a problem.
Hans[EMAIL PROTECTED] schrieb: On Wed, 2 Nov 2005, Alvaro Parres wrote:Hi listI have a problem on a PRI E1 card.
The connection diagram is:[ASTERISK] -- PRI-NET
at the console i send a show hints i get this:
-= Registered Asterisk Dial Plan Hints =-
112
:
SIP/112
State:Unknown
Watchers 1
- 1 hints registered
any idea ???
Thanks.
ALvaro Parres.
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Thanks with the upgrade they work... Now i only have one problem.
I create 3 hints.. (111 (SIP/111), 112 (SIP/112), and 102 (ZAP/35) )
the SIP/111 is a GrandStream ATA
the SIP/112 is a Polycom 301
the ZAP/35 is a Analogic Phone.
The SIP/112 hints works great. But the other 2 no.
The ZAP/35
But M play Hold Music.
And what we need, as the other to users ask, is to play a especific file while the phone is rinning.On 11/10/05,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
You can play music instead of providing a ringtone. ( I think it's the Moption for the dial command)We used this for
Hi does any one have the Polycom SoundStation IP 4000 files for FTP
becouse my SS IP 4000 take the files of my IP SoundPoint 301 and 501...
and it apper to be no the sames.
Thanks..
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I think i only need the sip.ld file..
Becouse when the SS is booting it said that the sip.ld is not for that phone.
I have actually the sip.ld 1.6.x version.
thanks.On 11/10/05, Alvaro Parres [EMAIL PROTECTED] wrote:
Hi does any one have the Polycom SoundStation IP 4000 files for FTP
becouse
Hi list, i have the next problem:
I create 3 hints.. (111 (SIP/111), 112 (SIP/112), and 102 (ZAP/35) )
the SIP/111 is a GrandStream ATA
the SIP/112 is a Polycom 301
the ZAP/35 is a Analogic Phone.
The SIP/112 hints works great. But the other 2 no.
The ZAP/35 is say is always in USE
Hi does any one have the sip.ld file of a SoundStatios IP 4000
Thanks.
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Hi list, i have the next problem:
I have conifgured hint for all my extension ( SIP and ZAP) but at the console
i send show hints and always all the channels are idle..
My config files:
at extension.conf
...
[sip-test]
exten = 101,hint,ZAP/35
exten = 101,1,Dial(ZAP/35)
exten =
Hola lista:
Para todos los de Mexico, desde hace aproximidamente 2 meses, estoy teniendo problemas con
la trasmicion de voz via protocolo SIP. Entre Infinitums o desde Infinitum hacia otros provedores.
Alguien mas esta experimentando esto ??? par poder corroborar que Telmex este bloqueando el
Esto lo hemos estado detectando en la ciudad de Guadalajara
On 11/15/05, Servers-R-Us [EMAIL PROTECTED] wrote:
Hola Alvaro,En qué parte de México tienes * y desde qué partes de México te conectas?Nosotros tenemos clientes en BCS, Oaxaca, Puebla, Morelos y el DF que se
conecan con Infinitum y no
How do you enable the parked soft key
On 11/22/05, Noah Miller [EMAIL PROTECTED] wrote:
Hi Anthony - Instead of asking a question, I thought I'd post an answer. I got the Polycom IP501 'Park' softkey working with * by doing the following:
You are my favorite person today!This rocks, and
i have the 1.6.3 firmware and also when i press park i need to dial another extension..
On 11/24/05, Adam Goryachev [EMAIL PROTECTED] wrote:
What firmware version did you use for the polycom phone ??I just tried it on my IP600, and when I press the park button, it waitsfor me to dial an extension
Hi... I have the polycom 301 with firmware 1.6.3
When i Press Park, i get a dialog to enter a extension.
A dial 700 ther
and the call get parked, and i recive a call announceme where the calls was parked.
is this normal ???
On 11/24/05, Alvaro Parres [EMAIL PROTECTED] wrote:
i have the 1.6.3
we have TFTP and also those files are created upgrade automatic.. And
also we create manually the file for the new phones so they have
the minimal addres book of the company.
On 11/25/05, Watkins, Bradley [EMAIL PROTECTED] wrote:
Hrmmm... I'm not sure how much more help I can be on this
Serach in the list, about 1 o 2 weeks ago.. there is a guide for how to setup the key with asterisk
On 11/25/05, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
But more importantly, what would you do with it if you found it?Hasanybody made this softkey interface with Asterisk's parking
Josheph:
I had have that problem, and it get solve when i take out the incominglimit from my sip.cfg
Also if you send you sip.cfg and extensions.cfg will be easier to help you
Tray it.
Alvaro Parres
On 11/28/05, BJ Weschke [EMAIL PROTECTED] wrote:
On 11/28/05, Kevin Hanson [EMAIL PROTECTED
Hi list...
I have been testing the hint extension. And i detect
that when i have in the sip.cfg of the extension the
incominiglimit=X (any number) the hint doesn't work all the
time show the extesion as idle.
If this is a bug or not ??
Thanks.
Hi, i have one Asterisk with a Digium E1 card, and a Meridian Nortel
Release 11.
I need to connect both of them. We are using MFC/R2 for this..
The Diagram:
[ NORTEL ] ( AMI )
(DIGIUM) [ ASTERISK]
we have green light at the digium card, and at
Hi, i have one Asterisk with a Digium E1 card, and a Meridian Nortel
Release 11.
I need to connect both of them. We are using MFC/R2 for this..
The Diagram:
[ NORTEL ] ( AMI )
(DIGIUM) [ ASTERISK]
we have green light at the digium card, and at
In witch part... at Nortell or at Asterisk ??
and how to do this ?
On 8/29/05, Jerry Geis [EMAIL PROTECTED] wrote:
Sir,Can you turn off Multipart SDP headers?That was the problem I had.jerry___--Bandwidth and Colocation sponsored by
Easynews.com
I have one GXP-2000 and i prefer the SPA 841 of SIPURA, (look their are same price) y the price is not a problem POLYCOM or SNOM
On 9/1/05, Joe McConnaughey [EMAIL PROTECTED] wrote:
Check out the Aastra 9133i. Fantastic phone for about $179. I have two of them and will be adding more. More bang
Hi list, any one can let me his config files for interconecting a Meridian Op 11 and Asterisk
via a E1 PRI CARD.
Actually i need the nortell config part, becouse my client nortell provider doesn't know
how to config the PRI card at his part.
Thanks all.
Yes it works, the only thing is that you need to patch you asterisk for support R2
On 9/23/05, Alex Kauffmann [EMAIL PROTECTED] wrote:
We have several in operation but with isdn and not R2. I know I've seen emails from people that use them with Telmex and have them operating, albeit with some
Yes with version 1.2. I have tried already with call-limit and the same.
On 11/28/05, Kevin Hanson [EMAIL PROTECTED] wrote:
Alvaro Parres wrote: Hi list...I have been testing the hint extension. And i detect
that when i have in the sip.fg of the extension the incominiglimit=X (any number
(ENGLISH VERSION AT THE END)
Hola lista:
Requiero saber si alguien tiene un cliente o empresa donde se encuentren montado algun
Asterisk como PBX de tamaño mediano (al menos unas 50 extensiones). Esto para dar una
demostracion a un cliente mio que esta interesado en invertir en Asterisk.
Les
Could you send it patch please.
On 11/30/05, Paradise Dove [EMAIL PROTECTED] wrote:
btw, i've patched this part of code and now its working fine for me.i'm going to upload it.Paradise Dove
On 11/30/05, Kevin Hanson [EMAIL PROTECTED] wrote: Paradise Dove wrote: Yes with version 1.2. I have tried
Hi list:
I'm having problem with some DIAXY ATA FROM DIGIUM, I have 3 of them in different points, all of them register
to a central asterisk server. If i call from any of the ATA's to Asterisk or Asterisk's to ATAs. But when any ATA's want to talk
to another ATA's.. TheATA's rings, but when the
PROTECTED]/5'
Any IDEA ??
On 12/2/05, Alvaro Parres [EMAIL PROTECTED] wrote:
Hi list:
I'm having problem with some DIAXY ATA FROM DIGIUM, I have 3 of them in different points, all of them register
to a central asterisk server. If i call from any of the ATA's to Asterisk or Asterisk's to ATAs
, *in } break;
default: ast_log(LOG_ERROR, update_call_counter(%s, %d) calledwith no event!\n, name, event); } // paradise dove if (p) ASTOBJ_UNREF(p,sip_destroy_peer);
else if (u) ASTOBJ_UNREF(u,sip_destroy_user); return 0;}Paradise DoveOn 12/2/05, Alvaro Parres [EMAIL PROTECTED] wrote: Could you send
Why using SIP instead of IAX2 ???
Only a question becouse i always use IAX
On 12/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
Well... not so perfectly.What I'm experiencing is that during certain call volumes, many calls
go thru from box1 to box2. However, there are some cases where I
Hi list..
I want to make Snom 360 and 30 to autoanswer so i can have a paging sistem.
I tried tu send intercom=true with the little patch to chan_sip.c and it didm't work
any one have and idea of ow to do this.
___
--Bandwidth and Colocation
Hi any one can recommend me a company in the USA that can sell me a Toll Free Number
and send me the call via IP.
Thanks.
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Jorge nosotros tenemos problemas de fax solo con lineas de telemex con
lineas de axtel no. Y no hemos podido tampoco detectar el problemas.
SI logras resolverlas hasnolo saber porfabor.
On 12/27/05, Martinez Felix [EMAIL PROTECTED] wrote:
no podria decirte, porqe tengo problemas con los scripts
Hi list
I'm havining problems with a E1 Digium Card (TE120) here are the
description of the problem:
Case 1: Zaptel 1.4.8 Kernel 2.6.22
The system start working correctly, but aleatory the asterisk
prosess give a kernel panic with the messages:
Process
Hi list:
I'm having problems, with a AEX800P card when plugin on HP ML115 G5
Server, when i load the mdule (wctdm24xxp), it loads with error con
dmesg, that say that is a kernel bug of invalid OPCODE 000 or
something like that.
If i plug the card on SUN Server, i don't have problems with
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