Massimo Nuvoli wrote:
Also you can use the unstable branch of debian, all things are near
ok, from the asterisk core to the kernel.
Bye
It may have been 2 years since I worked with Debian on production
systems, but in my experience there are alot of unstable packages in
unstable. So it's a
On Tue, 30 May 2006, olivier.taylor wrote:
Hi all,
I need your lights :)
There are many hardware provider for E1 cards on the market, what's your
exeperience with E1 and what's the preferred provider for Asterisk out of
Digium?
I prefer Eicon Diva Server cards, they have good features
In case anyone is interested, I have a Dialogic D/600JCT-2E1-120 that we paid
about A$15K for not so long ago. I am open to any serious offers.
--
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Also you can use the unstable branch of debian, all things are near
ok, from the asterisk core to the kernel.
Bye
It may have been 2 years since I worked with Debian on production
systems, but in my experience there are alot of unstable packages in
unstable. So it's a bad advice to run
On Tue, 2006-05-30 at 08:07 +0200, Attilla de Groot wrote:
It may have been 2 years since I worked with Debian on production
systems, but in my experience there are alot of unstable packages in
unstable. So it's a bad advice to run unstable on production systems.
the debian stable, testing,
I have been able to figure out the first part of my problem.
I wrote my scripts based on wrong assumptions. one of which was that the
line
exten = s,1,AGI(xyz.agi)
sends an undefined extension value to the script. This is definitely wrong.
This line actually sends an extension value of s.
Can't you use mpg123 as compiled under x86_32? I do on a few servers I
have. I found madplay better process wise than mpg123.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick
Perez
Sent: 29 May 2006 21:37
To: Asterisk Users Mailing
Hi, I have two asterisk machines side by side both running debian sarge, one running sarge's version of asterisk (1.0.7.dfsg.1-2) and the other running the version of asterisk from www.backports.org (1:1.2.1.dfsg-2bpo1). I also have a SIP provider who is routing blocks of DID's to both machines.
On 5/30/06, Stefan Reuter [EMAIL PROTECTED] wrote:
Are there any apt repositories which provide newer versions of the
software?
sure: http://pkg-voip.buildserver.net/debian
Hi Stefan, very nice. A related question, is there any way you could
share the process of how to create the asterisk
Can MAD crash a server like mpg123 can?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer
Sent: Tuesday, 30 May 2006 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] mpg123 or asterisk
Can't
I don't know as I never experienced any system crashes with either. But
I did notice many times that mpg123 was still running after asterisk had
been shutdown.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of MBIT
Technologies
Sent: 30 May 2006
I'm interested too to know about a quad E1 card...
I need to connect it to 2 differents ISDN providers in Europe and to
establish a third connection
with a Matra PBX.
The server ( IBM XSeries 346 ) has to serve about 60-70 simultaneous
calls ( IVR and max 30 conferences... )
I will also
autocreatepeer=yes
[ser_box1]
type=peer
username=ser_box1
insecure=yes
canreinvite=no
context=from-internal
host=ip.address.of.box
nat=yes
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
It doesn't work for me :-(
How do you have the peer configuration in asterisk, to connect ot
On Tue, 30 May 2006, Tristan wrote:
I'm interested too to know about a quad E1 card...
I need to connect it to 2 differents ISDN providers in Europe and to establish
a third connection
with a Matra PBX.
The server ( IBM XSeries 346 ) has to serve about 60-70 simultaneous calls (
IVR and
Hans Witvliet wrote:
Some consultants are not very keen on such boards, as ad/da are done in
software instead of hardare.
software DAC, LOL
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or
seems to be very good cards, but also, very expensive, isn't it?
Olivier
Armin Schindler a écrit :
On Tue, 30 May 2006, Tristan wrote:
I'm interested too to know about a quad E1 card...
I need to connect it to 2 differents ISDN providers in Europe and to establish
a third
Hi, (I tried to send this to the list earlier, it didn't seem to work- my apologies if you see this twice...)I have two asterisk machines side by side both running debian sarge, one running sarge's version of asterisk (1.0.7.dfsg.1-2) and the other running the version of asterisk from
It has crashed an SGI Altix 350 on a dialy basis.
MBIT Technologies wrote:
Can MAD crash a server like mpg123 can?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer
Sent: Tuesday, 30 May 2006 5:06 PM
To: Asterisk Users Mailing List -
On Tue, 30 May 2006, olivier.taylor wrote:
seems to be very good cards, but also, very expensive, isn't it?
these cards are more expensive than passive cards or other cards with less
features of course. But these cards are really very powerful and when I get
feedback from users, I always here:
Seems like there is no quad e1 diva server cards...
Does someone knows about digium and sangoma ?
Tristan
olivier.taylor a écrit :
seems to be very good cards, but also, very expensive, isn't it?
Olivier
Armin Schindler a écrit :
On Tue, 30 May 2006, Tristan wrote:
can someone overthere help?the server specs are as follows
HP DL380G4 Dual Intel Xeon 3.2GHz processor with 4GB RAM,
running fedora core 3
asterisk-1.2.5
ss7-0.8.3d.
using sip as advised to receive calls from another gateway in US.
using g729 in transcoding way.
however, I noticed the call hit
Sorry, asked the wrong provider, I just looked at the official website,
I was wrong...
Tristan a écrit :
Seems like there is no quad e1 diva server cards...
Does someone knows about digium and sangoma ?
Tristan
olivier.taylor a écrit :
seems to be very good cards, but
Hi All My company has vacancy for asterisk professional We are an INDIAN base comapny if any one interested please send mail your resume to [EMAIL PROTECTED] with number subject line -- Asterisk Resume (yrs of exp) DONT MAIL YOUR RESUME ON THIS MAIL ID
Ring'em or ping'em. Make PC-to-phone
Hi all I fancied playing with SER and * on the same box. So i thought
i'd just change the default sip port for * in sip.conf
[general]
port = 5065 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
restarted * and now when i issue a
Hi!
Probably any of you meet with the following problem:
asterisk is restarting in a minute (if no active call) if active call,
it says cannot receive a call due to restart in progress.
even if i starting with -c, i have no disconnected, but see the stuff
restarting.
i've tried to recompile,
Tristan schrieb:
Does someone knows about digium and sangoma ?
A lot ;-) What do you want to know?
Christian
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
bails wrote:
Hi all I fancied playing with SER and * on the same box. So i thought
i'd just change the default sip port for * in sip.conf
[general]
port = 5065 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
restarted * and
Tristan schrieb:
What would you recommend ?
Digium TE411P, Sangoma A104D, Eicon Diva Cards ?
Ah - I should have read this bevor my last answer. ;-)
I personally prefer the Sangoma E1 cards. The work in almost every PCI
system and the echo cancel - if you really need it - is far better than
Hi All My company has vacancy for asterisk professional We are an INDIAN base comapny if any one interested please send mail your resume to [EMAIL PROTECTED] or [EMAIL PROTECTED] with number subject line -- Asterisk Resume (yrs of
exp) DONT MAIL YOUR RESUME ON THIS MAIL ID SHOBHIT
I need different opinions about these cards to be sure about the one to
buy
because the server must be up 24/24...
What would you recommand for my needs ?
I need to connect the card to 2 differents ISDN providers in Europe (
EURO-ISDN ) and to connect also with a Matra PBX ( Maybe QSIG ).
I
Hello Joshua,
Joshua Colp wrote:
[portunity-out]
type=friend
host=iax.iaxport.de
username=XXX
secret=YY
context=incoming-portunity
notransfer=yes
Only if the username is specified as portunity-out when the other side dials
you. Otherwise your Asterisk has no idea what to
On Tue, 30 May 2006, Christian Victor wrote:
Tristan schrieb:
What would you recommend ?
Digium TE411P, Sangoma A104D, Eicon Diva Cards ?
Ah - I should have read this bevor my last answer. ;-)
I personally prefer the Sangoma E1 cards. The work in almost every PCI
sorry to ask that,
hi, Longing for your help.I came into a problem ,Now I want to configure asterisk sip peers from MYSQL database dynamic, flolling the introduction of asterisk realtime,i set the cofiguration of sip users,but I need to cofigure sip peers too. Where I can find some infomation about cofiguring sip
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
hi,
Longing for your help.
I came into a problem ,Now I want to configure asterisk sip
peers from MYSQL database dynamic, flolling the introduction of
asterisk
realtime,i set the cofiguration of sip users,but I need
Tristan [EMAIL PROTECTED] writes:
This server ( an IBM X-Series 346 dual 3Ghz Xeon with 2gb ram ) has to
serve about 60-70 incoming/outgoing PSTN simultaneous calls ( IVR and
max 30-40 conferences... ) and about 10-20 SIP calls to begin...
I tried the Quad-port Digium cards in this special
Wolfgang: What kind of CPU-load issue on A104? Could you give me a link or
something? I also use A104 and it works very good, but recently I noticed a
behavior which is maybe connected with this issue, so more info would be very
helpful for me :)
Thanks!
Alex
-Original Message-
From:
* Christian Victor [EMAIL PROTECTED] wrote:
Tristan schrieb:
What would you recommend ?
Digium TE411P, Sangoma A104D, Eicon Diva Cards ?
Ah - I should have read this bevor my last answer. ;-)
I personally prefer the Sangoma E1 cards. The work in almost every PCI
system and the echo
Hi Alex,
Asterisk [EMAIL PROTECTED] writes:
Wolfgang: What kind of CPU-load issue on A104? Could you give me a
link or something? I also use A104 and it works very good, but
recently I noticed a behavior which is maybe connected with this
issue, so more info would be very helpful for me :)
The place I currently work at has a Panasonic Key system
with 9 extensions, and no voicemail. It services 2 PSTN lines.
I am hoping to use Asterisk to host voicemail (I would like
to use the IVR also, but I dont even know if or how it would work).
Do I need to use a PRI between the
The place I currently work at has a Panasonic Key system with 9 extensions,
and no voicemail. It services 2 PSTN lines.
I am hoping to use Asterisk to host voicemail (I would like to use the IVR
also, but I don't even know if or how it would work).
Do I need to use a PRI
Thanks a lot Wolfgang!
I use Dell PE2850, so probably this issue does not directly affect my system.
But I will read thread anyway.
What is happening is that under higher load (60 calls, for example) the
Asterisk sometimes stops responding in a time manner (AMI messages are delayed,
etc).
I will give it a try, thank you.
Do you get ringing and on the phone statuses on the subscribed
extensions? What kind of phones are you using?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jason Bachman
Sent: Friday, May 26, 2006
I run ngrep when I call an IP number and when I call a PSTN number,
and the sequece is like that:
For PSTN Numbers:
Sipura --- Asterisk (Invite pstn number)
Asterisk---Sipura(407 Proxy Auth. Required)
Sipura --- Asterisk (Ack)
Sipura --- Asterisk (Invite with Proxy Auth.)
Asterisk---Sipura
Regarding echo cancel. Is there someone with hands-on experience
regarding the echo canceller performance of the Junghanns E1 cards
compared to for example the Sangoma ones?
Well - the Junghanns does the echocancel in software and the Sangoma
A104d does it in hardware.
So on the Sangoma echo
Hi,
as already said in others messages on this list, I'm rewriting my dialplan
using AMP/FreePBX as starting point.
I saw that AMP/FreePBX uses the concept of USERS/DEVICES, quite interesting
but not useful to me now. It defines USERS/DEVICES association in AstDB and
then uses dialparties.agi
Hi Armin,
I personally prefer the Sangoma E1 cards. The work in almost every PCI
sorry to ask that, but what does almost every PCI system mean?
First of all compared to the Digium TE4xx the Sangomas work in 3,3V and
5V PCI slots. That means they run in every PCI slot but PCIexpress.
In
Very well... However, if you don't send your dial plan, we don't know what
it is you are trying to do.
Obviously make sure you have set the dtmf to rfc2833 in your sip.conf.
bp
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Akpome
Akpoguma
Sent:
* Christian Victor [EMAIL PROTECTED] wrote:
Regarding echo cancel. Is there someone with hands-on experience
regarding the echo canceller performance of the Junghanns E1 cards
compared to for example the Sangoma ones?
Well - the Junghanns does the echocancel in software and the Sangoma
Hello,
I have a Gigaset S44 connected to a quadBRI NT port. Receiving calls
works phone, however when dialing out from the phone the call is dropped
to the 's' extension, as if no extension had been dialed:
-- Accepting voice call from '492389990' to 's' on channel 0/2, span 4
On Tue, 30 May 2006, Christian Victor wrote:
Hi Armin,
I personally prefer the Sangoma E1 cards. The work in almost every PCI
sorry to ask that, but what does almost every PCI system mean?
First of all compared to the Digium TE4xx the Sangomas work in 3,3V and
5V PCI slots. That
[entrada]
exten = s,1,Wait,11
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Dial(SIP/200,60,Ttr)
exten = s,5,Dial(SIP/201,60,Ttr)
exten = s,6,Dial(SIP/202,60,Ttr)
exten = s,7,Dial(SIP/203,60,Ttr)
He probado a añadir esto pero el error persiste , no se si me exprese
bien antes .. lo
Well - the Junghanns does the echocancel in software and the Sangoma
A104d does it in hardware.
Alright, i just had a look at their product lineup. It seems as not only
the A104d but also the low end of their E1 cards (i.e. A101) comes with
this onbard echo canceller (EDAC), right?
No -
solo cambia tu extension.conf
[entrada]
exten = s,1,Wait,11
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Dial(SIP/200,60,Ttr)
exten = s,5,Dial(SIP/201,60,Ttr)
exten = s,6,Dial(SIP/202,60,Ttr)
exten = s,7,Dial(SIP/203,60,Ttr)
I try it , but it doesn´t work , i want call to
On Tue, 30 May 2006, Christian Victor wrote:
Well - the Junghanns does the echocancel in software and the Sangoma
A104d does it in hardware.
Alright, i just had a look at their product lineup. It seems as not only
the A104d but also the low end of their E1 cards (i.e. A101) comes with
What process is taking up 100% CPU? Is it Asterisk processes or
something else? Also, is the load spread out over multiple processes,
or do you have one or two processes taking up 90% or more of your
total?
You also have dual CPUs (and hyperthreading, which to FC3 should look
like 4 CPUs if I'm
G729 is your problem.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
-Original Message-
From: Lachek Butalek [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 30, 2006 10:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] I guess my
Hello Francois,
this is my zaptel.conf:
# 12 red modules (FXO) on tdm2400
fxsks=1-12
fkoks=13-16
defaultzone=it
loadzone=it
This is my zapata.conf:
[channels]
context = outbound_zap
canpark = yes
echocancel = 128
echocancelwhenbridged = yes
faxdetect = incoming
language = us
musiconhold = default
Ok, I've been seeing this email floating around for a while now. If you are
using [EMAIL PROTECTED], don't use this, but if you are just using standard
asterisk with
your own custom dial plan... this should work for you.
I don't actually use this in my dial plan and haven't tested it but it
Why not do:
exten = s,1,AGI(xyz.agi|${MACRO_EXTEN})
bp
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Akpome
Akpoguma
Sent: Tuesday, May 30, 2006 2:55 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] AGI MySql
I have been able to
I would like to know what are the minimum hardware requirements for
Asterisk:
1. Linux kernel 2.4?
2. PC 486 50MHz?
3. Memory 64 Mbytes?
Thanks
Luis
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
Hi,
is there an Asterisk app (or AGI script) to look up names in a LDAP
directory?
--
Domenico Viggiani
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Which (preferible free :-) softphone that supports IAX and RSA
encryption do you recommend? It seems that IDEFisk doesn't yet.
Thanks a lot for your help.
--
Atly.
Alvaro Palma
___
--Bandwidth and Colocation provided by Easynews.com --
On Tue, 30 May 2006, Luis Uebel wrote:
I would like to know what are the minimum hardware requirements for Asterisk:
1. Linux kernel 2.4?
2. PC 486 50MHz?
3. Memory 64 Mbytes?
That depends on what you want to do with Asterisk.
The kernel is alright and the 64MB are also enough, but the 50MHz
Matthias Fechner wrote:
Hello Joshua,
Joshua Colp wrote:
[portunity-out]
type=friend
host=iax.iaxport.de
username=XXX
secret=YY
context=incoming-portunity
notransfer=yes
Only if the username is specified as portunity-out when the other side dials
you. Otherwise your Asterisk has no
Is there a way to control the name and location of the recordings made
with automon? I need to be able to send the file to the client when
they finish a call. How can I know which file belongs to the user?
--
Carlos Chavez Prats
Director de Tecnología
Telecomunicaciones Abiertas de
Hi,
I have a setup something like the following:
[callcenter] --IAX2-- [voip-server] ---IAX2-- [LD provider]
I also have PRIs going in and out of [voip-server].
The problem is with calls going between [callcenter] and [voip-server].
When a call comes in the PRI to [voip-server] and then to
I have a production server running a CVS Head release dated
8/27, which is pretty much 1.2 minus some last minute additions, 1.2 was
released at the end of august 2005.
There is a sip channel patch related to presence and sip
subscriptions that I wish to apply, but since the server has
Faris Raouf wrote:
But I need to get an LED to light up on a GS in Location2 when a line on
the Polycom at Location1 is in use. Is this possible? If so, can anybody
give me any pointers as to how?
Not at this time, no. There has been talk of building a method for doing
this, but so far there
Hi Attilla -
So, this left me only one conclusion. The application with the memory
leak is Asterisk.
I know every situation is different, but I just thought that I'd point
out that I have machines running 1.2.7.1 that I haven't restarted in
months. Of course, 1.2.7.1 hasn't been out that
Frank Pani wrote:
make -C gsm lib/libgsm.a
make[2]: Entering directory `/usr/src/asterisk-1.2/codecs/gsm'
as -o src/k6opt.o src/k6opt.s
Sorry, I missed that one when I did the first s390 fix. It's been taken
care of now.
___
--Bandwidth and
Damon Estep wrote:
There is a sip channel patch related to presence and sip subscriptions
that I wish to apply, but since the server has been very stable I do not
want to do a full upgrade to 1.2.7.1.
There have been hundreds of bugs fixed since 1.2 was released. I think
it would be much
I am having this issue.. asterisk 1.2.7 using native MOH without
mpg123. Calls get put on park and EVERYTIME it seems to start the
music off the same (even though I've chosen to do it randomly!)...
then after playing the first song it just goes to dead air.
STEPS TO REPRODUCE:
Xfer -- 70
Does anyone out there have a sample config they can share
for the Polycom 501? Is it possible to do sub configs like you
can with the Aastra 9133i? It could be just me but the boot configs seem a bit
cryptic compared to the aastra. Also do any of you have any comparisons between
these and
In fact the Sangoma has 128ms echo cancel per channel. As far as I know
neither the Digum harware echo cancel nor the available software
solutions offers this.
I thought the Eicon cards were the only ones with 128ms echo-cancel ;-)
I am happy that I could enlighten you a bit in this point.
Christian Victor schrieb:
I personally prefer the Sangoma E1 cards. The work in almost every PCI
system
Okay guys - I have to add something to this so it will not be misunderstood:
I never had problems with a Sangoma card in any mainboard and just added
the word almost to prevent somebody
Has anyone been able to compile Zaptel after upgrading to 2.6.9-34.0.1.EL kernel? I'm running CentOS and was unable to recompile Zaptel. I reverted back to 2.6.9-22.0.1.EL but even then Zaptel would fail to load. I tried pulling the latest zaptel sources from digium cvs, but they apparently don't
crontab? I restart my asterisk nightly with cron but a simple typo
could make that every minute instead of every day... shrug
Woodoo People .pGa! wrote:
Hi!
Probably any of you meet with the following problem:
asterisk is restarting in a minute (if no active call) if active call,
it says
The CVS server for app_conference seems to be down.
Can somebody mail me a recent copy of the sources please?
--
Henry J. Cobb
http://www.io.com/~hcobb/
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
In my extension.conf I have both
[ext-did]
include = ext-did-custom
exten = 0108680550,1,Set(FROM_DID=0108680550)
exten = 0108680550,n,Set(FAX_RX=disabled)
exten = 0108680550,n,Goto(timeconditions,3,1)
exten = _01086805XX,1,Set(FROM_DID=_01086805XX)
exten = _01086805XX,n,Set(FAX_RX=disabled)
make this :(BE SURE this is the kernel you are booting!!! make ls inside /usr/src/kernels)cd /usr/src/kernels/2.6.9-34.0.1.EL/include/linuxmv spinlock.h spinlock.h.oldwget http://
nerdvittles.com/aah27/spinlock.hshutdown -r nowThen make this command: rebuild_zaptelThis is a well known issue with
Hello Joshua,
* Joshua Colp [EMAIL PROTECTED] [30-05-06 12:41]:
happen as you might expect. You need to use permit and deny lines to get
the user entry matched. Check into the sample iax.conf to see how to do
this.
thx a lot!
The following entry in iax.conf is doing the trick:
You should use 2 FXO ports on asterisk to accomplish it, make sure the
ports supports flash, otherwise you will have a hard time transferring
back to the PBX. BTW, panasonic has one of the best documentation for
foreign voicemail integration, just tell the panasonic PBX that you
are using DTMF
yes, a2billing.php is in agi-bin:
[EMAIL PROTECTED] locate a2billing.php
/usr/src/a2billing/Chameleon/A2Billing_AGI/a2billing.php
/var/lib/asterisk/agi-bin/a2billing.php
Could be because of the missing pcntl php extension?
[EMAIL PROTECTED] rpm -qa | grep php
php-mysql-4.3.9-3
php-ldap-4.3.9-3
Yes, that worked like a charm. Thanks!On 5/30/06, Marco Mouta
[EMAIL PROTECTED] wrote:
make this :(BE SURE this is the kernel you are booting!!! make ls inside /usr/src/kernels)cd /usr/src/kernels/2.6.9-34.0.1.EL/include/linuxmv spinlock.h spinlock.h.oldwget http://
Steve Underwood:
Steve, why do some numbers give protocol errors? Ive noticed here in Mexico
that certain numbers when dialed return protocol failure and a busy tone.
Any idea why this happens and why with only certain phone numbers?
___
--Bandwidth
Trunk version of Asterisk has a cdr_radius module I believe now.
-John
Oliver Vermeulen wrote:
Hi List,
I'm looking for a Asterisk radius module ... Anybody has one ?
Thanks,
Oliver
*Oliver Vermeulen
*
*_World Venture Group Telecom_* *_
_*
*Corporate Address:**
*Str Avionului Nr
Did you just make this change recently? I just tried to download the 1.2
stream with revision 30861 and after trying make see the same thing.
Perhaps I need to excercise patients.Thanks again for your help - it'll
be good to say this runs on mainframe once we get it their.
eg:
Thanks for fast answer. I would like to hold 4 analog channels or 4 digital
channels,
voice mail with this system. I read that it's necessary 30 MHz per channel.
Is it true?
In this case, I will need 120MHz.
Luis
I would like to know what are the minimum hardware requirements for
Asterisk:
1.
The wiki has a lot of info, but the nerdvittles.com site has been indispensable.
--
--
Steven
http://www.glimasoutheast.org
Marco Mouta [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
make this :
(BE SURE this is the kernel you are booting!!! make ls inside /usr/src/kernels)
cd
We're using [EMAIL PROTECTED] 2.8 (same thing occurs with
earlier versions). We have it set up so that if we don't answer our
internal SIP phones it does "follow me" to our cell phones. When Asterisk
forwards the calls to our cell phones, the Caller ID shows our outbound number,
not the
Is there a way to control the name and location of the recordings made
with automon?
Set the variable TOUCH_MONITOR for arguments to send to the
application Monitor() in either the caller or calle channel. The
caller channel si always checked first. Check the documentation in
voip-info, someone
Frank Pani wrote:
Did you just make this change recently? I just tried to download the 1.2
stream with revision 30861 and after trying make see the same thing.
Perhaps I need to excercise patients.Thanks again for your help - it'll
be good to say this runs on mainframe once we get it
Hi List,
In one of my AGIs (using DeadAGI) I grab the answered time using:
my $res = $agi-exec (DIAL $dialstring);
my $answeredtime = $agi-get_variable (ANSWEREDTIME);
However this information differs from what's written in the Master.csv
file (which happens to be the correct
I already installed pcntl but the billing isnt workin.
I followed the Asterisk2Billing wiki and putted this line in the end of
sip.conf:
#include additional_a2billing_sip.conf
but when I dial, Asterisk answers 407 Proxy Authentication Required
If I do comment the line in sip.conf ( ;
I think Asterisk2Billing is trying to play some audio file to make the
callers put a PIN number.
But can I use it without the PIN, and configure Asterisk2billing to
check the database to see if the user exists?
Thanks
Joao Pereira
Vahan Yerkanian wrote:
Greetings,
pcntl is a required
Erick Perez wrote:
Are there any good scripts to stress test MoH?
I want to test this machine for 1000 calls on hold.
Steve Totaro wrote:
When I say high, I mean 1,000+ calls.
Erick and Steve,
You both speak of Asterisk systems capable of handling 1,000 calls. I
currently have a Dell
Anyone know if #include works in ael yet?
extensions.ael:
#include inc/pbx/global.conf
context test_context {
};
*CLI ael reload
May 30 13:56:45 NOTICE[8516]: pbx_ael.c:1120 handle_root_token: Unknown root
token '#include'
May 30 13:56:45 WARNING[8516]: pbx.c:3758 ast_merge_contexts_and_delete:
No, only works in the old language, or in AEL2 which is released in trunk.
On Tue, 30 May 2006, Douglas Garstang wrote:
Anyone know if #include works in ael yet?
extensions.ael:
#include inc/pbx/global.conf
context test_context {
};
*CLI ael reload
May 30 13:56:45 NOTICE[8516]:
Is Asterisk svn link down ?when I issue the folowing command, I gotsvn checkout http://svn.digium.com/svn/asterisk/trunk asterisk400 Bad Request (http://svn.digium.com)
Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or
I have the following extension to dial outside via SIP
it's like this:
phoneasterisk-internet-SIP providerUSA
exten = _91NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _91NXXNXX,2,Dial(${SIPtrunk}/${EXTEN},55,o)
exten = _91NXXNXX,3,Hangup
I want to strip the digit 9
1 - 100 of 198 matches
Mail list logo
