Mark Quitoriano wrote:
but that's already 1.2? is it advisable to upgrade my current version
1.0.9 to 1.2 already? any big changes to be done to my current setup to
upgrade it to 1.2?
If in three or four days you go to upgrade your version of 1.0.9, it will be
upgraded to 1.2.
So, why not do
On 11/13/05, Daniel Clark [EMAIL PROTECTED] wrote:
Thanks for the reply, it's an
approach I didn't think of to simply include the information from the
other contexts into where I would be including from. In most cases that would
work, but not in my case. Each user of my system will
I get an error when patching the makefile, seems the order is different.
Had the same problem with rc1 and 2.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: 13 November 2005 17:34
To: 'Asterisk Users Mailing List - Non-Commercial
Hello Sacha,
While it is not the best solution as far as quality is
concerned, I would suggest you at least try the aggressive canceller in
zconfig.h. I put it into use temporarily while I get an external echo can
setup. It takesa bit of getting used to (no simultaneous
speech/duplex),
=21DOCTYPE HTML PUBLIC =22-//W3C//DTD HTML 4.0 Transitional//EN=22
htmlheadmeta http-equiv=3D=22Content-Type=22 content=3D=22text/html; c=
harset=3DISO-8859-1=22
style type=3D=22text/css=22body=7Bmargin-left:10px;margin-right:10px;marg=
in-top:10px;margin-bottom:10px;=7D/style
/head
body
Anton Krall wrote:
Guys.
Do any have some already made scripts for load testing or creating lots of
calls for load testing an asterisk install?
Wanted to check with you first, since probably somebody has done this
before.
Use simpleclient command line client from the iax cvs repository.
cp napisał(a):
I still can’t get it to work.
My configuration is (extension.conf):
[macro-call]
exten = s,1,Dial(SIP/${ARG1},15)
exten = s,2,Goto(s-${DIALSTATUS},1) ;
NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER
exten = s-NOANSWER,1,Voicemail(${ARG1})
exten = s-NOANSWER,2,Hangup
exten =
Hi friends,
I am new in asterisk, i installed the Asterisk on my
Redhat EP. But i am not able to register any SIP
softphone. i am getting Unathurize message when in SIP
debug.
Here is my sip.conf configuration
[general]
context=default
realm=asterisk
port=5060
bindaddr=0.0.0.0
srvlookup=yes
Abdul Lateef napisał(a):
Please help me to find the problem.
What type of phone do you have? Try to upgrade the firmware in it, it
worked for me.
Also first try to register your phone without any username and password
(just comment them in sip.conf). Is it registering ok?
--
Best regards,
Rafael R. GV wrote:
PS: mysql-schema does not work properly in mysql 5.0.x because only one
timestamp with default now() in a table its allowed as you told me and
also I´ve found other issue related to auto_increment value:
ERROR 1064 (42000) at line 87: You have an error in your SQL syntax;
Title: ISDN card required
Can anyone point me in the direction of a quality, works with Asterisk, BRI card. I need minimum 2 port/4 channel.
Regards
Lee
###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more
great! i'll try it later tnx!On 11/14/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Monday 14 November 2005 07:20, Mark Quitoriano wrote: Hi, is there a howto to install
g.729 codec on
As I recall, the driver for the x100p was called wcfxs (or something
like that), and those driver functions were merged into wctdm about a
year ago. Now, wcfxs is an alias for wctdm.
What you recall is only partially correct:
In 1.0 wctdm is an alias for wcfxs. Furthermore, wcfxs
you mean the way you setup asterisk 1.2 dialplan is different with 1.0.9?On 11/14/05, Matt Riddell [EMAIL PROTECTED]
wrote:Mark Quitoriano wrote: but that's already 1.2? is it advisable to upgrade my current version
1.0.9 to 1.2 already? any big changes to be done to my current setup to upgrade
I have a problem with group and attended transfer.
I have tested below example dialplan with asterisk-1.2.0-beta1,
asterisk-1.2.0-rc1 and and astesik-HEAD on 11/14/2005.
I have simple test dialplan like:
[default]
exten = 210,1,Macro(stdexten,${EXTEN},SIP,test1)
exten =
Hi Everybody,
I'm trying to execute a MYSQL(UPDATE..) sql
command over a table I have previously red. I get a timeout and no update
happens.
I use * 1.0.9.
I wonder if MYSQL set of commands allows Update...
Best regards
Mauro Zanin
where can i find a howto about this?On 11/14/05, Michael Toop [EMAIL PROTECTED] wrote:
Hi,Not natively, but you can run a Bash command in your extensions.conf use Lame or Sox to do the conversion for you.Cheers,MICHAEL TOOPTel 011 602 9309Fax 011 656 1342
Mobile 083 364 2370Web
Rafael R. GV a écrit :
[...]
2.- Change value FG_TABLE_NAME=call t1 for FG_TABLE_NAME=calls
t1 in following files:
A2Billing_UI/Public/call-log-customers.php
A2Billing_UI/Public/invoices.php
A2BCustomer_UI/balance.php
A2BCustomer_UI/invoices.php
You forgot:
Easy:
show g729
This will show total in use and total available channels for g729
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel
Sent: Monday, November 14, 2005 6:55 AM
To: Asterisk Users Mailing List -
On Mon, 2005-11-14 at 07:46 -0500, Sean Cook wrote:
Easy:
show g729
This will show total in use and total available channels for g729
doesnt work for me, maybe its a version difference.
I do have g729 loaded, and that was verified.
--
Trixter http://www.0xdecafbad.com Bret McDanel
yes that's what i'm lead to believe as well. Only the SPA-2100
SPA-2002 support two simultaneous g.729 calls, the older/lesser models
don't have the processing power required to encode two g.729 streams.
Rich Adamson wrote:
I don't think they want to solve it. It's the same with the
On Mon, 2005-11-14 at 07:46 -0500, Sean Cook wrote:
Easy:
show g729
This will show total in use and total available channels for g729
doesnt work for me, maybe its a version difference.
I do have g729 loaded, and that was verified.
Do you have the Digium G.729 codec installed?
You need the libidn-devel package installed.
On 11/14/05, Zeeshan [EMAIL PROTECTED] wrote:
How do I install curl?
Zeeshan A Zakaria
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Sunday, November 13, 2005 10:41 PM
To: Asterisk Users Mailing List -
On Mon, 2005-11-14 at 13:57 +0100, Andreas Sikkema wrote:
On Mon, 2005-11-14 at 07:46 -0500, Sean Cook wrote:
Easy:
show g729
This will show total in use and total available channels for g729
doesnt work for me, maybe its a version difference.
I do have g729 loaded, and
On 11/10/05 15:02 Wayne Gemmell said the following:
When trying to call from this side to that side I get the following
-- Executing Dial(SIP/301-2d50,
IAX2/wayne:[EMAIL PROTECTED]/204) in new stack
Nov 10 08:37:21 WARNING[30785]: channel.c:455 ast_best_codec: Don't know any
of 0xf800
On 11/10/05 17:36 Wayne Gemmell said the following:
On Thursday 10 November 2005 10:55, Jason Walker wrote:
The statement of zaptel being required is strange...I use IX trunking
exclusively for my servers. Two of them have no zaptel/Digium hardware and
the trunk calls are fine.
I don't
show g729
From: Mark Quitoriano [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] How to check how many G729 codec
licenseinstalled
Date: Mon, 14 Nov 2005 19:13:29 +0800
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
I've got two customers on the same broadband
provider. Same Asterisk box on my end. Same CLEC.
One has an IAXy and the other has an Asterisk box
with an array of devices (Grandstream, Cisco, ATCOM, xten, etc.).
The people behind the Asterisk box have had no
audio quality issues. The person
Does the phone ocasionally prompt the user for a password? -Mike
-Original Message-
From: Richard Watson [mailto:[EMAIL PROTECTED]
Sent: Monday, November 14, 2005 5:00 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Snom clients deregistering
-BEGIN PGP
Michael Crown wrote:
Does the phone ocasionally prompt the user for a password? -Mike
Yes it does
How did you know?
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Easy:
show g729
This will show total in use and total available channels for g729
doesnt work for me, maybe its a version difference.
I do have g729 loaded, and that was verified.
Using cvs-head...
If you have the digium licensed g729, the 'show g729' looks like:
show g729
0/0
Lower speaker volume on the phone connected to IAXy.
On Mon, 2005-11-14 at 07:21 -0600, Mike Hammett wrote:
I've got two customers on the same broadband provider. Same Asterisk
box on my end. Same CLEC.
One has an IAXy and the other has an Asterisk box with an array of
devices
There is a setting on the Advanced page called Challenge Response on
Phone. Turn this setting to Off and your problem will be solved. Also, we
usually set the Proposed Expiry to 1 minute On the SIP page when phones
are behind a NAT.
-Mike
-Original Message-
From: Richard Watson
I've got two customers on the same broadband provider. Same Asterisk box on
my end.
Same CLEC.
One has an IAXy and the other has an Asterisk box with an array of devices
(Grandstream, Cisco, ATCOM, xten, etc.).
The people behind the Asterisk box have had no audio quality issues.
You will also experience this if the latency between the Asterix PABX
and IAXy is so high that echo cancel don't work.
Jan
Rich Adamson wrote:
I've got two customers on the same broadband provider. Same Asterisk box on my end.
Same CLEC.
One has an IAXy and
Are you running the g729 module from digium? Registered?
Sean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel
Sent: Monday, November 14, 2005 7:50 AM
To: Asterisk Users Mailing List - Non-Commercial
Hi *users,,
I'm researching on Asterisk PBX phone system initially I was successfull in
configuring 2 SIP users with DIAL rules in extension.conf and
configured 2X-Lite softphones to use my proxy
Registered successfully also able to dial and communicate.
Now i am trying to dial from softphone
I recently implemented a Sipura SPA-2002 with one of my Asterisk
installations. On internal calls, the SPA generates ringtone as
expected. However, when I dial out via my IAX-based service
provider, I hear
both the telco-generated ringtone as well as the SPA-generated
ringtone.
Does anyone know how to put an Aastra PT 390 in
headset mode, so it will only give a dial tone when
you are ready ? Right now I can't figure how to keep
it hung up? If I hit googbye it merely flashes (give
me a dial tone again).
Any help would be greatly appreciated?
After reviewing many other posts as well as wiki information
on canreinvite and asterisk media path I am not clear on whether asterisk still
manages sip signaling after a reinvite has been issued between a peer and a UA.
Here are the details;
UA g.711u Asterisk g.711u SIP long
Lee Archer wrote:
Can anyone point me in the direction of a quality, works with Asterisk,
BRI card. I need minimum 2 port/4 channel.
Ack. Like Mark pointed out, I also used Junghanns.net cards, works fine.
Cheers.
___
--Bandwidth and Colocation
Thanks to all. I'll probably go with the quadBri card they do.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kristof
Hardy
Sent: 14 November 2005 14:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
The VoIP Connection wrote:
There is a setting on the Advanced page called Challenge Response on
Phone. Turn this setting to Off and your problem will be solved. Also, we
usually set the Proposed Expiry to 1 minute On the SIP page when phones
are
In article [EMAIL PROTECTED],
Mauro Zanin [EMAIL PROTECTED] wrote:
Hi Everybody,
I'm trying to execute a MYSQL(UPDATE..) sql
command over a table I have previously red. I get a timeout and no update
happens.
I use * 1.0.9.
I wonder if MYSQL set of commands allows
I have a 4 port brooktrout PCI E1/T1 blade card (MPAC 1200) that
was used for some carrier server. Will Asterisk support this? Has
anyone used this
successfully before? Thanks! Stephen
___
--Bandwidth and Colocation sponsored by Easynews.com --
Hi All
Can anybody tell me the maximum number of SIP Phones
supported by Asterisk.
regards
ramakrishnan.n
__
Start your day with Yahoo! - Make it your home page!
http://www.yahoo.com/r/hs
___
Is there any way to adjust the sample size asterisk uses for VoIP
codecs? From what I have gathered it uses a fixed 20ms sample size for
all codecs. While some require at least this, some can be configured
for less. This results in more overhead, but can be tweaked to provide
more efficient
Sorry, I just saw the post.
Yes, it's the same format
Regards;
Chawki
--- Matt Riddell [EMAIL PROTECTED] wrote:
chawki hammoud wrote:
Hi:
I have been having this problem for sometime that
I am
not able to solve and I hope someone can help.
I can make VOIP calls between my
Hello,
Hello, I do not arrive has to connect me has a gateway h323, in
termination of call.
i have one ip for a termination call xxx.xx.xx.xx,
I do not know if the problem comes from my parameters oh323.conf or the gateway
i using a latest version asterisk (asterisk 1.2rc1),openh323 latest
On Mon, 2005-11-14 at 06:53 -0800, nr k wrote:
Hi All
Can anybody tell me the maximum number of SIP Phones
supported by Asterisk.
If I run asterisk on my ipaq not very many. If I run it on a real
server many many more.
Your question cant really be answered with the information you have
You need a Stun Server...vovida.org works for meOn 11/11/05, Enrique Leon [EMAIL PROTECTED]
wrote:Second postI have installed Asterisk on SuSE 10.0 with an active firewall/NAT filter.
The server has connection to my own Intranet (private IP) and to InternetEverything works well for clients behind
Hi,
I've a problem with a cisco 827-4v and asterisk (1.0.9) acting as
sip-to-pstn GW. The issue is that when a call comes in from the pstn,
asterisk correctly contacts the router, which in turns send a 183
Session progress. Obviously, asterisk thinks that the telephone is not
ringing (because it
Kristof Hardy wrote:
Lee Archer wrote:
Can anyone point me in the direction of a quality, works with
Asterisk, BRI card. I need minimum 2 port/4 channel.
Ack. Like Mark pointed out, I also used Junghanns.net cards, works fine.
Hi Kristof!
(sorry for the empty email)
Do you use it with
Hello,
Hello, I do not arrive has to connect me has a gateway h323, in
termination of call.
i have one ip for a termination call xxx.xx.xx.xx,
I do not know if the problem comes from my parameters oh323.conf or the gateway
i using a latest version asterisk (asterisk 1.2rc1),openh323 latest
Any way to comment out a line (or some text) in an AEL file?
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Michael Crown wrote:
Did you change the proposed expiry? -Mike
Yes, now set to 1 minute.
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
//comment
AEL ignores any text from // till the line end.
On Mon, 2005-11-14 at 07:39 -0800, Ed Greenberg wrote:
Any way to comment out a line (or some text) in an AEL file?
___
--Bandwidth and Colocation sponsored by Easynews.com --
hi,
723 needs 30ms + 5ms forward. 729 needs 10ms + 5ms forward. The
'standard' for 711 is actually 6ms (48 bytes). This would have to be
done per channel (or per codec), but I am not sure wherever Asterisk
allow per codec size or run's with one static size???
Jan
trixter aka Bret McDanel
On Mon, 2005-11-14 at 16:57 +0100, [EMAIL PROTECTED] wrote:
hi,
723 needs 30ms + 5ms forward. 729 needs 10ms + 5ms forward. The
'standard' for 711 is actually 6ms (48 bytes). This would have to be
done per channel (or per codec), but I am not sure wherever Asterisk
allow per codec size or
Hi all,
I am trying to install an AVM Fritz card USB v2.1 on
my Asterisk Box.
I am using Debian Sarge with 2.6.8 kernel.
I have compiled capi last drivers (fcusb2-suse93-3.11-07.tar.gz)
and have copied fcusb.ko to /lib/modules/2.6.8/extra/.
All modules seems loaded (capi, capifs,
Hi all,
Having read the various fax and asterisk pages on voip-info, am I right in
thinking I should be able to bridge Zap channels carrying fax without
reliability problems (which as I understand things plague Fax-over-IP)?
The reason for asking is in relation to a requirement where both fax
If you patch rc1's makefile manually, you can compile spandsp without
problems, but if you try the same thing with rc2, you'll notice that spandsp
seems to be broken against rc2.
Waiting for Steve to shed some light on this.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL
Hi guys,
this is the scenario:
PRI -Asterisk-SER
If I call from a Sip(SER) user everything is good, I can call
anywhere, but if I try to call from outside(PRI) everything is
wrong!!!
This is the CLI for an incoming call:
--
ast*CLI
-- Executing SetCallerID(Zap/14-1, outside) in
Hi Ramon,
I have used Asterisk 1.09, SuSE 9.3 with a TDM400M with 4 FXOs. I'm
planning on trying 10, but haven't found the time.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, November 13, 2005 9:19 PM
To:
Anton Krall wrote:
If you patch rc1's makefile manually, you can compile spandsp without
problems, but if you try the same thing with rc2, you'll notice that spandsp
seems to be broken against rc2.
Waiting for Steve to shed some light on this.
As far as I know spandsp does NOT require
Hi,
I have downloaded the latest release candidate v1-2-0-rc2 and I was
checking the readme's in the zaptel directory and I came across a
requirement I have not seen before. In the readme it says that for Linux
2.6 kernel you will need to have the 'CRC-CITT' functions compiled with
the
Hi all,
I am setting up a a proof on concept where a SIP phone sits on the
internet and connects to a * behing a NAT.
Right now the SIP phone connects to the * box just fine, I can dial
and I see the commands being executed on the * box, but I don't have any
audio on the SIP phone. Any
Chuck Bunn wrote:
You have
lsmod | egrep crc_ccitt
check in your kernel modules looking for crc_ccitt
but FC4 comes with that
regards
Saul
Hi,
I have downloaded the latest release candidate v1-2-0-rc2 and I was
checking the readme's in the zaptel directory and I came across a
Hi
i have installed FC4 with select everything
with your command i dont find any thing
results are null
thats means its not installed ??
ram
On 11/14/05, Saul Diaz [EMAIL PROTECTED] wrote:
Chuck Bunn wrote:You havelsmod| egrep crc_ccittcheck in your kernel modules looking for crc_ccitt
but FC4
On Nov 14, 2005, at 2:50 AM, Dinesh wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom
Rymes
Sent: Monday, November 14, 2005 1:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Anybody tried it
hi,
723 needs 30ms + 5ms forward. 729 needs 10ms + 5ms forward. The
'standard' for 711 is actually 6ms (48 bytes). This would have to be
done per channel (or per codec), but I am not sure wherever Asterisk
allow per codec size or run's with one static size???
There is a patch in Mantis,
There are a number of asterisk implementations in India.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tom Rymes
Sent: Monday, November 14, 2005 12:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Its illegal to interconnect it to the local pstn (from abroad).
Dinesh.
I still don't see how this would stop him from using no VOIP
protocols and plugging it in to the PSTN. Just use Asterisk as a PBX,
no VOIP, no bypassing the Indian telephone monopoly (assuming that
there is one).
On Mon, 2005-11-14 at 09:11 -0800, Dan Austin wrote:
hi,
723 needs 30ms + 5ms forward. 729 needs 10ms + 5ms forward. The
'standard' for 711 is actually 6ms (48 bytes). This would have to be
done per channel (or per codec), but I am not sure wherever Asterisk
allow per codec size or
Yes. But will not monitor more than 7 of the phones. It lists
them in order of the last name entered into the directory not even in order of
the speed dial setting.
Any ideas? I have more 20 polycom 601/501 phones deployed
Michael
I currently have broadvoice service for two lines and one analog line coming from Cox cable telephone. I would like to connect the single analog line to my asterisk server, and then use only IP-based telephony for all of my handsets.
What is the best method for accomplishing connecting a single
On Nov 14, 2005, at 6:21 AM, Markos Paraskevopulos wrote:
Hello everyone,
I’m new to VoIP and despite a lot of reading, I’m kind of more
confused than before.
I have following question – we currently have hardware Alcatel PBX
and approx. 50 phones in the company. I was wondering if we
I am in the process of replacing a Zultys MX250 with an Asterisk PBX
system. Right now, there is a PRI T1 line coming in from the phone
company that plugs directly into the MX250. I believe Digium offers
PCI cards that will provide this same functionality, but what I would
really like is
On Mon, 2005-11-14 at 12:15 -0500, Dean Collins wrote:
There are a number of asterisk implementations in India.
And the commercial ones are finally legal.. Right about the time they
arrest some guys running a VoIP shop they make it legal, all in the same
week. Biggest concern india appears to
On Mon, 2005-11-14 at 09:18 -0800, Dan Austin wrote:
Its illegal to interconnect it to the local pstn (from abroad).
Dinesh.
I still don't see how this would stop him from using no VOIP
protocols and plugging it in to the PSTN. Just use Asterisk as a PBX,
no VOIP, no bypassing
On Nov 14, 2005, at 11:57 AM, Andre Courchesne - Consultant wrote:
Hi all,
I am setting up a a proof on concept where a SIP phone sits on the
internet and connects to a * behing a NAT.
Right now the SIP phone connects to the * box just fine, I can
dial and I see the commands being
If 'capiinfo' does not work, chan_capi will fail too.
Do you have the node /dev/capi20 with correct permissions?
Armin
On Mon, 14 Nov 2005, Amaury BOSSE wrote:
Hi all,
I am trying to install an AVM Fritz card USB v2.1 on my Asterisk Box.
I am using Debian Sarge with 2.6.8 kernel.
On Nov 14, 2005, at 12:24 PM, Wylie Swanson wrote:
I currently have broadvoice service for two lines and one analog
line coming from Cox cable telephone. I would like to connect the
single analog line to my asterisk server, and then use only IP-
based telephony for all of my handsets.
You need a Digium TDM01B which will allow you to connect your single
analog (FXO) line to your Asterisk server. From there, you can go
Ethernet out of your Asterisk box, through a switch, to your IP endpoints.
Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com - Everything you
On Mon, 2005-11-14 at 11:57 -0500, Andre Courchesne - Consultant wrote:
Hi all,
I am setting up a a proof on concept where a SIP phone sits on the
internet and connects to a * behing a NAT.
Right now the SIP phone connects to the * box just fine, I can dial
and I see the commands
try typing modprobe crt-ccitt before the commands Chuck wrote, or
alternatively, if you've got your slocate database up to date, you could
try locate crc-ccitt.ko
ram wrote:
Hi
i have installed FC4 with select everything
with your command i dont find any thing
results are null
thats means
Look into the Lucent Max Tnt. It may be over kill for your application, but
people say they work flawlessly and will increase the stability of your
asterisk PBX. I'm in the process of setting one up right now.
Good Luck
Marc
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL
how can i check how many g729 are being used right now?On 11/13/05, Gentian Bajraktari [EMAIL PROTECTED]
wrote:Yes.- Original Message -From: Angelito Manansala
[EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com
Sent: Sunday,
Hello everyone,
Im new to VoIP and despite a lot of reading, Im
kind of more confused than before.
I have following question we currently have
hardware Alcatel PBX and approx. 50 phones in the company. I was wondering if
we would need to change the phone service provider, because they
Dependent on what Channel you are
using..
If you are using SIP then do:
*CLI sip show channels
Peer
User/ANR Call ID Seq
(Tx/Rx) Format Last Msg
Where Format will show the current codecs used at
that time..
Rg,
Gentian
- Original Message -
From:
Mark
Quitoriano
Have you try Arbitrator Open source based on Linux, there is called
ArbiQos which optimum for Voip and Video stream, that has priority the
bandwidth for voip and it can gone when no others voip or video stream used
Francois Meehan wrote:
Hi all,
We have an asterisk server inside a network
Don't know will this do, but a simple comparison
may give you a hint:
1.2-rc2 extentions.conf
[default]
switch = Realtime/[EMAIL PROTECTED]
switch = Realtime/[EMAIL PROTECTED]
;switch = Realtime/[EMAIL PROTECTED]
1.0.9 extentions.conf
[default]
include = astcc
include = internal
[astcc]
exten =
So far - the 4-port ISDN HFC chipset cards from Junghanns.net works well
and is half the price of a 4-port Eicon card.
On Mon, 2005-11-14 at 10:07 +, David Waugh wrote:
Hi Lee,
I use a Diva Server card here with Asterisk using Chan_capi.
The basic BRI card has one BRI port. They also
How do I install curl?
Zeeshan A Zakaria
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Sunday, November 13, 2005 10:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Installation exits with following
error
Have you try Arbitrator Open source based on Linux, there is called
ArbiQos which optimum for Voip and Video stream, that has priority the
bandwidth for voip and it can gone when no others voip or video stream
used
Francois Meehan wrote:
Hi all,
We have an asterisk server inside a
I can not see that its illegal to have Asterisk in India. The TDM400P
card should work fine - but it may not be approved to be interconnected
to the phone system. (This never stopped me doing similar things).
I'm assuming that its possible to connect a 2-wire phone to the Indian
phone system - ie
On Mon, 2005-11-14 at 12:21 +0100, Markos Paraskevopulos wrote:
Hello everyone,
I’m new to VoIP and despite a lot of reading, I’m kind of more
confused than before.
I had an asterisk system up and running then read some dox and becuase
what I read at that time wasnt well written it has that
That's what macro is useful for. Don't include these common contexts, but convert them to macros and call these macros from user's dialplan. Macros in asterisk dialplan are close to subroutines rather than to C-style #define.
On Mon, 2005-11-14 at 07:44 +, Daniel Clark wrote:
Thanks
Rich Adamson wrote on Sunday, 13 November 2005 7:36 PM:
I recently implemented a Sipura SPA-2002 with one of my Asterisk
installations. On internal calls, the SPA generates ringtone as
expected. However, when I dial out via my IAX-based service
provider, I hear
both the telco-generated
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all,
I have a server currently running Asterisk 1.0.7 placed out in the wild
(i.e. not behind NAT).
I have groups of sip clients all behind various NAT firewalls (mainly
adsl routers).
Up to now I've mainly used Sipuras and not had any serious
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