I have 4.5.4 CDRTool version. I patched my cdr_addon_mysql like this:
cd ../asterisk-addons
- Add a line into asterisk-addons/Makefile reading:
CFLAGS+=-DMYSQL_LOGUNIQUEID
- edit cdr_addon_mysql.c and replace the line reading
AST_MUTEX_DEFINE_STATIC(mysql_lock);
with
Hi,As you said, I have tested. But, still callerid is not displaying. On the console, Asterisk is giving below error: *CLI Aug 14 15:09:58 ERROR[27056]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-8) Aug 14 15:09:58 WARNING[27056]: chan_zap.c:6087 ss_thread: CallerID feed failed:
Absolutely. The SER/OpenSER documentation is terrible, and if you post to
the OpenSER mailing list, you get very cryptic replies.
___
Whilst I would agree with you regarding SER, the documentation of OpenSER is
far better.
Documentation of Asterisk
On 2006-08-15 13:10:05 -0700, Dovid Bender [EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
I have spoken to some one who is interested in investing into building =
equipment for asterisk. I am looking to find out what products that the =
asterisk community would like to
Hi all
Onsip.org is the best option for startup
and openser has many more option integrating with Voice mail with Astrisks
openser.org have lot of documentation
Ram
On 8/16/06, kjcsb [EMAIL PROTECTED] wrote:
Absolutely. The SER/OpenSER documentation is terrible, and if you post tothe OpenSER
Hi,As you said, I have changed my configurations. But, callerid is not displaying. What I have to do? Please tell me.ThanksRegards,Chandra.Rich Adamson [EMAIL PROTECTED] wrote: Crazy Boy wrote: Hi Friends, We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I have connected my
hi,
we've got a Diva Server BRI-2M PCI, SN:3485 card in our asterisk server.
we use the latest divas4linux-melware-3.0.g-106.628.1-1 driver for it.
the card is connected to a Bosch Integral33 PBX. the two system
connected with an S0 line in order the two pbx be able to call
eachother. when we call
Hi Curt,That probably suggests that with SIP they're handing off the RTP to their upstream provider and just dealing with the signalling which is very low overhead. With IAX they have to transport both unless they're interconnecting upstream by IAX and can transfer. In my experience the load is
Dear astreisk-users mailing list subscribers,
with my asterisk under debian (Version: 1:1.0.9.dfsg-5) i have the problem, that
the RTP stream is not set up correct with an outgoing call. Incoming calls are
working with no problems.
The problem is, that the RTP stream is initiated from IP A and
On Tuesday 15 August 2006 19:05, Doug Lytle wrote:
Jessee J Holmes wrote:
Doug,
That is correct you can only display the number on the BudgetTone 101,
102, and 200.
If you wish to display the name as well, you will need to upgrade to
the GXP-2000 phone.
I'm not, Guus is.
Doug
Hi
on my 7940 Phones here, this is the first Part of the Factory Reset
Procedure
after Step 3 and the Status Message you have to hit all Keys on the
Number Pad (1 - 2 - 3 - 4 - #) and then answer the Question by
hitting Number 2
Cu David
Maxx Lobo schrieb:
Fastest way (wipes
We make a call from a PBX through asterisk to ISDN
(E1) Denmark.
We use asterisk to record incoming and outgoing calls
Sometimes we experience that a outgoing call doesnt
get through. In the message.log we get WARNING[7594] app_dial.c: Unable to
forward voice
In the Master.csv we get
Hi, I couldn't find in you sip.conf of your central server, the line context=clientes-sip, did u forget to past, or u r missing it, or i'm missunderstanding?That could be the problem! You MUST define the context for your ATA devices in central server, so * will look for this context in
http://www.electronicscience.com/ has a good IAX2 softphone called ESC SoftphoneOn 8/16/06, David Thomas
[EMAIL PROTECTED] wrote:Sorry, poor reply.
Yes I use it on WM5, and have not seen any problems. I admit I don'tuse it a lot, but it does seem to work fine.regards,DaveOn 8/15/06, David Thomas
- Original Message -
From: Martin Joseph [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, August 16, 2006 7:50 AM
Subject: [asterisk-users] Re: New Device
On 2006-08-15 13:10:05 -0700, Dovid Bender [EMAIL PROTECTED]
said:
This is a multi-part message in
If the phone already had the SIP image running.
Check the SIPDefault.cnf file there may be a phone_password= string
this is the phone's current password use it
remember to change to number or uppercase if need be
Ferguson, Michael wrote:
Maxx,
Thanks much for the feedback. I will check
Hi,
I would like to test them with asterisk and sip. Could anybody send me
Telco systems BATM GW-232 sip firmware? Does anybody have experience
with BATM products?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
Hello all,
We have a pathetic legacy PBX that produces the most terrible SIP
INVITE packet. In the past we have found a phone that can hope and
just used that. We now want to connect the legacy PBX to asterisk, and
we're (well, I'm) having problems.
This is the INVITE that's sent to the
David and Barry,
Thanks for the help.
'preciate it.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Fawthrop
Sent: Wednesday, August 16, 2006 6:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco
I'm not a guru,but it could help if you post the dialled number as well as your extensions.conf.On 8/16/06, Nils Rasmussen
[EMAIL PROTECTED] wrote:
We make a call from a PBX through asterisk to ISDN
(E1) Denmark.
We use asterisk to record incoming and outgoing calls
Sometimes we
Hi,
In my zapata.conf, I have the following lines
signalling = fxs_ks
context = fromfxs
channel = 1
When there is an incoming Zap call at Zap channel 1, the context fromfxs
is entered
and the entry s extension in the context is executed.
Would it be possible to jump to a particular
rc.local:
touch /var/lock/subsys/local
setpci -v -s 00:1f.1 LATENCY_TIMER=4
setpci -v -s 02:0e.0 LATENCY_TIMER=4
setpci -v -s 0b:07.0 LATENCY_TIMER=4
setpci -v -s 0c:08.0 LATENCY_TIMER=4
setpci -v -s 10:0d.0 LATENCY_TIMER=0
setpci -v -s 06:02.0 LATENCY_TIMER=ff
sleep 5
echo UnLoading wct4xxp
Manually config to point to your boot server, which should have a
good copy of the software and it should go get it. If not sniff the
traffic in/out and see what it IS doing.
I have had several firmware updates get interrupted in the past
corrupting the image and this has always worked.
I receive the following error in the Asterisk console when I try to
execute the Page() application:
WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page'
for extention (intercom, *, 1)
EXTENSIONS.CONF
[Default]
Exten = *80,1,Goto(intercom,s,1)
[intercom]
exten = s,1,Answer
Dovid Bender wrote:
I have spoken to some one who is interested in investing into building
equipment for asterisk. I am looking to find out what products that
the asterisk community would like to see be built. This can be
products that already exists but lack certain functionality as well as
IAX based outboard device with FXO and/or FXS that overcomes the many
shortcomings of the IAXy
Should also support pulse dial, server address expressed in other than
an IP, and all the good features of the SPa 2000 and 3000
JMO
John Novack
Dovid Bender wrote:
I have spoken to some one who
Just a quick note that Edvina in cooperation with Digium is starting
the fall season of trainings again.
Coming trainings are:
* Asterisk Bootcamp, Boston - next week!
We still have a few seats available
* Asterisk Beachcamp, Malaga, Spain
A class in a beach hotel in beautiful Malaga on
[EMAIL PROTECTED] -
There will be a lot of Asterisk-related activities at Voice On the
Net FALL - Von - in Boston.
Apart from Digium booth (#819), there will be Asterisk presentations
as well as developer meetings.
For the [EMAIL PROTECTED] agenda, see http://www.pulver.com/asterisk/
-
Hi there;
Did you load the respective module?
Regards;
LK
On 8/16/06, Dennis P. Clark [EMAIL PROTECTED] wrote:
I receive the following error in the Asterisk console when I try to
execute the Page() application:
WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page'
for
What is the module I should be loading and how do I load it?
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leonardo
Kamache (Gmail)
Sent: Wednesday,
*lol* The cryptic replies have been exactly my problem as well!
-Original Message-
From: kjcsb [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 16, 2006 12:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk
-Original Message-
From: kjcsb [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 16, 2006 12:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk
Absolutely. The SER/OpenSER documentation is terrible, and if
Hello
I am curious as to what hardware folks are using
successfully from HP or DELL. I will likely be running just a quad span T1
card with the system.
I appreciate your input.
Thanks,
Dave
___
--Bandwidth and Colocation
How
did you find out about 468*??? It's sure as poop not documented in the Polycom
Admin Guide anywhere.
-Original Message-From: Dovid Bender
[mailto:[EMAIL PROTECTED]Sent: Tuesday, August 15, 2006
11:16 PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject:
I just got done implementing this on a Realtime system and it works
flawlessly. You need to create a macro named page that you call from
the dialplan. Please refer to the wiki for more details:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
Good luck!
Joe
Dennis P. Clark
Dennis P. Clark wrote:
I receive the following error in the Asterisk console when I try to
execute the Page() application:
WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page'
for extention (intercom, *, 1)
What version of Asterisk are you running?
Doug
--
Ben Franklin
Anyone tried any softphone on Cingular 8125 or HTC Wizard?
On 8/16/06, Rajeev Natarajan [EMAIL PROTECTED] wrote:
http://www.electronicscience.com/ has a good IAX2 softphone called ESC Softphone
On 8/16/06, David Thomas [EMAIL PROTECTED] wrote:
Sorry, poor reply. Yes I use it on WM5, and have
I've had an HP Proliant DL-360 G2 (Pentium IV - 3 GHz with 1 GB memory)
in production for a couple of years with 20 Cisco phones and a single
T-1. The load is generally nill. I've never seen the load over 1.0, its
usually more like 0.1. It is also recording (in WAV format) all inbound
calls
Yeah, that's exactly the problem that I am having here (also switched to
g729 and gsm).
However, Teliax has told me that the g726 issue is with the * 1.2.10 release
and as a result not an issue with their service. I'm not entirely convinced,
but since we also use g726 for some of our internal
On Wed, 2006-08-16 at 10:57 -0400, David Sampson wrote:
Hello –
I am curious as to what hardware folks are using successfully from HP
or DELL. I will likely be running just a quad span T1 card with the
system.
HP DL380 G4, 4GB mem, 2x 146GB U320 in RAID1, dual hotswap PS
HP DL360 G4, 2GB
1.2.10
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Wednesday, August 16, 2006 11:10 AM
To: Asterisk Users Mailing List -
Salve *!
see:
http://linuxdevices.com/news/NS8030785497.html
It is based on a dual-core Marvell (formerly Intel) XScale processor
clocked at 312MHz[and] The Greenphone's baseband processor/modem
is a Broadcom BCM2121.
I think the BCM chip is for the GSM stuff, for GUI and applications
the
I was wondering which of these cards would be better for a 1-2
line SOHO. I would like room to grow as well
as I am concerned with voice quality and life expectancy of the product. Any
input into which one and why would be greatly appreciated.
Thanks,
Jon
I would get the same error when trying to use sftp. Switching to ftp
eliminated the problem.
Curt Shaffer wrote:
I posted earlier about an application not found error. I have manually
pointed the phone at the server but it just does not seem to ever even
hit it. I am going to do some network
On Wed, 2006-08-16 at 11:14 -0400, Curt Shaffer wrote:
After loading the application successfully on other phones I get
config error 0x4020 and it just keeps rebooting through this whole
process.
Which version of the bootrom does the phone have?
I have been told that once you upgrade to
Once in a while, the same questions are asked in this mailing list
about Unicall and MFCR2. I wrote a document in spanis about 2 months
ago about MFCR2 signaling and how to debug it with testcall. I have
translated the document into english per users request, and made some
other improvements.
Hi Tzafrir
I'm still testing so I start asterisk manually after boot. You are right
that 99 would normally be too late but as long as zaptel + modules are
loaded first with a lesser boot sequence number and is running when
asterisk starts on boot up it shouldn't matter where it is done in the
I'm running SVN-trunk-r16869M (compiled 2006-04-01) with iax trunks to
five remote systems most of which are v1.2.10. No problems with any of
those trunks using g726.
Teliax is the only system that I've had any issues with using iax and
g726. I've not tried sip to them and don't have any
Dennis P. Clark wrote:
1.2.10
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
-Original Message-
Dennis P. Clark wrote:
I receive the following error in the Asterisk console when I try to
execute the Page() application:
WARNING[24360]:
in the CLI do:
show applications like page
if you something there then you have it loaded, otherwise do:
load app_page.so
if that fails my guess is you need zaptel loaded first.
On 8/16/06, Dennis P. Clark [EMAIL PROTECTED] wrote:
1.2.10
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
the Goto command is your friend
On 8/16/06, Chan Kwang Mien [EMAIL PROTECTED] wrote:
Hi,
In my zapata.conf, I have the following lines
signalling = fxs_ks
context = fromfxs
channel = 1
When there is an incoming Zap call at Zap channel 1, the context fromfxs
is entered
and the entry s
thanks CF,
I did change the PRI CAUSE to unavailable, or reject.
only that it still shows
Accepting overlap call from.
just before this -Executing SetVar(Zap/12-1, PRI_CAUSE=27)
does anyone knows if this call being picked up at anytime?
Problem is, this is a reverse charge
Jonathan Borden wrote:
I was wondering which of these cards would be better for a 1-2 line
SOHO. I would like room to grow as well as I am concerned with voice
quality and life expectancy of the product. Any input into which one
and why would be greatly appreciated.
The sangoma a200d does
The A200 is a far better card. More forgiving of Motherboards, MUCH more
expandable, slightly lower cost. Only real drawback is modules are in
pairs, so if you want 4 FXO, you need to buy 4. It expands to 24 ports
using one PCI slot
Also, if you ever are rich, hardware echo cancel is an option
I have setup a Quad T-1 on a Dell 850 but it trunks all calls to a main voip server that is a Dell 2850. We chose to use a Sangoma T-1 card to side step some of the possible problems with motherboards/Dell servers that Patrick mentioned.
On 8/16/06, Patrick [EMAIL PROTECTED] wrote:
On Wed,
On Wed, 2006-08-16 at 00:22 -0700, Crazy Boy wrote:
extensions.conf file contents:
[incoming]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten =
Hi Eric,
I'm working in a small call center, but with special requirements. We
currently have a couple of clients, all of them have
specific phone numbers configured in our system, so when we get a call
for a specific client we take down the information via a webpage
then it sent via email to
Hi!
I have connected my analog phones to an asterisk box with sipura spa2002
devices.
I can do an attended transfer by taking the call which should be
transferred, pressing the flash button, dialing the number to which the
call should be transferred and now i can hang up or talk to the person
who
The call is not being picked up.
Manrique Feoli wrote:
thanks CF,
I did change the PRI CAUSE to unavailable, or reject.
only that it still shows Accepting overlap call from. just before
this -Executing SetVar(Zap/12-1, PRI_CAUSE=27)
does anyone knows if this call being picked
Has anyone ever tried to run multiple instances of Asterisk on a single system,
running each with a different username, and each in a separate base directory?
Something like /home/pbx/business-1, home/pbx/business-2 etc?
Did it work? I assume for every service that Asterisk runs, on each
Sangoma has a better PCI interface so no interrupt or
compatibility issues like you get with the Digium card. Sangoma will also
upgrade the card to a version with hardware echo cancellation if you cannot
solve your echo problems with the software echo cancellers. I believe you
send the card
Doug,
Note: Don't take this email serious, I'm just messing with you, but it
sure as poop is ;).
In version 1.6.x released 18th of July 2005 in section 2.2.1.4, Reset
the Factory Defaults
"To perform this function on all phones except the IP4000,
simultaneously press and hold 4,6,8 and *
Anyone help :-(
I did find one but s I said it only had png's and
xml's in it
Thanks
- Original Message -
From:
Paul A Brown
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, August 15, 2006 9:41
PM
Subject: [asterisk-users] 7970 SIP
I have two asterisk (trixbox) connected by IAX2 Trunk. Both of them have
interfaces TE205P configured and working fine.
I can places calls to PSTN on both of them. I can place calls from SIP
phones connected on asterisk one, using the IAX2 Trunk, to SIP phones
connected on the asterisk two.
I
You might be able to use virtual NICs to eliminate the problem with non-standard ports for a company's SIP phones. Or real NICs using a couple of multi-homed cards.I haven't tried it, though.
On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Has anyone ever tried to run multiple instances of
On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc?
Did it work? I assume
There is a good page on the wiki about this:
http://www.voip-info.org/wiki-Asterisk+non-root
CP
On Aug 14, 2006, at 6:44 PM, Forrest Beck wrote:
Does anyone have a listing on file/directories that asterisk needs
ownership of to run as a user other than root?
I know about the major items ---
Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Wednesday, August 16, 2006 11:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 1.2.10 - g726 Issues
I'm running SVN-trunk-r16869M
Is it possible to use Asterisk RealTime and also config files (like
sip.conf) at the same time?
As much as I know, only one thing can be used and I need them both
working!...
Thanks,
Ricardo.
___
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set group is your friend:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup
On 8/16/06, Damien Gabrielson [EMAIL PROTECTED] wrote:
I have been trying to restrict incoming calls for some time and I have
not had any luck yet so I hope someone may have done this already.
I
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Douglas Garstang wrote:
Has anyone ever tried to run multiple instances of Asterisk on a single
system, running each with a different username, and each in a separate base
directory? Something like /home/pbx/business-1, home/pbx/business-2 etc?
Douglas Garstang wrote:
Has anyone ever tried to run multiple instances of Asterisk on a single system,
running each with a different username, and each in a separate base directory?
Something like /home/pbx/business-1, home/pbx/business-2 etc?
Did it work? I assume for every service that
I just got my first IP601 and put together my first * system (yay!)
I have the first 2 buttons set up to be for the extension for the phone.
I was wondering how I could make the remaining 4 into speed dials?
IE: label button 3 Sales mgr and have it dial extension 246.
TIA,
Warren
Well,
we're talking about several dozen, maybe 100, companies, per Asterisk box
here.
-Original Message-From: David Freeman
[mailto:[EMAIL PROTECTED]Sent: Wednesday, August 16, 2006 11:36
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re:
Yes.
On Wed, 2006-08-16 at 19:03 +0100, Ricardo Carvalho wrote:
Is it possible to use Asterisk RealTime and also config files (like
sip.conf) at the same time?
As much as I know, only one thing can be used and I need them both
working!...
Thanks,
Ricardo.
Dell PE 1800s are our standard build. They
are tower or Rack capable, have 3 open slots for expansion (2 if you get the
remote access card). They are big though (5U) which is both a good and a
bad thing. Good in that they have GREAT air flow inside the system so
there is rarely any concern
You beat me to it Matt. =)-brandonOn 8/16/06, Matt Riddell (NZ) [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-Hash: SHA1Douglas Garstang wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a
Excellent! I'm not sure how I missed this before.
Thanks,
Damien
C F wrote:
set group is your friend:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup
On 8/16/06, Damien Gabrielson [EMAIL PROTECTED] wrote:
I have been trying to restrict incoming calls for some time and I
-Original Message-
From: Matt Riddell (NZ) [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 16, 2006 12:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 'Hosting'
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Douglas
I usually run the RV series of router for this. Much better thoughourput on
the VPN. Remember these low end devices can usually only handle about
1Mbps - 3Mbps of encryption max depending on the unit. Other than that, I
have had up to 8 behind a VPN such as this. I do generally recommend
-Original Message-
From: Jeremy McNamara [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 16, 2006 12:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 'Hosting'
Douglas Garstang wrote:
Has anyone ever tried to run
Doug,I'd suggest using contexts, but then having two servers for redundancy also. That way, if one asterisk box goes down, you don't have 50-100 clients completely down.-brandon
On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Well,
we're talking about several dozen, maybe 100,
Then virtual would be the way to go...I'm no expert, so you'd have to do some research on how many virtual interfaces you could use reliably.But some of the other suggestions I've seen might be a better option? Separate contexts for each entity, etc.
On 8/16/06, Douglas Garstang [EMAIL PROTECTED]
I trying to setup a outbound trunk with IPSmarx. It's working, but when I make
a call, the ring dialtone stills ringing on my side, even after the other side
picksup the phone. I got a NOTICE message from Asterisk that I hope you can
help me:
-- Called [EMAIL PROTECTED]
On Wed, 2006-08-16 at 19:03 +0100, Ricardo Carvalho wrote:
Is it possible to use Asterisk RealTime and also config files (like
sip.conf) at the same time?
As much as I know, only one thing can be used and I need them both
working!...
Yes, you can use both at the same time. The
Greetings,
has anyone ever set up Asterisk and Speakeasy VOIP? It uses a Motorola
VT1005 - any luck with this?
TIA
-Mike
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Nitin Gupta wrote:
Hi,
did anyone try do load-testing on asterisk, for sip channel calls?
I want to have a rough estimate about - how many calls, an asterisk server,
running on say dual 240 opteron with 1 GB memory, can handle?
Also how much internet bandwidth does a typical call requires?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Douglas Garstang wrote:
Well, we're talking about several dozen, maybe 100, companies, per Asterisk
box here.
Surely all the more reason to do it with contexts than instances.
- --
Cheers,
Matt Riddell
You could use Xen on Fedora Core 6 and virtualize each instance if you feel the need is there.On 8/16/06, Douglas Garstang
[EMAIL PROTECTED] wrote: -Original Message- From: Matt Riddell (NZ) [mailto:
[EMAIL PROTECTED]] Sent: Wednesday, August 16, 2006 12:06 PM To: Asterisk Users Mailing
and each in a separate base directory? Something like
/home/pbx/business-1, home/pbx/business-2 etc?
Use VPSs, like www.openvz.org
Pablo
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To UNSUBSCRIBE or
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Douglas Garstang wrote:
Because Asterisk wasn't designed with carrier class features in mind. It was
designed for a single enterprise. The dialplan, and config files, start to
get very very complicated after you add more than a few companies.
I tried to do sip over vpn with with a linksys router handle just one
phone. When I tried it, it worked fine. Once i shipped it out we had
all types of problems.
at first it was fine, then 1 out of 5 calls would sound like cell
phones. Now I can call him be he can't hear anything. Everything
Brandon,
Thanks. We're a litle past that stage of complexity. I'm just
throwing the question out there because it's becoming obvious that trying to
provision hundreds of customers on a cluster of Asterisk systems is going to be
very hard to manage.
-Original Message-From:
Actually, because there's no documentation, I don't have anything that I can
use.
-Original Message-
From: Dovid Bender [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 15, 2006 12:54 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re:
Hello people, I am having some issues with my new SIP provider.
The sip provider gives me only an IP address to configure my sip account,
since they do allow by IP address and not by username password.
This all configuration appears to work well, since I can originate a call
and it will ring the
-Original Message-
From: Moises Silva [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 15, 2006 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Manager Interface API's
Douglas. Please take this as a constructive comment. I have
Use a virtual private asterisk system. You'll be happier if you did.
http://www.telephreak.org/papers/vpa/
Has anyone ever tried to run multiple instances of Asterisk
on a single system, running each with a different username,
and each in a separate base directory? Something like
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Douglas Garstang wrote:
It's obvious that Asterisk was designed more for the enterprise (ie a single
company), rather than for the carrier (ie multiple companies). It's a bit
hard to explain here, but even with more than a few companies, the
Douglas Garstang wrote:
Well, we're talking about several dozen, maybe 100, companies, per
Asterisk box here.
Ok - And the problem is?
Jeremy McNamara
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-Original Message-
From: John Novack [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 15, 2006 10:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Manager Interface API's
I, for one, didn't take his comment as anything other than
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