Anyone has an idea what happend to voip-info? it stopped working about 24
hours ago.
Nir S
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:[EMAIL PROTECTED] On Behalf Of Philipp
Kempgen
Sent: Thursday, January 18, 2007 12:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX call limit
Nir Simionovich wrote:
Stupid and silly question - is there a way to limit the number of
concurrent
Hi All,
Stupid and silly question - is there a way to limit the number of concurrent
calls an IAX client can make? something in the similar sense of incominglimit
and
outgoing limit on SIP?
Regards,
Nir S
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Hi All,
Had anyone got an idea of there exists an LDAP backend for SIP and IAX?
I've read that there is a patch for LDAP realtime, but I hadn't seen any
type of
relevant configuration information.
Any information on the above would be highly appreciated.
Regards,
Nir S
directions. More then that, when running tcpdump, it
appears as if asterisk isn't even sending any RTP to the outbound SIP gateway.
This was seen on both 1.2.10 and 1.2.11
--
Kind Regards,
Nir Simionovich
Chief Technology Officer
Atelis PLC
not sending RTP
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Nir Simionovich wrote:
Hi All,
Here's a funny bit of a problem. I've got an asterisk server which appears
not to be sending any RTP out of the system. Any ideas why such a weird issue
would arise?
I've tested this scenario
:[EMAIL PROTECTED]] On Behalf Of Nir Simionovich
Sent: Sunday, September 03, 2006 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk not sending RTP
Hi Matt,
I'm dumping only the eth0 interface, as this is the only interface configured
not sending RTP
Nir Simionovich wrote:
Any ideas anyone ?
Do you have a compatible codec?
What does the SDP show?
Is sip.conf binding to a valid IP address?
Jeremy McNamara
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To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, September 3, 2006 4:45:08 PM GMT+0200
Subject: Re: [asterisk-users] Asterisk not sending RTP
Nir Simionovich wrote:
Sep 3 10:06:00 VERBOSE[6124] logger.c: Capabilities: us
:
http://lists.digium.com/mailman/listinfo/asterisk-users
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Kind Regards,
Nir Simionovich
Chief Technology Officer
Atelis PLC
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Title: RE: [asterisk-users] Inbound Calls SIP/2.0 404 Not Found
Hmmm...
Appears as if the SIP invite request is ill-formed. Can you send the SIP debug
of the session to the list, so we may examine it?
Nir S
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL
Hi all,
I'm experiencing weird cutoffs on TE110P. All cut offs are pre-seen with
an indication 5 coming from the PRI. I've talked to the telco, and they
indicated that they don't see any issues.
I've also modified the sync source to be the telco, and that didn't
solve the problem either.
Well, the documentation states that Video Conferencing is possible. I've
tried working with EyeBeam, which yielded nice
Results, but anything beyond that - I can't comment.
Nir S
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dakota
Sent: Sunday,
Well, the documentation states that Video Conferencing is possible. I've
tried working with EyeBeam, which yielded nice Results, but anything beyond
that - I can't comment.
Nir S
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dakota
Sent: Sunday,
Hi All,
I've been asked by a prospective client if Asterisk can is compliant
with v5.1 and v5.2 - which I
never heard about till today. After trying to figure out what I'm
dealing with, it appears as some kind
of signaling protocol, run on E1 lines.
I was wondering if anyone has more
Hmmm...
I feel that this is a little unfair towards GrandStream and other like
vendors. Any vendor on the market has issues with their firmware, I can
list many:
Sipura/LinkSys SPA 841 (Latest firmware):
1. Phone doesn't re-register upon network loss
2. Phone firware becomes stalled, without
phones is total crap. Anything rising even slightly above that level
wins awards for excellence. :-)
Steve
Nir Simionovich wrote:
Hmmm...
I feel that this is a little unfair towards GrandStream and other
like vendors. Any vendor on the market has issues with their
firmware, I can list many
can't, I have no control over the extensions. I basically interconnect via a
PRI to an external Avaya CTI system, thus, I have no way of implementing queues
in the system - due to constraints by the Avaya CTI system.
Regards,
Nir Simionovich
indicates that something else is
wrong.
So, anyone has an idea of what's going on here? Or better yet, a proposed
course of Action?
Regards,
Nir Simionovich
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Title: Accepting Inbound SIP Connections
Hi
Roger,
Can you please send a 'sip debug' output, so we can see the actual SIP trace of
the messages ?
Nir
S
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roger
JohnsenSent: Tuesday, November 29, 2005 12:30 AMTo:
Check the location specified in the kernel Makefile, and validate that is
installs the modules to the propler /usr/lib/modules/bla blabla directory.
Nir S
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kong
Sent: Thursday, November 24, 2005 9:01 AM
To:
Hmm...
I've also had some issues with choppy sounds, but my situation is somewhat
weird.
I've disabled APIC completely on the box, so not /proc/interrupts looks like
This:
CPU0 CPU1
0: 18394300 0 XT-PIC timer
1: 2 0 XT-PIC
Hi All,
I've recently encountered a very funny problem, which wasn't happening in
the past. I will describe this in detail:
During the past 4 weeks, our production Asterisk box had been experiencing
PRI (E1 lines) slips over and over at random intervals. When digging into
the available
to disable this device , for
example USB.
George
At 10:42 AM 2005-11-15, Nir Simionovich - CTO wrote:
Hi All,
I've recently encountered a very funny problem, which wasn't
happening in the past. I will describe this in detail:
During the past 4 weeks, our production Asterisk box had been
Hey Mark,
Looks like I'll be heading your way soon in that case ;-)
Nir S
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Ackroyd
Sent: Tuesday, November 15, 2005 1:11 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
again what IRQ the card gets. If you see that the card shares an
IRQ with another device try if its possible to disable this device , for
example USB.
George
At 10:42 AM 2005-11-15, Nir Simionovich - CTO wrote:
Hi All,
I've recently encountered a very funny problem, which wasn't
happening
DiscussionSubject: Re:
[Asterisk-Users] E1 PRI slips on TE410P
The other possible way to change IRQ is to change the PCI slot. Why
don't you run zttool and check for
missing interrupt? Also zttest could be a
good idea to check for E1 slip frames
On 11/15/05, Nir
Simionovich - CTO [EMAIL PROTECTED] wrote
Hi All,
What you say is fairly news to me, as I've been using Intel based boards
for the past 3 years with Asterisk - and hadn't had any issues that relate
to any of your concerns. I've had several issues with HyperThreading but
Other than that, nothing else major.
I've used both desktop
Hi Joe,
Well, as a practice, we
use Intel board based servers in our company. Small servers are usually based
on Pentium 4
while bigger ones are based on XEON. In terms of compatibility, I've used
Torypine's, Buckner's, ClearWater
and
the latest VolcanoPeak boards, all exhibited nice
Hi Samy,
Well, I've ran into the same issue a while back, and ended up solving it
by using an AGI script to
Actually performa a manager based originate with variables. This was I
actually dialed a specific extension
In a specific context, with pre-loading the variables I needed for the
target
Well,
I had an issue with them charging funds on PayPal
for stuff they never sent out, and they justsat on their hands for 3 months
till I contacted them to get a refund back (took me some time to check my
paypal), and then it took them 3 weeks to refund me.
Nir S
From: [EMAIL
Hi Marco,
As far as I can recall, the IBM setup utility can enable you to change the
IRQ of the SCSI controller.
In addition, I've never seen a WildCard board bound to IRQ7 on any box,
which is very weird in it self.
I'm flying over to Ireland today (actually, at the airport right now), and
Hi All,
I've been observing a very odd behaviour of Asterisk, when
relating to SIP connections.
Here's the scenario:
Ast1 is an Asterisk box originating calls via a predictive
dialer
Ast2 is an Asterisk box connected to 3E1 circuits
Ast1 originates calls to Ast2 via SIP, in order to
Hi all,
One of my clients had been sending me issues with G729 codec by
Digium. According to him,
the Digium codec is able to send calls into a Cisco AS54xx and AS53xx
gateway via SIP, however,
when calls are originated from the AS, asterisk using Digium G729 is
unable to receive the call
Ok,
Here's a bad story about using CVS head:
I have a client using Asterisk as a predictive dialer. The dialer
originates calls to people via a Zap channel,
and awaits input from the users. Problem is: the DTMF's are never picked
up. Now, when I tried using
the stable branch - it worked.
or a full T1?
On Monday 13 June 2005 21:14, Nir Simionovich wrote:
do a little math (23+1)*64 = 1536kbps = 1.536Mbps, hence the speed for
a single T1 circuit.
Your math's a little off.
T1 = 24 8-bit channels + 1 frame bit sent 8000 times a second.
24*8 = 192+1 = 193 bits sent 8000 times
Damon,
I have no idea where you are getting your information from, but what
you said makes no sense.
DSL based lines, be it ADSL or SDSL, are based upon a connection
technology in the ATM
family. As a result, the upstream and downstream of the connection can
be controlled seperately.
If
Marcelo Pacheco wrote:
SDSL has symmetrical speeds and full duplex communications.
Of the widely deployed lan/wan technologies, the only one I know of that is
half-duplex is 802.11{b,g}.
802.11b/g are standards used in wireless (Wi-Fi) connections, there is
no relation to the symetrics or
Hi David,
You are correct, I always get those 2 confused. Thanks for the clearing.
Nir S
David Coulson wrote:
Nir Simionovich wrote:
Now, E1 and T1 lines are based upon a channel based connection, which
means you get a line
with X number of data lines and a single control/signalling
Well,
I think you are asking the wrong question here, I think the proper
question would be:
In a 20 extension iPBX environment, what combination of signaling and
codec would provide the
best performace on a hardware of [specify your hardware here]?
Nir S
mr. barker wrote:
As the topic
If required, I'd be more than happy and willing to let voip-info.org be
hosted on my hosting server.
We are currently hooked up to the net with a 6MB symetrical connection,
and it should be enough
for voip-info. In addition, I can perform a daily incremental back to
it, in the same manner I
Cool, so you have satisfied yourself that you are licensed to use the
G.729 codec and not get your ass sued by the IP holders. Now you can
simply use the no-license-required codecs from here...
http://kvin.lv/pub/Linux/Asterisk/
I've tried using these codecs in the past, but usually ended up
Hi All,
I'm trying to connect to a SIP carrier who never connected with
Asterisk.
I
managed to connect with a sipura phone or a grandstream, no
problem.
When I configure asterisk, I'm able to send out calls to the
carrier no problems,however, receiving calls doesn't work, and I keep
nirs]type=friendhost=dynamicnat=nocanreinvinte=nousername=nirssecret=nirscontext=nirs
[bveraz1]type=friendhost=62.244.xx.xx
nat=nocanreinvite=nodisallow=allallow=g729
And just for knowladge, I do have the g729 licenses installed on the
box.
Any thoughts on the issue would be highly
appreci
Sounds like a wonderful idea!
I can tell you from personal experience that the performance of Asterisk and
its stability are in a one-to-one relation to the hardware that you're
using. We've been using mostly Intel boards for Asterisk, mainly the
ClearWater (XEON) and TorryPine (P4) boards for
Hi All,
I was wondering if anyone had been experimenting with Asterisk Video over
SIP support? I've tried making it work with Huawei Video phones, which
worked nice but tended to give poor performance when working on the WAN.
I wasn't able to make the Huawei interop with the X-Ten EyeBeam
Hi There,
I didn't understand your question completely, so I'll answer 2 questions:
Q. Can a PBX be connected to a Cisco ATA 186 box, where the ATA box acts
as
the PSTN connection?
A. In general terms, the answer would be YES. The FXS ports on the ATA
actually require that you
Hi All,
Has anyone installed multiple TDM cards on the same box? I'm trying to run
such a configuration
With [EMAIL PROTECTED], and it fails for some reason. Any pointers ?
Nir S
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Hi All,
I'm trying to install 2 TDM400x cards on the same [EMAIL PROTECTED] box, and I've currentlyhaving
issues where one card is identified by ztfcg, and the other isn't at all. Any
idea what
i
may be doing wrong here? has anyone got an [EMAIL PROTECTED] working in such a
manner?
Nir
Hi All,
Roy, I'm not entirely sure if you paid attention to what you wrote, as you
said: Last time I bought a sangoma card, I was given excellent support by
the reseller from .pl.
Now, this means that Sangoma has a local reseller that renders support for
Sangoma products. Most of the people
Hi Pepe,
You can't! As far as I can tell, once Asterisk eliminates an AGI upon
hangup, it doesn't send any signal information to the AGI script. If you
need to run some clean ups, the proper way to do so would be to execute
an AGI upon hangup, utilizing DeadAGI.
Nir S
-Original
Well,
It all depends what you want to do. We've already implemented a system
that can handle roughly 1000 channels of SIP using Asterisk. Of course we
used an Intel Cluster to reach that number, but the possibility exists.
It's all a question of design. I admit that using Asterisk would
You are correct, FastAGI is a valid option. However, if he's basing his
application on Asterisk Stable, FastAGI is not available in the stable
version.
Nir S
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stefan Reuter
Sent: Wednesday, March 16, 2005
: [Asterisk-Users] AGI kill
On Wed, 2005-03-16 at 13:08 +0200, Nir Simionovich wrote:
You are correct, FastAGI is a valid option. However, if he's basing his
application on Asterisk Stable, FastAGI is not available in the stable
version.
My version of Asterisk 1.0.6 includes FastAGI support and works
Those are the two valid methods. However, if you intend to generate many
calls, using the spool directory isn't a good method, as the spool is a very
slow means to do so. Using the manager proves more efficient for this task.
Nir S
-Original Message-
From: [EMAIL PROTECTED]
Hi Chetan,
Per my understanding to the chan_h323 operation, if no codecs are loaded,
Asterisk will perform a pass-through function, which means that signaling is
passed via Asterisk, but RTP passes between the endpoints.
This is NOT proxy function, as with proxy function means that RTP passes
To my knowledge, there is
no such formula. However, you can obtain a database
of the entire ITU E164 numbering plan at http://www.numberingplans.com, which
have
an updated database of all that information.
Nir S
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hi,
I think you are going the wrong way, let asterisk register all the calls,
and then simply query accordingly. In example, lets say you use the MySQL
CDR backend, after all the CDR's are in the DB, simply run:
'SELECT * from cdr where dialednumber like 9% order by calldate asc'
That should
I completely agree with Yair, especially considering the fact that we used
to share the same work place. It is one thing to endorse a platform, it's a
different thing endorsing your own platform in a coat of I'm a happy user.
Dimi Telecom also provides calling card platforms and various voice
://lists.digium.com/pipermail/asterisk-dev/2004-May/004151.html
It's not foolproof.
- Original Message -
From: Nir Simionovich [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, March 03, 2005 8:46 AM
Subject
/pipermail/asterisk-dev/2004-May/004151.html
It's not foolproof.
- Original Message -
From: Nir Simionovich [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, March 03, 2005 8:46 AM
Subject: RE: [Asterisk
Well,
I have no idea where the wiki is hosted, but if the wiki needs to be moved
to a more stable location, our hosting facility in Israel is as stable as
you can get. We have 2 circuit running in, BGP4 and an uplink of 4Mbps. I'm
confident it should be enough, no?
Nir S
- Original
Hi
All,
I've been fiddling around with the RealTime configuration. For SIP and IAX it's
really cool,
and the switch thing is cool too. But I've tried performing a GOTO from one
RealTime context,
to a second RealTime context. That didn't really work.
Any idea how to make it work ?
In general terms, all of the answers are correct and valid, but you need
to take into considerations a few more issues.
A. If you are utilizing SIP connections to the asterisk box, you need to
remember that SIP will generate traffic and I/O consumption even in idle
state. The more SIP
Did you put txfax in caller mode with the |caller parameter?
Regards,
Steve
Nir Simionovich wrote:
Hi all,
Ok, I've been bashing my head for a few hours now on this, trying to
figure out if I've
done something wrong, but everything seems to me hunky-dory. So here's
the deal:
1
:[EMAIL PROTECTED] On Behalf Of Steve
Underwood
Sent: Tuesday, January 18, 2005 2:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] spandsp and app_txfax
Did you put txfax in caller mode with the |caller parameter?
Regards,
Steve
Nir Simionovich wrote
,
Steve
Nir Simionovich wrote:
Hi Steve,
Well, I've tried that, and I admit there is an improvement. Now Asterisk
dials out and appears to be communicating with the Fax, however, the pages
come out blank. Is there some issue with Panasonic Fax machines ?
Nir S
-Original Message-
From
Hi
all,
Ok,
I've been bashing my head for a few hours now on this, trying to figure out if
I've
done something wrong, but everything seems to me hunky-dory. So here's the
deal:
1.
I've compiled the spandsp 0.0.2pre10 source code successfully and also the
asterisk
application
via SIP to the Asterisk Box at
Location B.According to a set of rules on the Asterisk box, the box would
route the SIP/H.323to one of the the other Cisco boxes, in order to
terminate the call.
What do you think, is this
possible?
Nir Simionovich
P.S.
Excuse me for the poor ASCII art, I
didn't
program that checks the originator and routes
according to pre-defined settings]
Again, this is just a concept, not the actual configuration. But in
theory, I don't a reason why this can't
work.
Nir Simionovich
- Original Message -
From:
Abdul
Hakeem
To: [EMAIL PROTECTED
at
Location B.According to a set of rules on the Asterisk box, the box would
route the SIP/H.323to one of the the other Cisco boxes, in order to
terminate the call.
What do you think, is this
possible?
Nir Simionovich
P.S.
Excuse me for the poor ASCII art, I
didn't want
this is often enough to give a gentle stream of
bit errors. Cat 5 cable is OK. The EtherNet plugs are OK. They just need
to be wired differently.
Regards,
Steve
Nir Simionovich wrote:
Hi All,
Well, as some of you alrady saw myself and Optimus on the IRC
channel, I guess you
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