Displaying 20 results from an estimated 28 matches for "simionovich".
2009 Apr 16
1
AGI Programming
...hought some of you might find it interesting.
Packt Publishing approached me a few weeks ago and asked if I would like
to review a book or two for them on my blog.
The first one they sent me is called Asterisk Gateway Interface
Programming and has only just been released. It was written by Nir
Simionovich.
You can read my review here:
http://www.theopensourcerer.com/2009/04/16/asterisk-agi-programming-with-packt/
They have also sent me a book on Trixbox CE 2.6 which I will get round
to reading and reviewing over the coming month or so.
Cheers
Alan
2007 Mar 14
3
What happend to voip-info?
Anyone has an idea what happend to voip-info? it stopped working about 24
hours ago.
Nir S
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2009 Feb 18
6
AGI pdf book
Dear Sir,
Can someone help me please to find a free ebook talking about AGI scripting
through asterisk?
Regards
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2003 May 03
0
* as a SoftSwitch/Router solution
...long the way, the transcoding of one of the channels got jipped,
or this crazy setup is too crazy to work (although, logic suggests that it
should)
Had anyone else conducted crazy tests like these? especially with 3rd party
vendors? I would really like to know the outcomes.
--
Regards,
Nir Simionovich
nirs@net-gurus.net
Net-Gurus.Net - Security by Design
2005 Jan 17
1
spandsp and app_txfax
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 associated with it.
2. Receiving a fax at asterisk works fine, at least appears to be working
2006 Dec 12
1
SIP and IAX configuration from LDAP
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
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2007 Jan 18
1
IAX call limit
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
2009 Jun 19
0
Asterisk and EC2 today at 12 Noon EDT
Nir Simionovich is about to become a father. He will be joining our
conference at 12 Noon EDT today from the Maternity Ward to talk about
Amazon EC2 cloud computing with Asterisk. Nir gave a very good
presentation on this at AMOOCON a few weeks ago (see
http://www.amoocon.de for more on that). The advantage here t...
2003 Apr 13
3
Recording Prompts
...one help me out to set this up ? or tell me who has done it in
the
past. If it is easier to do in Linux could someone tell me how ? i am
planning to use the linejack to re-direct calls or place voip calls.
Your help is appreciated.
Thanks,
Francisco
----- Original Message -----
From: "Nir Simionovich" <nirs@net-gurus.net>
To: <asterisk-users@lists.digium.com>
Sent: Friday, April 11, 2003 7:08 PM
Subject: Re: [Asterisk-Users] Recording Prompts
> Hi Omar,
>
> Well, I use a windows box to record WAV files, then sox in 8000
sampling
> rate to convert
> to GSM, th...
2005 Sep 26
3
IBM x306 - some progress
Hi,
I asked yesterday about a problem with x306 and IRQ sharing, didnt get
much info, now, i was playing with lspci, and see something strange,
lspci -v shows me the TDM400P card is on IRQ 7, and the SCSI card is
also on IRQ 7,
lspci -bv (from the man - b - shows "bus-centric view, as seen by the
BUS and not by the kernel) shows me the TDM400P is on IRQ 5, why does
the kernel puts it on
2017 May 30
0
Asterisk 13.16.0 Now Available
...--------------------------------
* ASTERISK-26088 - Investigate heavy memory utilization by
res_pjsip_pubsub
(Reported by Richard Mudgett)
* ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report
channel name with res_hep_rtcp when using chan_sip
(Reported by Nir Simionovich (GreenfieldTech - Israel))
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.16.0
Thank you for your continued support of Asterisk!
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2017 May 30
0
Asterisk 14.5.0 Now Available
...--------------------------------
* ASTERISK-26088 - Investigate heavy memory utilization by
res_pjsip_pubsub
(Reported by Richard Mudgett)
* ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report
channel name with res_hep_rtcp when using chan_sip
(Reported by Nir Simionovich (GreenfieldTech - Israel))
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.5.0
Thank you for your continued support of Asterisk!
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2005 Jun 05
4
Digium G729 licensing - is it worth the trouble?
I have been impressed with the quality and meagre bandwidth of the G729
codec from Digium. I am in a testing phase of our roll out, we are using 5
Asterisk PBXs in various countries to provide connectivity for our
employees, owners and family. As we are testing, and our setup is somewhat
complex due to the peculiarities of our connectivity, there has had to be a
lot of changes to servers, cards to
2005 Jun 10
19
Should I choose DSL @ 1.5 or a full T1?
I'm looking to expand my bandwidth for my Asterisk PBX.
Why should I choose a T1 over DSL for my asterisk server?
I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal?
Thanks
Bart
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2015 Feb 06
0
Asterisk 13.2.0 Now Available
...es_pjsip: Add user=phone option (Reported by
Matt Jordan)
* ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for
connection-oriented transports. (Reported by Matt Jordan)
* ASTERISK-24412 - [patch]Incomplete channel originate/continue
handling with ARI (Reported by Nir Simionovich (GreenfieldTech -
Israel))
* ASTERISK-24678 - [PATCH] Added atxfer* settings to
features.conf.sample (Reported by Niklas Larsson)
* ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported
by cloos)
* ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by...
2015 Feb 06
0
Asterisk 13.2.0 Now Available
...es_pjsip: Add user=phone option (Reported by
Matt Jordan)
* ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for
connection-oriented transports. (Reported by Matt Jordan)
* ASTERISK-24412 - [patch]Incomplete channel originate/continue
handling with ARI (Reported by Nir Simionovich (GreenfieldTech -
Israel))
* ASTERISK-24678 - [PATCH] Added atxfer* settings to
features.conf.sample (Reported by Niklas Larsson)
* ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported
by cloos)
* ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by...
2004 Nov 28
17
Wiki down?
Hi All,
The wiki seems to be struggling this evening. Anyone else seeing this?
Michael
--
Michael Graves mgraves@pixelpower.com
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc. mgraves@mstvp.com
o713-861-4005
o800-905-6412
c713-201-1262
2005 Jun 06
1
Issue with SIP inter-op
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 getting the following
messages:
<-- SIP read from 69.xx.xx.xx:5060:
INVITE
2005 Aug 24
6
GXP 2000 Firmware 1.0.1.2
Greetings all
Grandstream released a new firmware and it seems like the speaker phone
problem has been fixed. However we updated to firmware
1.0.1.12<http://1.0.1.12>to fix the echo problem but found other
problems were now
created. The worst of these new problems is that the whole phone starts
degrading, the volume starts getting lower and lower. The ringing starts
fading and the calls
2017 Dec 21
0
Certified Asterisk 13.18-cert1 Now Available
...(Reported by John Fawcett)
* ASTERISK-26088 - Investigate heavy memory utilization by
res_pjsip_pubsub
(Reported by Richard Mudgett)
* ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report
channel name with res_hep_rtcp when using chan_sip
(Reported by Nir Simionovich (GreenfieldTech - Israel))
* ASTERISK-26864 - res_pjsip_session: Add support for overlap
dialling
(Reported by Richard Begg)
* ASTERISK-26846 - chan_sip: Add rtcp-mux support
(Reported by Sean Bright)
* ASTERISK-23828 - pjsip - Need a command to list active SIP
subs...