Hi,
If u want to setup for 4500 or more phone then better to user OpenSER +
Asterisk.
OpenSER easily work for 10,000 calls.
You need to setup one server for OpenSER and all phone register on this
server. You need to write routing logic in OpenSER server to call connect
and if u need to play
I don't think there's anything inherently wrong with the bug tracking system.
It's more of a resource issue with many conflicting priorities. Officially
letting off some of the pressure from older branches does help. I would like
to be making faster progress through bug reports and
29 apr 2011 kl. 01.49 skrev Leif Madsen:
Well the issue is that we currently have over 900 open issues in the Asterisk
project alone, and with only one primary bug marshal (myself) sometimes things
accidentally get closed if it looks like a configuration issue.
What's the reason that we only
Le 29/04/2011 00:42, Russell Bryant a écrit :
- Original Message -
Sure. Please follow the 2 next stories:
- had a customer running 1.4.26 We upgraded to a new server and
installed 1.4.39, last version at this time. Bang: voicemail doesn't
work as it should, had to fallback to 1.4.26
28 apr 2011 kl. 16.53 skrev Russell Bryant:
- Original Message -
PS. Please don't start a discussion about 1.8 quality in this thread,
that's a separate issue. I just want to know what you think about
closing 1.4 support now. If you want to discuss 1.8 quality, start a
new thread.
Hi,
Can someone please recommend me the Hardware Server Configuration/8 or 4
port PRI Card to make Outbound Call at the rate of around 320 outbound
Calls/min ?
Thanks and Regards,
Kaushal
--
_
-- Bandwidth and Colocation
Thnaks a Lot.
So i will look for openser integration with asterisk!
[]'sf.rique
On Fri, Apr 29, 2011 at 3:12 AM, RAJNIKANT VANZA rajniva...@gmail.comwrote:
Hi,
If u want to setup for 4500 or more phone then better to user OpenSER +
Asterisk.
OpenSER easily work for 10,000 calls.
You
Yes I have it there, here the content of the file:
i think the code is buggy,
here is a comment from the function which generated the error
(ast_odbc_smart_execute in res_odbc.c line 155 )
/* This is a really bad method of trying to correct a dead connection. It
* only ever really worked with
True, we had setup before openser with asterisk and it works great. I
have wrote small document on voip-info related my project.
--
Sent from my iPhone
On Apr 29, 2011, at 8:23 AM, Henrique Fernandes sf.ri...@gmail.com
wrote:
Thnaks a Lot.
So i will look for openser integration with
Can you post later t he link for it ?
I read alot that page.
[]'sf.rique
On Fri, Apr 29, 2011 at 9:35 AM, Satish Patel satish...@hotmail.com wrote:
True, we had setup before openser with asterisk and it works great. I have
wrote small document on voip-info related my project.
--
Sent
On 11-04-29 02:59 AM, Olle E. Johansson wrote:
29 apr 2011 kl. 01.49 skrev Leif Madsen:
Well the issue is that we currently have over 900 open issues in the Asterisk
project alone, and with only one primary bug marshal (myself) sometimes
things
accidentally get closed if it looks like a
you running GSM FWTs with asterisk ?
On Mon, Apr 25, 2011 at 6:51 AM, Abid Saleem abid_aster...@hotmail.comwrote:
HI,
I am trying to setup a Class 4 termination setup using a kind of channel
hunting scenerio. I have some SIP DID numbers assigned from the local
telecom provider for
Don't expect lots of thing because I have just post my basic config
and method to integrate openser with asterisk and I did that 3 year ago.
http://www.voip-info.org/wiki/view/Asterisk+and+OpenSER+integration.
I would say search on google today lots of material are there and I
have
Thanks!
Would apreciate the book!
But i am already researching
[]'sf.rique
On Fri, Apr 29, 2011 at 10:10 AM, Satish Patel satish...@hotmail.comwrote:
Don't expect lots of thing because I have just post my basic config and
method to integrate openser with asterisk and I did that 3 year ago.
I have sent you book in PM.
-S
Date: Fri, 29 Apr 2011 10:39:56 -0300
From: sf.ri...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Multiple Asterisk
Thanks!
Would apreciate the book!
But i am already researching
[]'sf.rique
On Fri, Apr 29, 2011 at 10:10 AM,
could you send me book?
On Fri, Apr 29, 2011 at 9:48 AM, satish patel satish...@hotmail.com wrote:
I have sent you book in PM.
-S
--
Date: Fri, 29 Apr 2011 10:39:56 -0300
From: sf.ri...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re:
Thanks i got it
Another think you may know.
Openser have been forked into opensip and kamailio does you have anyidea
wich one is better ?
I guess i will start with opensips, becasue old openser.org point to there.
Thanks again!
[]'sf.rique
On Fri, Apr 29, 2011 at 10:49 AM, vip killa
- Original Message -
1) We have adopted peer code reviews as common practice for all
non-trivial changes going into Asterisk. This alone has _greatly_
increased the quality of the code going in. It is rare that a patch
goes up for review where someone doesn't point out some sort of
I never worked on kamailio but its pretty similar to OpenSER. I would say
OpenSIP would be good and on internet there are lots of comparison regarding
this topic.
One more thing OpenSER is pretty simple because in configuration its using SIP
messages. If you have good knowledge of SIP
Hi,
I have been getting reports phones ringing only a tiny moment and then going
to voicemail. CLI output shows:
-- SIP/user-0006fcdd is ringing
-- Got SIP response 400 Bad Request back from 23.23.23.23
-- SIP/user-0006fcdd is circuit-busy
== Everyone is busy/congested at this
Try to look in 'sip set debug peer user'.
On 29.04.2011 18:10, Mike wrote:
Hi,
I have been getting reports phones ringing only a tiny moment and then
going to voicemail. CLI output shows:
-- SIP/user-0006fcdd is ringing
-- Got SIP response 400 Bad Request back from 23.23.23.23
Hi,
There's a quite complex dialplan scenario and I found out that CDR of
main channel is flushed right after hangup on Local channel. I will try
to simplify my scenario:
[incoming]
exten = 555,1,Noop(do something before using local channel, fill some
variables, play IVR menus and so on)
same =
What I am looking for? Here is a snippet, with some info obfuscated. I can see
the bad request, but why there is such a message isn’t obvious.
--- SIP read from UDP:23.23.23.23:23725 ---
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK742ee2b8;rport
From: JOHN
Hey Matt,
I have download irc linux base CLI client and connect to irc.freenode.net i
can see bunch or channels but i didn't find any #asterisk or #asterisk-bugs
name. Am i looking at wrong place ?
*** #asterisk You're not on that channel
*** #asterisk Cannot join channel (+r) - you need to
Satish,
You must register your handle with freenode, because the asterisk
channel only allows registered people in.
http://freenode.net/faq.shtml#nicksetup
-M
On Fri, Apr 29, 2011 at 11:41 AM, satish patel satish...@hotmail.com wrote:
Hey Matt,
I have download irc linux base CLI client and
You're using 1.4.2. Why not try upgrading to a more recent release of 1.4 (I
believe 1.4.41 is current) and see if your issue has been resolved.
Thanks,
--Warren Selby, dCAP
On Apr 29, 2011, at 7:32 AM, Rizwan Hisham rizwanhas...@gmail.com wrote:
Yes I have it there, here the content of the
Now imagine that 1.4 stays at only security level. For first case we
have 2 options: upgrading for security reasons to last version but
then no more voicemail, or staying with 1.4.26. In the second case,
upgrading both servers to test with 1.8. If it's still not working, it
was time
To add another shilling to the pot -
Asterisk as a whole and 1.4 specifically is a very good product. Problems
are introduced (IMHO) when y'all take something that works perfectly well
and try to over-engineer it as a release bell-and-whistle instead of an
add-on. Voicemail and Multi-tenant
Hi,
I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf,
but its not yet production ready. Can someone please pitch in about HA
feature in Asterisk ?
(HA - High Availability.) Also, What would be the pros and cons of using
AsteriskNow over Asterisk ? Are the versions same in
For the High Availability part check out the HAAST add-on for Asterisk at
www.generationd.com
It detects a variety of failures, shuts down the failing system, starts
asterisk on the peer, moves the IP over, etc. Runs with every Asterisk variant
and every Linux distro. No special hardware
I thank everyone, for their fruitfull informations.
Regards,
Ashik Ali
On Fri, Apr 29, 2011 at 2:04 AM, Gilles codecompl...@free.fr wrote:
On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali
beaasteriskg...@gmail.com wrote:
Anybody can explain me why asterisk is unable to detect ringback tone
from
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