On Fri, 21 Nov 2008, Al Baker wrote:
Remember - You are going from a CARRIER GRADE purpose built piece of
hardware with Software built under a rigid CMM with extensive
soak-testing to software that has been developed under , shall we say,
a somewhat less rigid and stringent methodology.
You
Strongly suggest to consider a Freeswitch/OpenSER implementation instead.
Regarding purpose built and supported software.sometimes throwning
billions of CMM software development to a product does not guarantee a good
product... look at Micro$oft Vista.
E
http://Gpro.ws
2008/11/21 Yehavi Bourvine [EMAIL PROTECTED]
Hello,
Our university has to upgrade soon its old Nortel PBX's which holds
around 10,000 extensions tied to 5 PBXes. Up to now we thought about
commercial solutions but now there is a window openning for open source
solution. However, I need
Hello list,
I recently bought a Linksys SPA400 as a PSTN gateway. The gateway is
connected to
an * server and i have 10 users using this setup. I do have some
problems in establishing
a call to an outside location (call that goes through the SPA400). The
first attempt doesn't
get through.
I
server problem's
2008/11/21 Luis Morales [EMAIL PROTECTED]
Does any know what happens with svn repository on svn.digium.com ?
--
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
Thanks to everyone who replies so far!
We have Nortel PBX'es with a support contract from one of the local VARs
(Nortel does not give direct support here). In the last two weeks we had one
of our exchanges down for three half days; one was after a failure, and the
other two were when the
Jeffrey Phelps schrieb:
But how do I get it to run a script?? I don't have any SMDI Interfaces,
so I wouldn't be able to put anything in the config...
I thought Cisco CallManager had SMDI so that might have been an
alternative solution.
All I can tell is that the normal externnotify command in
Yehavi Bourvine wrote:
Thanks to everyone who replies so far!
We have Nortel PBX'es with a support contract from one of the local
VARs (Nortel does not give direct support here). In the last two weeks
we had one of our exchanges down for three half days; one was after a
failure, and
Hi all,
I have around 100 SPA2100 registered in my provider openSER.
I'd like to add an Asterisk registered into openSER, to the network, to
deploy voicemail service for those SPAs.
Due to administration access levels, I have no access to SER box, so I'm
wondering if that possible:
- Some
Hi Robert,
I'd suggest tweaking the Dial() arguments so that you (1) allow early
audio, (2) don't force it play ringing to the calling party, and (3)
modify any other options to be as relaxed as possible. if you make those
changes, you'll start hearing the operator message recordings and those
Here is a Dirty solution - create a PERL or other script to listen for
changes to voicemail DB/Dir. When VM is deleted, launch script to turn off
Cisco MWI (should be simple since you are turning on with script). Not
Best solution, just workable one. I'm doing similar thing with my VM - I
look
Just switching from Nortel to something else may not eliminate
hardware/software failures, or prevent those without experience from
pushing the enter key at the wrong time. You have to consider the two
professionals actually cost considerably more than just salary, due to
taxes, 401k, benefits
Jason Aarons (US) wrote:
Just switching from Nortel to something else may not eliminate
hardware/software failures, or prevent those without experience from
pushing the enter key at the wrong time.
One also has to keep in mind - Asterisk, like any large open-source
project, gets a lot
Danny Nicholas schrieb:
Here is a Dirty solution - create a PERL or other script to listen for
changes to voicemail DB/Dir. When VM is deleted, launch script to turn off
Cisco MWI (should be simple since you are turning on with script). Not
Best solution, just workable one.
Yeah. If all
Alex Balashov wrote:
Jason Aarons (US) wrote:
Just switching from Nortel to something else may not eliminate
hardware/software failures, or prevent those without experience from
pushing the enter key at the wrong time.
One also has to keep in mind - Asterisk, like any large open-source
Al Baker wrote:
Remember - You are going from a CARRIER GRADE purpose built piece of
hardware with Software built under a rigid CMM with extensive
soak-testing to software that has been developed under , shall we say,
a somewhat less rigid and stringent methodology.
You will be moving
Ping
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Hi Dan -
I found the maxusers defined in meetme.c, but I'm
not sure how this value is set. Does anybody know
if one can limit the number of users permitted in a
meetme conference? I know there's MeetmeCount(), but
I'd rather avoid the dialplan logic and just set
maxusers instead.
That
Due diligence is required on anything 10,000 people are going to be
pounding on. Undersizing is common,
I think due diligence is THE key with any open source solution,
including asterisk. I'll admit that I pretty badly screwed up one
asterisk installation because I didn't adequately prepare it
Noah Miller wrote:
With that many extensions, I'll second using a SIP registrar like
Freeswitch or OpenSer. Just use asterisk to provide the services.
Third.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Roderick A. Anderson wrote:
And if that ain't confusing I don't know what would be.
I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago
and ended up never using it. Passed it along to a friend who is having
some problems with it. (He isn't on this list.)
We've both
On 11/20/08, Steve Totaro [EMAIL PROTECTED] wrote:
On Thu, Nov 20, 2008 at 3:38 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Nov 20, 2008 at 08:25:54AM +0100, Olivier wrote:
2008/11/17 Philipp Kempgen [EMAIL PROTECTED]
Tilghman Lesher schrieb:
On Thursday 13 November 2008
On Fri, Nov 21, 2008 at 4:59 PM, Sebastian Milioto [EMAIL PROTECTED] wrote:
Ping
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On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller [EMAIL PROTECTED]wrote:
[..snip..]
With that many extensions, I'll second using a SIP registrar like
Freeswitch or OpenSer. Just use asterisk to provide the services.
Is Asterisk even needed?
- Gonzalo
Armed with a little more information, here is a more realistic reply.
In the 1.6.0.1 code, app_meetme.c defines maxusers in line 369 and sets the
max value in line 870 to 0x7fff.
Therefore changing line 870 would allow you to limit the maxusers.
-Original Message-
From: [EMAIL
Gonzalo Servat wrote:
On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
[..snip..]
With that many extensions, I'll second using a SIP registrar like
Freeswitch or OpenSer. Just use asterisk to provide the services.
Is
Noah Miller wrote:
and is only one of the roads that
leads to Hell (I prefer Patterson Lake Road myself since I drive in from
the North East).
Hmm. You must live near Ann Arbor.
No, northern suburbs of Detroit. M-59 to US-23 S to M-36 W..To S.
Howell St..Patterson Lake Rd..To
Hi all,
I upgraded from asterisk 1.2.23 and zaptel 1.2.19
to asterisk 1.4.18 and zaptel 1.4.12.1
I use polycom 501 phones internally.
Everything seems fine. I can pick up the phone and call out,
calls coming in work just fine.
The issue I see is when the system first calls me,
then calls
Hi,
I've read RFC3428 which presents SIP MESSAGE.
Is there any extension or encoding scheme working with SIP MESSAGE that
would enhance text display with blinking or underlining attributes ?
This could be useful to notify SIP hardphone users with some important
events such being in Do Not Disturb
On Fri, Nov 21, 2008 at 1:48 PM, Alex Balashov [EMAIL PROTECTED]wrote:
Gonzalo Servat wrote:
On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
[..snip..]
With that many extensions, I'll second using a SIP registrar like
Is Asterisk even needed?
Potentially, no. But if you intend to provide subscriber/PBX features,
it is needed as a UA feature box(s).
And FreeSWITCH can't handle that?
Freeswitch can provide many PBX features with additional modules, but
asterisk can provide more, and its implementations
Gonzalo Servat wrote:
On Fri, Nov 21, 2008 at 1:48 PM, Alex Balashov
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Gonzalo Servat wrote:
On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
Gonzalo Servat wrote:
On Fri, Nov 21, 2008 at 1:48 PM, Alex Balashov
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Gonzalo Servat wrote:
On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
You could trying changing this in sip.cfg
AES voice.aes.hs.enable=0
To
AES voice.aes.hs.enable=1
It's at line 324 in mine. Results not guaranteed.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Friday, November 21, 2008 10:28 AM
To:
You could trying changing this in sip.cfg
AES voice.aes.hs.enable=0
To
AES voice.aes.hs.enable=1
Just tried that - rebooted my polycom and still half audio.
Thanks,
Jerry
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On Fri, Nov 21, 2008 at 2:49 PM, Noah Miller [EMAIL PROTECTED]wrote:
And FreeSWITCH can't handle that?
Freeswitch can provide many PBX features with additional modules, but
asterisk can provide more, and its implementations of such items are
more time tested. One of freeswitch's big
You could try un-commenting duplex=2 in rpt.conf and changing it to
duplex=3.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Friday, November 21, 2008 11:05 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] upgrade
I know that in the past there have been people on this list who have
done very large scale asterisk deployments. Not sure if any of them
are still around to comment.
With that many extensions, I'll second using a SIP registrar like
Freeswitch or OpenSer. Just use asterisk to provide
Yehavi Bourvine wrote:
OK, but I still did not get a reply to my original question: Why using
SIP registrar in front of Asterisk and not simply use bare Astersik?
can't it handle the load? (remember - in my case it doesn't handle the
RTP, only signalling). Can't it handle so much
Hi,
VERBOSE[6120] logger.c: -- Got SIP response 500 Server Internal Error
I just noticed that i sometimes get those back from provider. They are
currently general SIP message log entries with verbose level 3.
I wonder if such SIP fails could generate at least WARNING in log?
Currently i'm
Alex Balashov wrote:
Yehavi Bourvine wrote:
OK, but I still did not get a reply to my original question: Why using
SIP registrar in front of Asterisk and not simply use bare Astersik?
can't it handle the load? (remember - in my case it doesn't handle the
RTP, only signalling). Can't it
Atis Lezdins wrote:
Hi,
VERBOSE[6120] logger.c: -- Got SIP response 500 Server Internal Error
I just noticed that i sometimes get those back from provider. They are
currently general SIP message log entries with verbose level 3.
I wonder if such SIP fails could generate at least
On Fri, Nov 21, 2008 at 7:32 PM, Alex Balashov
[EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
Hi,
VERBOSE[6120] logger.c: -- Got SIP response 500 Server Internal Error
I just noticed that i sometimes get those back from provider. They are
currently general SIP message log entries with
Atis Lezdins wrote:
On Fri, Nov 21, 2008 at 7:32 PM, Alex Balashov
[EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
Hi,
VERBOSE[6120] logger.c: -- Got SIP response 500 Server Internal Error
I just noticed that i sometimes get those back from provider. They are
currently general SIP
Hi,
I've started noticing these messages today myself specifically with
Broadvox.
Are you using this carrier or someone else?
Tom
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins
Sent: Friday, November 21, 2008 12:28 PM
To: Asterisk Users
Hi,
I noticed that my hint priority stops working when I add to many
extensions/channels. It looks like everything exceeding 80 characters is
discarded.
By stop working I mean the status is and stays Unavailable.
This works
exten = *1,hint,SIP/loicvoip1_1IAX2/loicvoip1_1SIP/loicvoip1_1_a1
On Fri, Nov 21, 2008 at 5:46 PM, Danny Nicholas [EMAIL PROTECTED] wrote:
Armed with a little more information, here is a more realistic reply.
In the 1.6.0.1 code, app_meetme.c defines maxusers in line 369 and sets the
max value in line 870 to 0x7fff.
Therefore changing line 870 would
On Friday 21 November 2008 09:42:12 Matt Florell wrote:
On 11/20/08, Steve Totaro [EMAIL PROTECTED] wrote:
You also have
people like Matt Florell who have continued to add functionality to
1.2 but since Digium won't take them, or the dev doesn't want to sign
over their first born,
Hey Everyone,
Here’s an email I received from a client who has a trixbox system that has
contracted with me for some custom dialplan programming.
While I was away at a conference on Tuesday, our server crashed same as before
(it was “responsive”
Matt Riddell wrote:
On 18/11/2008 9:46 a.m., Matthew Fredrickson wrote:
Singer X.J. Wang wrote:
We've had the same issue. For calls that go between a SIP connection
(desktop phones) and Zaptel connections, there was a lot of problems
with half duplex. We switched
from the Digium card to
What is the call volume on this box? Depending on the version of Asterisk,
maybe there are some memory leaks present causing calls to fail but everything
else to keep working?
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Gregory Malsack wrote:
Hey
Sounds like a deadlock. Its not a Trixbox issue but an asterisk one.
Your best bet is to upgrade asterisk itself. No need to reinstall
Trixbox. Just download the asterisk source and compile/install.
And yes, we have installed several Trixbox systems that became unstable
eventually. The
On 11/21/08, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Friday 21 November 2008 09:42:12 Matt Florell wrote:
On 11/20/08, Steve Totaro [EMAIL PROTECTED] wrote:
You also have
people like Matt Florell who have continued to add functionality to
1.2 but since Digium won't take
Thanks, Tzafrir, for your reply!
At 13:25 11/19/2008, Tzafrir Cohen wrote:
On Wed, Nov 19, 2008 at 01:14:55PM -0600, Doug wrote:
Hi folks,
I am building a new box. Want it to look
pretty much like an older Asterisk 1.2,
Debian box that is in production. The new
box will used as a
Just a guess, but since extensions.conf is basically a card file, there
may be a 80 character limit to the line or data size.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Loic Didelot
Sent: Friday, November 21, 2008 11:52 AM
To:
Loic Didelot schrieb:
I noticed that my hint priority stops working when I add to many
extensions/channels. It looks like everything exceeding 80 characters is
discarded.
By stop working I mean the status is and stays Unavailable.
This works
exten =
I am using an AGI to setup the call to the first person,
then jumping into the dialplan with some Variables set.
Is the AGI messing up my channel???
My dialplan at that point looks like:
exten =
call_cont,1,Dial(${CONT_CALLAT},${CONT_DIAL_TIMEOUT},${CONT_ONHOLD}tT)
CONT_CALLAT is
On Fri, Nov 21, 2008 at 10:42 AM, Matt Florell [EMAIL PROTECTED] wrote:
On 11/20/08, Steve Totaro [EMAIL PROTECTED] wrote:
On Thu, Nov 20, 2008 at 3:38 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Nov 20, 2008 at 08:25:54AM +0100, Olivier wrote:
2008/11/17 Philipp Kempgen [EMAIL
On Fri, Nov 21, 2008 at 10:42 AM, Matt Florell [EMAIL PROTECTED] wrote:
just keep in mind that in
my opinion the 1.4 tree did not become usable until 1.4.18 when most
of the major bugs were finally fixed.
The longer you drag out the adoption curve, the longer it will take for
1.6 to catch
On 11/21/08, Alex Balashov [EMAIL PROTECTED] wrote:
On Fri, Nov 21, 2008 at 10:42 AM, Matt Florell [EMAIL PROTECTED] wrote:
just keep in mind that in
my opinion the 1.4 tree did not become usable until 1.4.18 when most
of the major bugs were finally fixed.
The longer you drag out
Thanks for the reply
Could you be a little more specific?
Thanks
Robb
Martin Smith wrote:
Hi Robert,
I'd suggest tweaking the Dial() arguments so that you (1) allow early
audio, (2) don't force it play ringing to the calling party, and (3)
modify any other options to be as relaxed as
Hi Robert,
I'd recommend the following options for Dial() so that you corroborate
operator messages w/ cause codes:
1. remove R and r - we've found this can supress operator recordings on
early audio
2. likewise, remove m to disable MOH
Also, check the values of DIALSTATUS to compare to
Just curious but why would you want to have a lot of devices all have
the exact same state information?
Philipp Kempgen wrote:
Loic Didelot schrieb:
I noticed that my hint priority stops working when I add to many
extensions/channels. It looks like everything exceeding 80 characters is
Hey, all. When I last was heavily into Asterisk (1.0.x), setting up to
receive faxes was, well, a PITA, what with having to patch the Asterisk
install with various driver patches and this, that, and the other.
Is that still true? Is there a fax HOWTO out there that reflects Asterisk
1.4.x?
Did anybody tried MozIAX extension? It is Mozilla IAX2 soft-phone.
http://moziax.mozdev.org/
I tried it yesterday on eee pc, connected to asterisk on local LAN and the
performance is terrible!
The delay is about 2sec or 3sec. and very bad echo.
I think it is the implementation of their IAX2 in
Yehavi Bourvine wrote:
OK, but I still did not get a reply to my original question: Why
using
SIP registrar in front of Asterisk and not simply use bare Astersik?
can't it handle the load? (remember - in my case it doesn't handle
the
RTP, only signalling). Can't it handle so much
I've looked at doing various things to chan_sip to improve signaling
performance (hash tables for call lookups, etc.) I gave up when I
realized that the overhead of handling the RTP was so far above the
overhead of processing SIP signaling that it didn't really matter
much. The only reason
Hi,
I'm having 2 problems:
1) MOH in realtime is not working, I have configured it but never go to
look at the database, no warning or error found and I can do a query using
realtime and the family from the cli.
2) I have SIP phones via realtime, if I register one of them and a
Hi all,
Is it possible to have * playing an mp3 file in the way old tape system
worked?
Sincerely,
Robert Augustyn
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My second problem is resolved, qualify=yes did the trick.
I'm still having problems with MOH
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Sebastian
Enviado el: Friday, November 21, 2008 9:09 PM
Para: asterisk-users@lists.digium.com
Asunto: [asterisk-users] MOH Realtime
Date: Fri, 21 Nov 2008 16:20:28 -0600
From: Terry Wilson [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Large Asterisk installarions (~10,
000
extensions), preferably at universities
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
I've taken the liberty of starting a new thread to discuss the design
of the Asterisk CDR mechanism. The discussion has been kindly
initiated by murf putting together a proposal:
http://svn.digium.com/svn/asterisk/team/murf/RFCs.
After reading the proposal I still don't think it's the right way
Hello All
One of our client Bank has 900 employees working in different locations.
They need to record all internal and external calls. Can any body suggest
Call Recording Solution for this requirement. We need to know the Hardware /
Bandwidth and all requirements and costing.
Regards,
--
Hi All,
I am searching about asterisk IM message passing with eyebeam.
but i am not able to send instant message to another registered users.
i am working in asterisk 1.4 branch.
i have tested within call and without call but there is no message recieved.
and every time i got error user not found
During compiling I have not seen an error, however, when I start
asterisk again it ends with:
app_morsecode.so = (Morse code)
== Registered custom function 'SYSINFO'
func_sysinfo.so = (System information related functions)
Segmentation fault (core dumped)
How can I figure out what is wrong?
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