Hello
Le 04/12/2023 à 10:56, Benoit Panizzon (by way of Benoit Panizzon
) a écrit :
Hi List
We have some CPE which run an embedded asterisk 13 with chan_sip.
Unfortunately, when a registration is rejected, those stop trying.
I am familiar with pjsip which allows to configure
OK. I'm getting out the fireproof suit because it's coming and my
hackles have been raised by a number of comments on the list of late.
Disclaimer:
No disrespect intended to the individuals of any *specific* thread. I'm
a little frustrated over energy wasted on pedantic top/bottom posting
);
close(fd);
}
On 12/04/2011 8:16 AM, asterisk-users-requ...@lists.digium.com wrote:
Message: 3
Date: Mon, 11 Apr 2011 18:21:39 -0500
From: Don Kellyd...@donkelly.biz
Subject: Re: [asterisk-users] Asterisk as a Condo door opener/intercom
To: 'Asterisk Users Mailing List - Non-Commercial
:
http://www.asterisk.org/hello
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Hello everyone. I have a client with specific requirement, here's the
scenario:
Call comes in
Ivr menu, press 1 for new record 2 for existing 3 for operation blabla..
on pressing 1, list of 5 categories A,B,C,D .. when customer selects a
category a 4 digit pin needs to be generated
of core dumps before a blip would show up in df unless
it's constantly core dumping, which from what he said I suspect is not
the case.
-Ron
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if it had
100GB drives, a coat of paint probably dries faster.
-Ron
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?
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RE Kushner List Account wrote:
Paul Hales wrote:
In the order in which people normally read text they don't
repeat the entire conversation from the beginning each time
a question is asked either... Bottom posting is just as bad!
I am strongly against anyone posting
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, that someone
broke tor2. Is someone writing code for this thing with no hardware to
test it on?
-Ron
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up putting the general contractor out of business
in the end after the lawsuits.
It was one of the most impressive events I've ever seen.
-Ron
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like a jack booted thug is beyond me.
-Ron
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Alex Balashov wrote:
RE Kushner List Account wrote:
Doug wrote:
Why the BitTorrent guys want to give themselves
even a worse reputation is beyond me. We tell
our customers that they are not allowed to
download copyrighted material. But for other,
legal BitTorrent transfers, we
thought. If your ISP lets you that is.
-Ron
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..To Hell
Ann Arbor is quite a bit South of Hell. Actually it's been some time
since I've been to Hell but I'm sure it's frozen over today ;-)
-Ron
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.
-Ron
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- September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
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Tim Nelson wrote:
http://svn.digium.com/view/zaptel?view=revrevision=4121
Pure VoIP is the wave of the future!
Yeah, April fools...
-Ron
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Steve Totaro wrote:
On Thu, Mar 20, 2008 at 3:01 PM, RE Kushner List Account [EMAIL PROTECTED]
wrote:
Al Baker wrote:
Quote
This code is pre-Asterisk 1.0... It processes quite a few calls daily, I
have about 1,800 DID numbers pointed at it,
Are you SURE on that figure
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DID numbers pointed at it, there are several thousand
wrong number calls a day besides the traffic I send through it.
-Ron
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this guy
needs to recompile zaptel and look for errors in compiling.
-Ron
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Hello
Can anyone suggest sip phones with headset for use in call centers . They
should be fully inter operable with Asterisk over sip .
Thanks
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accept a support call about this issue?
I need to make sure this gets fixed.
Thanks,
-- Nick
On Mon, 18 Feb 2008, James Finstrom wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I would try make clean/make/make install.
also add tor2 to the black list and remove it from any
and thoughtful features.
Thanks a lot.
--
I never look back darling, it distracts from the now, Edna Mode (The
Incredibles)
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on either my work or home computers
(both on different ISPs). DNS issue, perhaps?
Jay
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that either :P ) .
On 09/08/2007, Stephen Bosch [EMAIL PROTECTED] wrote:
Mail list wrote:
Just got mail from them saying my NY DID will be deactivated in few days
. Funny thing is their site is still showing orderable DID's of same
area code . Anybody else got this ?
Wow. That is totally
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it was stopped with mpg123 /
madplay and at the beginning of a random sound with native mp3.
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Hi list,
I have a strange comportment of the MOH system on my asterisk.
When i respond to a call and after fews second i set this call in hold
mode the correspondent listen the music fine.
When i re-take my correspondent at T0 instant the music is paused. And
when i re-hold him at T60 (60 second
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Hello
Anyone here uses Vitelity as voip provider ? Their pplans looks good but i
need some feedback from existing customers if any here .
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datacenter, and none of them have ever
had a cataclyzmic failure. I'm not sure what the program with
voip-info is.
Just a note.. the address is voip-wiki.us http://voip-wiki.us, not
voip-info.us http://voip-info.us :)
On 3/15/07, *OCOSA List Acct. * [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote
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sense; but it has to be structured better, at least a little bit
the way Wikipedia is operated.
-Stephen-
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Anyone able to do this?
- Original Message -
*From:* Aaron Paxson mailto:[EMAIL PROTECTED]
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com
*Sent:* Friday, June 30, 2006 1:44 PM
*Subject:* [Asterisk-Users] Cannot get
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Is this site good ? They ship on time ? Any reviews on their card .
Thx
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did u try modprobe zaptel first ? also check makefile if ztdummy is marked
for compilation or not
On 19/02/07, Chris Blunt [EMAIL PROTECTED] wrote:
Hi List,
I am having some trouble with installing the latest version of ztdummy on
a CentOS Kernel 2.6 system.
I have installed a few
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there is nothing I can do in that context.
Has anyone else found a way to show that this a call returning and not a
new call coming in?
Thanks
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Had the exact same issue with the hardware canceller. If I set echocancel=no
then the problem goes away.
Very weird that the static only happens on our side and we only hear it.
- Original Message -
From: Webster, Andrew [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non
(the cards themselves support hardware gsm/g726 codecs,
for instance).
Alberto.
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Seems these two are closer to what's asked for.
http://www.vikingelectronics.com/products/view_product.php?pid=343
http://www.vikingelectronics.com/products/view_product.php?pid=217
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
What zap device do you have that encodes/decodes g729?
- Original Message -
From: O.Kamal
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, December 18, 2006 4:37 PM
Subject: Re: [asterisk-users] ZAP problem
Why do I need g729 license?, i am
until someone starts a bonfire in the living room.
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use multiple ports in this format port=5060,5061,5091 or
bindport=5060,5061,5091
That's what I have read
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Sales
*Sent:* Saturday, December 16, 2006 2:00 PM
*To:* Asterisk Users Mailing List
on port 5091 . What am i doing wrong ?
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in this situation ?
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= and it must be in [general]
Mail list wrote:
I am using a month old svn version of asterisk 1.2 . I have set
bindport=5091 for a sip peer ( type = friend) and nat=yes .. in sip show
peer it shows port 5091 for peer but asterisk isnt listening on port
5091 at all . I tried both port=5091 as well
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yum install subversion
On 09/12/06, Kovar Petr [EMAIL PROTECTED] wrote:
svn is application called subversion, you should download and install
it first.
- Original Message -
*From:* Ed Nuñez [EMAIL PROTECTED]
*To:* Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users
running Linux on 2006-09-27 19:41:58 UTC
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We've never used it in a load balancing situation
but it works great in a failover config.
- Original Message -
From:
Dean Collins
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, November 14, 2006 10:41
AM
Subject: RE: [asterisk
http://hotbrick.com/
- Original Message -
From:
Dean Collins
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, November 14, 2006 9:38
AM
Subject: RE: [asterisk-users] Dual Wan
Router with Failover
Are you looking for
load
Are you sure you have the phone setup to use TFTP
and not FTP?
- Original Message -
From:
Klaverstyn, David C
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, November 02, 2006 1:51
AM
Subject: [asterisk-users] Polycom 601
Phone can
- Original Message -
From: J. Oquendo [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 19, 2006 3:22 PM
Subject: Re: [asterisk-users] plainvoip - down ???
Joseph wrote:
Is plainvoip down?
I've
be ZAP/21-24 and Slot 5
would be ZAP/17-20
That would mean you need pairs 17-24 in your current config.
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- Original Message -
From: Steve Glaus [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, September 22, 2006 4:25 PM
Subject: [asterisk-users] Very high ping times from 7960 phones
I've asked this here before
the large time.
Now I'm thinking it has something specifically to do with 3com switches and
these phones.
_
Mobilcom
http://www.mobilcom.net
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- Original Message -
From: Christopher Corn
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, September 19, 2006 4:48 PM
Subject: Re: [asterisk-users] grandstream gxp 2000 does not display
nameswhencalling out
im sorry if im not being clear. when im calling
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hank you,
Nate Burke
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specific mailing
list:
http://www.trixbox.org/modules/newbb/
_Mobilcomhttp://www.mobilcom.net
- Original Message -
From:
Tom Vile
To: [EMAIL PROTECTED] ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, July 28, 2006 12:49
PM
by root @ asterisk1.local on a i686 running Linux on
2006-07-18 20:08:40 UTC
_
Mobilcom
http://www.mobilcom.net
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Are you using the Non-CallManager version?
_
Mobilcom
http://www.mobilcom.net
- Original Message -
From: Tong [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 17, 2006 8:56 PM
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risk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
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to the PC port behind the GXP-2000.Any suggestions?Thanks,DanielOn Jun 7, 2006, at 8:49 PM, list mail wrote:What do they do on the internet? Heavy surfing, large transfers, myspace. How are these units connected to the network? Are they passing through the same switch?I don't think it is the phones
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vegne af Olivier Krief Sendt: 7. juni 2006 16:18 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] I can hear only one way when I use nokia e-60withX-lite 2006/6/7, Jon Schøpzinsky [EMAIL PROTECTED]:Hello Be aware that the Nokia E60, E61 and E70 does
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List
Sent: Tuesday, 30 May 2006 12:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Asterisk receiving call from Panasonic TDA extension issue
Asterisk, Zap and Libpri version from Asterisk SVN-branch-1.2-r27093
Error:-
-- Accepting overlap call from '123' to '6
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server especially in a
multiple server environment (ex: dialing a missed call)
_
Mobilcom
http://www.mobilcom.net
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It sounds like they referring to a load balancer
that would provide one live external IP and handle sessions for multiple
internal servers.
_Mobilcomhttp://www.mobilcom.net
- Original Message -
From:
Gabriel
Afana
To: Asterisk Users Mailing List
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- Original Message -
From: Nabeel Jafferali [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, March 09, 2006 10:42 AM
Subject: RE: [Asterisk-Users] 7940/60 SIP 8.2
So has anybody tried installing the new
I haven't been through everything line by line but I did notice a new Security
Configuration where you can set an Encrypt Key
_
Mobilcom
http://www.mobilcom.net
- Original Message -
From: Aaron Daniel [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non
I believe they've done that the entire time. I've never known them to be real
supportive of competing third party solutions.
_
Mobilcom
http://www.mobilcom.net
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
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