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
TCSANOW,&port_config );
ioctl(fd, TIOCMSET,&set_bits );
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 Kelly"
Subject: Re: [asterisk-users] Asterisk as a Condo door opener/int
:
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
er overall file
IO. I'd hate to see a four or five drive RAID 5 system rebuild if it had
> 100GB drives, a coat of paint probably dries faster.
-Ron
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re
that 60GB.
It'd take weeks 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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illary?
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rtc issues.
-Ron
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hing r-tarded like that.
-Ron
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t of hand.
-Ron
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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!
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s than a year, that someone
broke tor2. Is someone writing code for this thing with no hardware to
test it on?
-Ron
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refused to pay.
They did pay but ended 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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d you'll see it's not my original
thought. If your ISP lets you that is.
-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
>>> do
movie I paid
for. How you remain in business acting like a jack booted thug is beyond me.
-Ron
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through at least 1.6.1.
-Ron
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- September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
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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
>> >
ar.
This code is pre-Asterisk 1.0... It processes quite a few calls daily, I
have about 1,800 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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ded.
I have seen this in 1.4 and in the 1.6 betas. I also think 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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To
ESSAGE-
>> Hash: SHA1
>>
>> I would try make clean/make/make install.
>>
>> also add tor2 to the black list and remove it from any zaptel init stuff.
>>
>> Finally once your systems up (note asterisk wont be) try loading the
>> module with insm
le via SVN and there are some really cool
> and thoughtful features.
>
> Thanks a lot.
>
> --
> "I never look back darling, it distracts from the now", Edna Mode (The
> Incredibles)
>
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-Ron
http://www.m102.com
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ed there are some calls with
> > billsec greater than duration, duration is always 0 in those calls.
> >
> > How can this happens ? Am I missing something ?
> >
> > Tnx in advance
> >
> > Regards
> >
> > Edoardo Serra
> &
gt;>
> > Site came right up for me
> >
>
> Shush. You're not supposed to side with the spammer. :P
>
> That said, it doesn't work for me on either my work or home computers
> (both on different ISPs). DNS issue, perhaps?
>
> Jay
>
> ___
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 . An
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ld it restart at the moment when it was stopped with mpg123 /
madplay and at the beginning of a random sound with native mp3.
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pg123, madplay, mp3 native in
asterisk (asterisk-addons).
And the result is the same for all this tests: the music is stopped
and resumed after.
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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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ver 80 machines in our 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 Acc
ng to be obsolete before it's finished, so the wiki model makes
good sense; but it has to be structured better, at least a little bit
the way Wikipedia is operated.
-Stephen-
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om/hosting/colo/index.asp
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TRUNK
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:* [Aster
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
=mobileinnovations.com.au
canreinvite=yes
secret=
username=073...
annexb=no
disallow=all
allow=g729
[610]
type=friend
secret=
host=dynamic
callerid="James - Office" <610>
defaultip=192.168.1.230
disallow=all
allow=g729
[EMAIL PROTECTED]
port=5060
dtmfmode=auto
canreinvite=no
call-limit=1
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xt that Asterisk autogenerates
so 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 Use
a suitable asterisk driver
> (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 Mai
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 not
ou con 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 M
Linux/*BSD/Windows
Victorville, California PGP:0xE3AE35ED
It's all fun and games until someone starts a bonfire in the living room.
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et part or possibly on a PRI or
> whatever connected to your asterisk, over which you would like to dial
> out.
>
> BR
> Anselm
>
>
>
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>
> ast
dport= 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
situation ?
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on port 5091 . What am i doing wrong ?
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ancel out
(ctrl-c) it just drops.
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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-Commer
s), for the moment you should
> look into something else than Asterisk.
>
> Cheers,
> Jean-Michel.
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oot @ asterisk1.local on a i686 running Linux on 2006-09-27 19:41:58 UTC
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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
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: [ast
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
- Original Message -
From: "J. Oquendo" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, October 19, 2006 3:22 PM
Subject: Re: [asterisk-users] plainvoip - down ???
Joseph wrote:
Is plainvoip down?
I&#x
card in Slot 6 would 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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phone
that reports 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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aster
- Original Message -
From: "Steve Glaus" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, September 22, 2006 4:25 PM
Subject: [asterisk-users] Very high ping times from 7960 phones
I've asked this her
- 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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www.mobilcom.net
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e appreciated.
Thank you,
Nate Burke
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f the Asterisk 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, 20
risk1*CLI> show version
Asterisk 1.2.10 built 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?
_
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http://www.mobilcom.net
- Original Message -
From: "Tong" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, July 17, 2006 8:56 PM
Su
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computers are connected 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&#
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N
2 Sweex HFC-PCI cards
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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 do
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:-
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s the call to the correct server especially in a
multiple server environment (ex: dialing a missed call)
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List -
Non-Commercial Discussion
Sent: Friday, March 10, 2006 2:11
AM
Subject: [Asterisk-Users] Some ignorance
here,what exactly is a Session Control Border? (Verizonis asking me about
this)
Hey everyone,
I am working with Verizon to
terminate SIP calls for me.
MY
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 User
I haven't been through everything line by line but I did notice a new "Security
Configuration" where you can set an "Encrypt Key"
_
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http://www.mobilcom.net
- Original Message -
From: "Aaron Daniel" <[EMAIL PROTECT
- Original Message -
From: "Nabeel Jafferali" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Thursday, March 09, 2006 10:42 AM
Subject: RE: [Asterisk-Users] 7940/60 SIP 8.2
> So has anybody tried ins
ww.mobilcom.net
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