I do not experience the same issues as below. I personally think g729 is
more then worth the $10 per licensed channel. I have 10 licensed
channels and they are worth every penny. The call quality is far
superior to gsm or g726 and is nearly identical to a g711 call. My
application is probably
I'll make my entry into this Flamewar BKW I believe the term is
pot, as there are many colors of felines... As to the content... None
of my business.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Monday, January 03, 2005 12:03 AM
I do about 500 calls per day on average volume and about 750 on heavy
volume and find it necessary to run a logger rotate every other day...
other then that I can go on for a couple weeks until I need a full
reboot.
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Subject: RE: [Asterisk-Users] Is asterisk that unstable
On Thu, 2004-12-30 at 09:50 -0500, Luke Catranis wrote:
I do about 500 calls per day on average volume and about 750 on heavy
volume and find it necessary to run a logger rotate every other day
Logger rotate from cli
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Randy
MacKay
Sent: Thursday, December 30, 2004 10:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Is asterisk that unstable
I
How much time did you waste on that?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Taylor
Sent: Sunday, August 22, 2004 10:24 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] Another
Xml services for cisco 7960, setup a broadcast stream. Check the wiki.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Svensson
Sent: Friday, December 17, 2004 3:28 AM
To: Wilson Pickett; Asterisk Users Mailing List - Non-Commercial
Discussion
Solar winds... free tftp
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
users
Sent: Thursday, December 02, 2004 10:18 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: cisco 7940 help
Hi,
I spent 3 days trying to upgrade my 7960's with
Get a vonage line unlimited for $24.99 per
month and get a softline and have vonage set it up as
a rollover line.
Grand total $34.98 per month.
Route outbound to the ZAP/VONAGE first,
then to the SIP/VONAGE NEXT, if need be setup an IAX termination account with voicepulse or iax.cc
Question?
Is it possible to pause during a dial statement?
The Issue I have is that I have to enter an IVR bypass code before I
enter the dialed number from my phone. The SIP client I'm sending my
calls too is either not getting the dialed number {EXTEN} or it's coming
through wrong...
Anybody?
Yes go restore it to a factory config and your all set...
Via the web interface.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Halverson
Sent: Wednesday, October 27, 2004 3:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Any tips, tricks or treats out there? I'm building a new system and
would like to move away from my SQL based call rating solution...
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To
http://store.yahoo.com/asteriskpbx/witd1pofxs.html
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of christophe de coninck
Sent: Thursday, October 21, 2004 1:17 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] modem
question
Hey,
I've got
Wondering if anyone could give me a tip on controlling access under the
following scenario.
I have an ATA connected to a legacy pbx as a trunk line. I want to
control who can make calls on this trunk. I cannot set restrictions on
the users via the pbx, so I would like to be able to assign a
,210573 or 052173 if in US),
and
then stripted the six digits before dialing. You only tell the people
you
want to be able to dial out the six digit code.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Luke
Catranis
Sent: 20 October 2004 15:55
The cnf files are in plain text, I don't know where you got the firmware
upgrades, but you should be able to get the SEPDefault.cnf and the
SEPMAC.cnf off the CISCO site. You also need the O97XX.txt or whatever
it is...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
:[EMAIL PROTECTED] On Behalf Of Luke
Catranis
Sent: Friday, October 15, 2004 8:35 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] sccp cisco 12sp HELP !!!
The cnf files are in plain text, I don't know where you got the firmware
upgrades, but you should
Nearly everyone of my clients is between 600 and 700 ms and there is no
choppyness to the calls. They are on VSAT uplinks with SCPC return channels,
but I would say that latency is certainly not your problem.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I've seen a couple posts about SMP, and I'm wondering if anyone has an idea
what I can handle with this box:
Dual P2 Xeon 450 MHZ with 2MB L2 Cache
1 GB PC100 SDRAM
RAID 5
Running SuSE 9.1 2.6 SMP Kernel...
Currently I have 16 users,
10 CISCO 7912G Using ULAW
3 Sipura 1000's Using GSM or g726
gafachi
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of lists-jmhunter
Sent:
Did you change the voicepulse context to user in you iax.conf?
This is what they told me to do when I had the same problem 3 weeks ago:
[voicepulse]
type=user
context=voicepulse-in
;auth=md5
;secret=mysecret
host=gw5.voicepulse.com
qualify=yes
Oops... mybad
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Luke Catranis
Sent
I tried to get it working, even opened several tickets with them, but
eventually hired a programmer friend to build something similar for me...
I'm sure with the proper programming and time you could get it working, but
it won't have nearly the same level of integration as a it would with a
CISCO
Yes in the voicemail.conf
Set the default to no and follow the instructions in the voicemail.conf file
i.e (not sure if this is exact
X = ,Joe Foo,[EMAIL PROTECTED],,||attach=yes
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Try voicepulse connect connect.voicepulse.com
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
I must do the same with the proxy... one note... the t stands for transfer
per the wiki: t : Allow the called user to transfer the call
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Dial
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Anybody use them... I signed up for $20 to see how there system works..
They're at $.02 per minute for US Termination and their other ITX rates
aren't too shabby.
Sadly my IAX registration is rejected... maybe a glitch, wondering if
anyone's had a similar issue.
Luke
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Luke Catranis
Sent: Tuesday, August 03, 2004 10:42 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Gafachi?
Anybody use them... I signed up for $20 to see how there system works..
They're at $.02 per minute for US Termination
Testing MTA VT1005, one specific issue, RFC3389 support incomplete, please
turn off on client... I 've tried all sorts of RTP settings (there's only 4
possible options). And no luck... Any help?
Luke
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Luke Catranis
Sent: Thursday, July 29, 2004 12:12 PM
To: [EMAIL PROTECTED
Set the minimum voicemail length in the voicemail.conf file.
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-Original Message-
From: [EMAIL PROTECTED]
The Scripts are in GSM format... locate *.gsm or look in /etc/sounds,
/usr/src/asterisk/sounds
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-Original Message-
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Luke Catranis
Sent: Thursday, July 29, 2004 2:43 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Voice mail problem
of assistance.
Luke Catranis
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I've successfully connected to my SQL server with tsql (freetds) and isql
(unixodbc). I've followed the wiki, and everytime I try and run * shutdowns
with a segmentation fault. Any help?
Luke Catranis
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