On Thursday 09 Jun 2005 23:45, Andrew Kohlsmith wrote:
On Thursday 09 June 2005 13:15, Bob Goddard wrote:
The Via processors emulate the i686 just fine. The problem has always
been with GCC.
Got some proof of that? It's generally regarded as common knowlege in
these circles that the via
Hi
Everyone,
Is it possible to
have a SIP Phone work remotely if it's behind a Router performing NAT without
connecting the Router to a VPN? The Asterisk Box will be in the
DMZ.
Thanks
Dan
CYTEXONE
Dan Levine
[EMAIL PROTECTED]
CYTEXONE | Your Technology Specialists
I would be willing to Pay $500 for a good Asterisk / Exchange
Integration
Dan Levine
[EMAIL PROTECTED]
CYTEXONE | Your Technology Specialists R
877.CYTEXONE x 810
212.477.0990 x 810
212.208.6889 FAX
502 Laguardia Place, Suite 2B
New York, NY 10012
http://www.cytexone.com
James H. Thompson wrote:
Voip-info is back up -- in-spite of Murphy's law.
This was phase I (install latest version of O/S) of an upgrade to
improve performance and functionality.
Hopefully with Phase II we will see much better performance and new
functions.
For those that asked, the
I've made some modifications to zttest in order to use
it as a frame clock accuracy tester / slip detector.
I'm not certain if that was it's original purpose, but it
seems that a lot of folks try to use it that way.
The result is something that I'm calling ztclock for now
to help avoid confusion.
There's an update to Firefly on Virbiage
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
lots of bug fixes - see if that helps
-Adam
Paul Redstone wrote:
Hi
We've recently set up and are using with success 1.0.7 using a Junghanns
quadbri card to BRI ISDN, and Firefly with
I am trying to use swift in PHP/AGI.
function swift($text, $escape_digits='', $frequency=8000, $voice=NULL,
$fnameIn='')
During swift speaking some text I want the caller to be able to press
1, 2 or 3 to do thing 1, thing 2 or thing 3.
How are these digit defines and then caught?
Thanks,
Hallo,
You are nearly right. We are working with some artists and they have many
funny ideas with Asterisk. Regarding my question, the fact is that we can do
this technically with any PC: you play a music file with RealPlayer and at
the same time another music file with Winamp...So theoretically,
Hi Michiel,
-Original Message-
Since you already have done something on this, can you tell
us what your plan was?
Complex :) ENUM was a part of a larger setup concerning roll-out of voip
technology over wireless networks.
Do you already have some docs about what to do and why, or
Good day all
Is there someone who's got a sirrix 4 port working in NT mode
I got one working good in TE mode.
Apparently I must add 8 jumpers in make the cross cable a straight cable
But what about the sirrix.conf? Do I just change the mode from TE to NT?
Please Help or advice?
Thanks
Altus
That is the entire package as it was submitted to us from Grandstream.
Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:[EMAIL PROTECTED]
-Original Message-
From: Peter Svensson [mailto:[EMAIL PROTECTED]
Sent: Friday, June 10, 2005 1:46 AM
To:
On Fri, 10 Jun 2005, The VoIP Connection wrote:
Have you received an updated tftp config template as well? We
asked for and received one with a 1.0.1.9 early beta version.
That is the entire package as it was submitted to us from Grandstream.
We requested and received the template
I did that once on a cheap linejack card. Took a week to get the smell
out of the office, and the bright orange from inside the server was
quite interesting :) Only took 1 second to start a small flame going,
but fortunately I cought it quick.
I wonder if the zaptel cards have any kind of
On Thu, 2005-06-09 at 16:00 -0400, list wrote:
according to RFC's your required to have reverse lookups on ur mail server,
so blocking based on this is perfectly legitimate.
My ISP has the option of reverse lookups, I still get blocked by some
other ISPs :(
--
Dave Cotton [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Dave Cotton
Sent: Friday, 10 June 2005 5:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ATTN: Keith
On Thu, 2005-06-09 at 16:00 -0400, list
Hello,
Does Asterisk support G.729AB and does anyone know how to
enable G.729AB codec? s it free?
Thanks for your interest.
Erdem HAKI [EMAIL PROTECTED]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi,
I have install asterisk and it works fine.
But when I use Sjphone and I use Ethereal a Client send Request:OPTIONS
sip:obelix.foo and Server answer Status: 404 Not found.
But i can talk with two client and asterisk.
When I use Xlite i don't have this request it's clean.
I don't
Good day all
We installed a box a long time ago and they bought g729a licenses
Now we want to upgrade and reinstall,whats going to happen with the
codec,if I give the box the same ip as always will it work?
Please Help
___
Asterisk-Users mailing list
See this:
http://lists.digium.com/pipermail/asterisk-users/2005-June/110524.html
Free for non-commercial use.
- Original Message -
From: Erdem HAK [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, June 10, 2005 11:53 AM
Subject: [Asterisk-Users] G.729AB codec support
hi all
i have * box with 4 ext using sj phone, i wanna try to
make a conference. i am using ztdummy and look fine
when i install it because there is no erros message. I
checked with lsmod the zaptel and usb-uhci using
ztdummy. but why i still get error says no
application meetme ...
here is
That is completely wrong, the intel code might be free for non
commercial use, but you will still need a license to operate the g729,
whoever wrote the code.
The cost for 1 channel is 10$, and you can buy the only legal codec from
digium (www.digium.com).
Zoa.
Soner Tari wrote:
See this:
I'm saying free for non-commercial use, you're saying Intel is free for
non-commercial use. And I point to the Intel code. And there is no fee for
the licence for non-commercial use.
So what is completely wrong about my post?
- Original Message -
From: Zoa [EMAIL PROTECTED]
To:
Im saying that the code is only an implementation of g729.
The intel sources clearly states that you need a license for g729, not
from intel but from the g729 patent holder.
Zoa.
signature.asc
Description: OpenPGP digital signature
___
On Friday 10 June 2005 05:09, altus wrote:
We installed a box a long time ago and they bought g729a licenses
Now we want to upgrade and reinstall,whats going to happen with the
codec,if I give the box the same ip as always will it work?
Please do a modicum of research, hell even contact the
I understand what you're saying, but I am not the one who makes the
decisions. That decision is made already, so since I am actually getting
your point and I agree with that, the only thing I can try to do right
now, is try to avoid having Cisco Unity in the other 3 offices. I would
love to
On Friday 10 June 2005 02:28, Olle E. Johansson wrote:
I would like to use this moment to say a big THANK YOU from the
community to you and Commpartners for providing this resource to the
community...
I agree; while I personally dislike wikis I can't deny (as is evidenced by all
the posts
On Friday 10 June 2005 04:08, Terry H. Gilsenan wrote:
Received: from source ([81.56.129.44]) by exprod5mx8.postini.com
([64.18.4.10]) with SMTP; Fri, 10 Jun 2005 00:29:16 PDT
Your MTA claimed it was called SOURCE but rDNS tells the recipient MX
that it is called: mail.linuxautrement.com
I
On Friday 10 June 2005 02:15, Dan Levine wrote:
I would be willing to Pay $500 for a good Asterisk / Exchange
Integration
What do you consider good Asterisk and Exchange integration? More than a
handful of words, please.
-A.
___
Asterisk-Users
Hello all,
I've a little clue with zaphfc used to connect to a BRI linethat probably can
be a configuration issue (really I hope so)
Here, telcos (expecially mobile operators) use to substitute the dialtone with
some vocal indication without answer the line. (Indications like The
customer
On Friday 10 June 2005 02:07, Bob Goddard wrote:
The Via C3 processors lack the CMPXCHG8B (CMOV) instructions and I
assume others which are listed in the Intel documents as being
optional. GCC assumes that they are always there.
Look at http://radagast.bglug.ca/epia/epia_howto/x1098.html,
Thank up very much for the response
Its appreciated and it will help me allot
I hope u have a nice Monday or is it Friday?
ALtus (the early-morning BOER!)
On Fri, 2005-06-10 at 06:05 -0400, Andrew Kohlsmith wrote:
On Friday 10 June 2005 05:09, altus wrote:
We installed a box a long time ago
sylvain garcia wrote:
Hi,
I have install asterisk and it works fine.
But when I use Sjphone and I use Ethereal a Client send Request:OPTIONS
sip:obelix.foo and Server answer Status: 404 Not found.
But i can talk with two client and asterisk.
When I use Xlite i don't have this request
Title: Message
Hi,
I use
SIP softphone that is not registered at Asterisk.
When I
dial some extension defined in the dial plan ([EMAIL PROTECTED])with my SIP softphone,
Asterisk will not ask me for username/password (will not return response 407) as
I expected.
The
response 407 -
I would like to support these plans for exchange/outlook integration with at
least $250 as well.
Please have a closer look at http://www.click-and-call.com/ .
Mediastreams has developed their product e-phone, which we could test a
couple of months ago. Their Outlook Integration is really great:
-
For some reason, this didn't go through the first time, maybe because I
had JUST signed up.
Hello,
I'm trying to configure Asterisk and my Handytone 488 to pass incoming
calls coming over PSTN through the FXO port to Asterisk, which will
process the calls with voicemail, or some such service.
I
Hi,
I have recently
found a bug when using Steve Underwood chan_unicall with Asterisk 1.0.x
(including 1.0.8RC)
When you place a
call from a SIP phone with dtmfmode=rfc2833 or dtmfmode=inband through MFCR2 via
chan_unicall all goes well until you press a dtmf key. When you do this,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Kohlsmith
Sent: Friday, 10 June 2005 8:16 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] ATTN: Keith - Seriously OT
On Friday 10 June 2005 04:08, Terry H. Gilsenan
Hello,
In feature list I see that asterisk supports call redirect feature(as this is
basic PBX feature :)).
I am trying to implement this feature on my sip phones (avaya 4602). The need is
to enable some feature access code for example *40 so, that user can dial it
and redirect all calls to other
On Friday 10 June 2005 07:34, Terry H. Gilsenan wrote:
Your server your rules, however in this day of increasing trojan SMTP
engined boxes, you should expect to get les and less deliverability.
I fail to see how a reverse pointer that == forward record means a more
reliable message. How many
If required, I'd be more than happy and willing to let voip-info.org be
hosted on my hosting server.
We are currently hooked up to the net with a 6MB symetrical connection,
and it should be enough
for voip-info. In addition, I can perform a daily incremental back to
it, in the same manner I
Title: Mensaje
Hi
all,
I have to
interconnect Asterisk with a Siemens HIPATH 3750. In siemens we can configure
ECMA-QSIG Master, ISO-QSIG Master,Point to Point link withCRC4 and
Point to Point link withouthCRC4): Siemens has BNC connector.
I use a balun with BNC and RH45
connectro. I
It sounds like there are quite a few people willing to aid in
bandwidth for voip-info. I was just wondering if it wouldn't make sense to
mirror the site across several locations with a round-robin DNS for a little
bit of load balancing? Any thoughts?
Chris Coulthurst
[EMAIL PROTECTED]
I've made some modifications to zttest in order to use
it as a frame clock accuracy tester / slip detector.
I'm not certain if that was it's original purpose, but it
seems that a lot of folks try to use it that way.
The result is something that I'm calling ztclock for now
to help avoid
On Fri, 2005-06-10 at 05:35 -0700, Chris Coulthurst wrote:
It sounds like there are quite a few people willing to aid in
bandwidth for voip-info. I was just wondering if it wouldn't make sense to
mirror the site across several locations with a round-robin DNS for a little
bit of load
Hello again,
I relaized that older version of Asterisk supports g729 ( Pass-thru
only unless g729 license obtained - in any case I want). Do you know that latest
[EMAIL PROTECTED] or CVS version provide us g729 pass-thru options?
Thanks for your interest
Erdem HAKI [EMAIL
Darn, and here I was thinking small town Melbourne,
FL, USA =(
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
jurgenSent: Thursday, June 09, 2005 11:16 PMTo:
Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and
Business-Oriented Asterisk
If you get this working, please let me know -- I'm testing out the same
situation, using [EMAIL PROTECTED], and have 3 SIP phones -- one softphone
on a Samsung i700, one Avaya IP Phone and one softphone on a PC.
The latter two are behind NAT and the i700 softphone is not, but I
can't originate an
Michael...
I don't believe that PHPAGI supports this currently. What you are looking
for is a combination of 2 functions: get_data() and swift().
PHPAGI code is very easy to follow so build your own function to do what you
want and add it to your copy of PHPAGI.php. Ain't OSS wonderful?
I
We have Cisco Callmangler V4 in one office and several * servers in others, we
use a SIP trunk out of the Cisco and it works perfectly.
Peter
-Original Message-
From: Simone [mailto:[EMAIL PROTECTED]
Sent: 10 June 2005 10:15
To: Asterisk Users Mailing List - Non-Commercial
WHy not do it for free. Start the 'VoIP documentation project' on
sourceforge. It provides bandwidth, filesystem for images and all, php,
Erk! My vote is against Sourceforge, definately -- although it's free,
you get what you pay for. Clumsy interface and *shockingly* slow load
times.
I have many old channel banks around that I would like to
use to generate analog extensions. Will most channel banks work with Asterisk?
Dave
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I am trying to patch the latest release version of Asterisk (1.0.7) with
SpanDSP(0.0.2pre18). It seems that the Makefile for Asterisk was
revamped since SpanDSP was released and the patch file that comes with
SpanDSP for adding rxfax.c and txfax.c no longer work. I am not
familiar with how
I use client Sjphone which work fine but i have Sniff a traffic..
- Sjphone send packet with OPTIONS to Asterisk
- Asterisk ask with 404 not found
This sequence come back often in my log.
I don't understand why Sjphone Sens OPTION, and 404 not found..
Thanks for your help
Hi,
I am looking for a softphone (sip or iax) that works in Solaris/SPARC
with sunray100 terminals. I found iaxcomm but it doesn't work. Also I am
trying sip-communicator but I have several errors from JMF/RTP.
Does anyone have a softphone working over this platform? which one? I
don't care
On Friday, June 10, 2005 3:16 AM, Andrew Kohlsmith
[SMTP:[EMAIL PROTECTED] wrote:
On Friday 10 June 2005 04:08, Terry H. Gilsenan wrote:
Received: from source ([81.56.129.44]) by exprod5mx8.postini.com
([64.18.4.10]) with SMTP; Fri, 10 Jun 2005 00:29:16 PDT
Your MTA claimed it was
-Original Message-
From: Gregory Wiktor - ADCom Corp. [mailto:[EMAIL PROTECTED]
I did that once on a cheap linejack card. Took a week to get
the smell
out of the office, and the bright orange from inside the server was
quite interesting :) Only took 1 second to start a small
-Original Message-
From: magnus [mailto:[EMAIL PROTECTED]
From my perspective, not sure I would want Exchange (Which
is difficult enough to manage) to be cluttered up with
potentially large voicemail files,
That's a concern, especially since bugs in current Asterisk versions
On Fri, 2005-06-10 at 08:12 -0500, Nathan Pralle wrote:
WHy not do it for free. Start the 'VoIP documentation project' on
sourceforge. It provides bandwidth, filesystem for images and all, php,
Erk! My vote is against Sourceforge, definately -- although it's free,
you get what you pay
-Original Message-
From: Iassen Hristov [mailto:[EMAIL PROTECTED]
Dumb, hacky idea...but just so crazy it might work:
Have Asterisk include a read receipt request when sending the voice mail
message. Write a script, triggered from a sendmail alias or .forward file,
that will parse the
check /etc/asterisk/modules.conf and make sure that you have load = meetme.so
best regards
On 6/10/05, craz sead [EMAIL PROTECTED] wrote:
hi all
i have * box with 4 ext using sj phone, i wanna try to
make a conference. i am using ztdummy and look fine
when i install it because there is no
Seems things have just got worse. Just got reports that 800 numbers
are not terminating. For example, can not dial:
800-888-9358
or
800-922-4684
Had to pull voipjet out of our routes until this gets fixed.
On 6/9/05, Moody [EMAIL PROTECTED] wrote:
We have been having serious quality problems
Hi friends,
Has anybody used a D-Link DVG-1402S VoIP gateway with * ?Please. Can send me any information to configurate
thisgateway?
Many thanks in advance.
Luis
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hello,
Sergio Serrano wrote:
I have to interconnect Asterisk with a Siemens HIPATH 3750.
I have configured siemens like Point to Point with and withouth CRC4 and
Asterisk with ccs,hdb3 ( with CRC4 and withouth CRC4), with pri_net and
pri_cpe and signalling=euroisdn
Anyone has
Hi
Everyone,
Is it possible to
have a SIP Phone work remotely if it's behind a Router performing NAT without
connecting the Router to a VPN? The Asterisk Box will be in the
DMZ.
Thanks
Dan
CYTEXONE
Dan Levine
[EMAIL PROTECTED]
CYTEXONE | Your Technology Specialists
nat=yes in sip.conf
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan
LevineSent: Friday, June 10, 2005 10:27 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Is it
possible to have a remote Phone work behindNat without a
VPN?
Hi
Everyone,
Is it
Hi All
i used cangoma card, connected with E1, using unicall. asterisk 1.1.x. when i dial to asterisk server. asterisk show error as belows:
-- Unicall/9 extension '9' in context 'from-pstn' from '71811242' does not exist. RejectingcallJun 10 16:47:59 WARNING[28159]: chan_unicall.c:2655
I have had a number of occasions where asterisk stopped
working. (1.0.7)
When this occured I tried to issue an asterisk -rx "stop now"
and nothing happened.
I then killall -9 asterisk, and it stops - but mpg123 is still hung.
I then killall -9 mpg123 and it stops.
I then restart asterisk and
most yes
On 6/10/05, David Sampson [EMAIL PROTECTED] wrote:
I have many old channel banks around that I would like to use to generate
analog extensions. Will most channel banks work with Asterisk?
Dave
___
Asterisk-Users
Hi,
I am brand new with asterisk
Just finished to install it
Have some problems to configure it
1st case:
IPphone LAN-- asterisk server
LANFW--internetdiax software
2nd case:
GSMtelephone lineasterisk
serverLAN--FWinternetdiax software
I would to have communication
It seems that configuring span=1,1,0,ccs,hdb3 and changing jitterbuffer=16
resolves or masks the issue. What I will do now is reduce again jitterbuffer
to default to see what happens.
To answer some of the questions I don't see hard disk activity when the
clicks appear, also the hard disk has
its a good idea to read all the comments in the configuration files in
/etc/asterisk/
in special asterisk.conf, extensions.conf, sip.conf, iax.conf and zapata.conf
best regards
On 6/10/05, Georges Henroteaux [EMAIL PROTECTED] wrote:
Hi,
I am brand new with asterisk
Just
Hey all,
Just getting started playing around with my Polycom 600. According to
the wiki, it looks like it's recommended to run BootRom 2.6.1 and SIP
1.4.1. Is that info still current, or is it safe to upgrade to 3.0.1
and 1.5.2?
Justin
--
___
Hello all,
I've installed AAH 1.1 on my VIA C3 powered mini PC. I've made the necessary
changes to the * makefile, so the compilation went well. The first thing I
did was configuring two extensions from AMP, namely 200 and 201. Then I
installed X-lite on two PC's and configured them with one of
Why would you even be routing 800 numbers out voipjet? They CHARGE you!
On 6/10/05, Pedro [EMAIL PROTECTED] wrote:
Seems things have just got worse. Just got reports that 800 numbers
are not terminating. For example, can not dial:
800-888-9358
or
800-922-4684
Had to pull voipjet out
I'm using the east coast server and am not experiencing any issues
either US based or international.
On 6/10/05, Pedro [EMAIL PROTECTED] wrote:
Seems things have just got worse. Just got reports that 800 numbers
are not terminating. For example, can not dial:
800-888-9358
or
Hi,
Its possible to make a pass-trhu conection with alaw or ulaw?
Thanks
--
Este mensaje ha sido analizado por C4I S.A. Mail Server
en busca de virus y otros contenidos peligrosos,
y se considera que está limpio.
MailScanner agradece a transtec Computers por su apoyo.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Alejandro G
Sent: Friday, June 10, 2005 8:12 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Clicks in audio with TE100P PRI
It seems that configuring span=1,1,0,ccs,hdb3 and
Hi Justin - Just getting started playing around with my Polycom 600. According to the wiki, it looks like it's recommended to run BootRom 2.6.1 and SIP 1.4.1. Is that info still current, or is it safe to upgrade to 3.0.1 and 1.5.2? I've been testing 1.5.2 for a few weeks now, and I'd have to say
Hi,
Both of those are fully uncompressed codecs and free to use.
Regards,
Sahil Gupta
VoiceValley
On Fri, 10 Jun 2005, Edgardo Bermejo wrote:
Hi,
Its possible to make a pass-trhu conection with alaw or ulaw?
Thanks
--
Este mensaje ha sido analizado por C4I S.A. Mail Server
en busca de
We are a VoIP provider and need to push out 100,000 - 200,000 minutes
per month (ie. need a carrier-level package - not a Vonage, etc.). To
date I have not found a wholesale SIP/IAX VoIP provider provide 800
termination for free. However, if you have one, please provide the
information and I
Darren Wiebe wrote:
The new version has an update database button. Install over your
old version and then press the update-database button in 'configure'.
This worked for me but... I think the default is not to use pins but
it is very easy to set yourself.
Unfortunately my case is not
Hello,
I use the ${BLINDTARNSFER} variable for transfers from SIP accounts, but
this variable seems to be unavailable for IAX channels. Is this supposed
to be this way, is there another variable???
Many thanks for your help,
Marc
___
Here is the setup:
Phone -SIP G729- AsteriskA -IAX G729- AsteriskB -SIP G729- Carrier
The call completes but AsteriskA prints on the screen a ton of those
Dropping Frame of G729 messages starting about 5 seconds into the call:
Jun 10 11:17:14 NOTICE[14277]: frame.c:135 __ast_smoother_feed:
Actually, they are compressed, but they are free to use :-)
Steve
Sahil Gupta wrote:
Hi,
Both of those are fully uncompressed codecs and free to use.
Regards,
Sahil Gupta
VoiceValley
On Fri, 10 Jun 2005, Edgardo Bermejo wrote:
Hi,
Its possible to make a pass-trhu conection with alaw
http://www.voip-info.org/tiki-index.php?page=Asterisk+call+forwarding
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of
[EMAIL PROTECTED]Sent: Friday, June 10, 2005 3:02
PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Cell
On 6/10/05, Clarke Kawakami [EMAIL PROTECTED] wrote:
Michael...
I don't believe that PHPAGI supports this currently. What you are looking
for is a combination of 2 functions: get_data() and swift().
That's what I was beginning to think but kept getting thrown off by
the escape digits
I tested all again. No matter if span=1,1,0 or span=1,0,0 if I configure
jitterbufer=4 I have glitches that I'm almost sure that are holes in
audio.
If I raise jitterbufer=16 the problem disappear (or becames impercetible).
Anyway I am interested in understand what is happening.
Your issue
Good things are happening.
Another aside from having done this before:
If configuration requires the user to do anything or the user
to load a piece of software it won't work.
Everything must be configured from an admin consol or it won't
work. You will go crazy trying to keep
Good Idea, but not practical as it breaks the second commandment of IT
user management.
1. Thou shall not require any brain cells on the part of the end-user.
2. Thou shall not require any settings to be set on the users equipment.
...
More rules to follow...
Race the tyrant Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
Alejandro G
Sent: Friday, June 10, 2005 9:57 AM
To: Asterisk
Subject: [Asterisk-Users] Clicks in audio with TE100P PRI
I tested all again. No matter if span=1,1,0 or span=1,0,0 if
I configure
When connecting from providers (I have tried 3 now) in the UK and having the calls routed to my asterisk server in the US, I am suffering a call disconnect problem.
The problem occurs whenever I record a call, either using record or sending the call to the voicemail application. This however
))) Please see comments inline.
From my perspective, not sure I would want Exchange (Which
is difficult enough to manage) to be cluttered up with
potentially large voicemail files,
That's a concern, especially since bugs in current Asterisk versions
require
you to use uncompressed
not sure but this may help you
http://voip-info.org/tiki-index.php?page=Asterisk%20call%20forwarding
Additionally, i can tell you that im using AGI to detect redirection
number. I Allow my users to set redirection from their Web based User
Panel, they can check their calls, and edit their
http://www.freeworldialup.com/advanced/peering_numbers
But I'm not sure if they would like you to terminate a lot of minutes over
it, just check it out.
Roman
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pedro
Sent: Freitag, 10. Juni 2005
Thanks a lot to all for the input.
I have now switched to the voipjet east coast back-up server and everything
seems to be back to normal now.
Thanks,
Roman
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Freitag, 10. Juni 2005 17:58
Justin Ellison wrote:
Hey all,
Just getting started playing around with my Polycom 600. According to
the wiki, it looks like it's recommended to run BootRom 2.6.1 and SIP
1.4.1. Is that info still current, or is it safe to upgrade to 3.0.1
and 1.5.2?
I am still running BootRom 2.6.1 with
I'm using bootrom 2.6.1 with 1.5.2 for the same reason. I would suggest the
upgrade to 1.5.2 for some non trivial enhancements such as multiple line/call
appearance. Also the menu system is significantly improved.
Louie
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Just did the same and it seems (cross fingers) to be fine now too.
However, I have to wonder. What happens to the load on that East Coast
box when we all switch over to it. Sure would be nice to hear from
VoipJet. Considering hwo many times I have recommended them, it would
make me feel better.
I have several toll free numbers that get forwarded to a single local
number assigned to a trunkgroup. I've asked the telco to not forward
those toll free numbers but to assign them as DIDs to the trunkgroup, so
that I can differentiate via DNID.
They said that they can't do that. That toll
I have several toll free numbers that get forwarded to a single local
number assigned to a trunkgroup. I've asked the telco to not forward
those toll free numbers but to assign them as DIDs to the trunkgroup, so
that I can differentiate via DNID.
They said that they can't do that. That
1 - 100 of 156 matches
Mail list logo
