Don't you just love outsourcing ?
- Original Message -
From: "Michelle Dupuis" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Tuesday, June 26, 2007 3:39 PM
Subject: Re: [asterisk-users] Best wifi IP phone for asterisk
(LINKSYSSUPPORT QUALITY
Have a look below:
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
It used to be easynews. Maybe your servers didn't take notice to the new DNS
changes for a bit ?
- Original Message -
From: "lenz" <[EMAIL PROTECTED]>
Zeeshan Zakaria wrote:
> I had the same situation and I had to replace my T1 card with the one
> with hardware echo canceller. All other solutions were failed. May be
> you need to do the same if you're on a PRI or using PSTN lines. If
> you're on a pure VoIP network, then its the phones.
>
> O
Darryl Dunkin wrote:
> Licenses are stored in /var/lib/asterisk/licenses, not in the module
> itself. Won't need any reactivation between versions either.
>
> There is no real need to delete the modules folder between minor
> versions like this, 'make install' will overwrite the modules and war
Licenses are stored in /var/lib/asterisk/licenses, not in the module
itself. Won't need any reactivation between versions either.
There is no real need to delete the modules folder between minor
versions like this, 'make install' will overwrite the modules and warn
you if there are any extra ones
I'm planning to upgrade my asterisk 1.4.4 to 1.4.6.
usually for asterisk upgrade i delete modules directory and include, then
compile the new version.
Since i have couple of G729 Licenses on this server installed, would i need to
call Digium to reactivate these Licenses?
Is there any better and f
I had the same situation and I had to replace my T1 card with the one with
hardware echo canceller. All other solutions were failed. May be you need to
do the same if you're on a PRI or using PSTN lines. If you're on a pure VoIP
network, then its the phones.
On 6/30/07, Jordan Novak <[EMAIL PROTE
Keshav K. escribió:
> I have installed asterisk 1.4.5 on Centos 4.4 with the kernel, 2.6.9-55.ELsmp
> In starting it showed some errors related to gtk2, while running make.
>
> After updating gtk It has been installed and working , I have test few basic
> thing , which are fine , but it has ec
I have three polycom 501 that are all hearing echo. The other party sounds fine
but you can hear yourself rather well. The volume does help if lowered but that
also makes the other party extremely quiet. Is there any way to control the
gain of the mic or stop the microphone from picking up so m
On Fri, Jun 29, 2007 at 07:40:30AM -0700, bilal ghayyad wrote:
> Hi Steve;
>
> I did what I told me below, and look like going fine
> but I do not know how can I know that zaptel
> compilation was implemented successfully specially I
> do not have a message in the end indicate this,
Generally su
Paul wrote:
> I'm going to top post in this situation.
>
> Kevin - Commands that operate on the channel variables won't help if we
> are using a call file. We will have a new channel.
>
Agreed, I misread and thought he was trying to generate a call file.
-Kevin
> This syntax works with asterisk
Keshav K. wrote:
> I have installed asterisk 1.4.5 on Centos 4.4 with the kernel,
> 2.6.9-55.ELsmp
> In starting it showed some errors related to gtk2, while running make.
You are probably talking about output from the configure script. You can
ignore
it, as it is not actually a problem. I'll
Hi all,
I am using mISDN with a Fritz!PCI v2.0. The card works without
problems until I use concurrently the two channels. In that moment
appears a little echo. I have reduced the tx gain but the echo not
disappear. I see in messages:
kernel: ECHOCAN: i:4000 TXBUF Overflow txbuflen:496 txcancelle
Jonathan Unai Marquez wrote:
> Thanks for your answer Jared, but I also tried that with no luck:
>
> Connected to Asterisk 1.4.5 currently running on moe (pid = 22879)
> -- Remote UNIX connection
> Verbosity is at least 6
> moe*CLI> keys show
> No such command 'keys show' (type 'help' for help
I have installed asterisk 1.4.5 on Centos 4.4 with the kernel, 2.6.9-55.ELsmp
In starting it showed some errors related to gtk2, while running make.
After updating gtk It has been installed and working , I have test few basic
thing , which are fine , but it has echo very much..
i'll test it furth
In zaptel compilation, in the newer version in both 1.2 and 1.4, make linux
will not work.
To install the zaptel-1.4 you can use these lines:--
cd zaptel1.4.XX
./configure
make menuselect
make
make install
Regards,
Keshav
Doug Lytle <[EMAIL PROTECTED]> wrote: bilal ghayyad wrote:
> Hi Li
Hi Everyone,
Im testing a ML100 G4 (Pentium D) Server from HP with a TDM400P from
Digium.
I just installed, with success, the following O.S. with Asterisk 1.4.5
1) Centos 4.4
2) Centos 4.5
3) Centos 5.0
Id like to receive a recommendation about whats S.
Figured it out.
I forgot that /etc/zaptel.conf got clobbered during the install. Once
I fixed that and ran:
/sbin/ztcfg -
It worked. Zaptel.conf looks like this:
loadzone = us
defaultzone=us
fxoks = 1-2
fxsks = 4
I have 1 FXO module & 2 FXS modules.
Thanks for putting me on the right t
Thanks for responding Russell!
> What output do you get if you run "module unload chan_zap.so"
== Unregistered application 'ZapSendKeypadFacility'
> and then "module load chan_zap.so" ?
== Registered application 'ZapSendKeypadFacility'
== Parsing '/etc/asterisk/zapata.conf': Found
[Jun 30
you should turn on sip debug on asterisk and median and see, if sip/180
ringing messagess are propagated through mediant to avaya,
avaya should react to sip/180 ringing with generating ringback to
calling phone...
sip/183 is progress message, in this case is audio path "open" to
playback progres
bilal ghayyad wrote:
> Hi List;
>
> I am facing a problem relaed to menuselect when I am
> trying to compile zaptel -1.4.2.1, the error as
> following:
>
> [EMAIL PROTECTED] zaptel-1.4.2.1]# make linux26
> m
You no longer need to do a make linux26. Just delete the zaptel
directory and extract a
Hi guys
I'm at a loss in getting ./configure to complete successfully with asterisk
1.4.6 on
Fedora 7 x86_64, as it complains about no termcap support, even though it is
installed
(see below).
Any ideas where to go next?
checking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... no
configure: error:
Hello
I've spent the past hour searching and reading and searching some
more. I've jet to come to a solution.
I want to have some of my extensions in extensions.ael, while some
should reside in Realtime.
Realtime works just fine, so does my extensions in AEL. But not on the
same time.
This is h
It's messages from the list that get delivered after a few days.
l.
In data Fri, 29 Jun 2007 23:31:44 +0200, Mojo with Horan & Company, LLC
<[EMAIL PROTECTED]> ha scritto:
> Is it taking a while for _your_ messages to post to the list, or do you
> mean messages from the mailing list software t
On 6/30/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
I just upgraded my 7970G to the SIP firmware. What I'd like to do is have
the 8 line buttons be able to make outbound calls using the same account
(for practical purposes, same caller-ID). Since the phone is going to have a
single public D
For this you have to make entry in sip.conf.
it will be better if you use host=dynamic in both the phones in sip.conf
and what is the IP you are putting in phones which are on your PC.
Also check that your both sip phones which are on PC, are sending requestr to
asterisk server or not.
Kesh.
Hi the list,
600v3 with last firmware works fine with Asterisk and SIP.
I use it every days with success, no issue.
I recommend it and think it's more reliable than WiFi for a great number of
handsets or industrial deployment with multicells.
Best Regards,
Francois BERGERET,
France
-Message
27 matches
Mail list logo
