--- Klaverstyn, David C [EMAIL PROTECTED] wrote:
Can you provide some specific details as I would like to implement
something like this.
I wrote this application a while ago for FreePBX, maybe it helps:
http://samyantoun.50webs.com/asterisk/callback/callback.gif
Hi,
I was wondering if it is possible to connect a skype phone adapter, for
example:
http://zonetusa.com/DispProduct.asp?ProductID=191
http://www.actiontec.com/products/communications/ipw_usb/index.php
http://www.eradian.com/ERadianUS/staticpages/SkytoneRST301Details.htm
Hi,
I noticed that the sound directory is missing from asterisk-1.4.0-beta4.tar.gz.
This directory (7 M) witch existed in asterisk-1.4.0-beta3.tar.gz has GSM Core
Sounds and some MOH.
Does anyone know why it has been removed from the latest beta?
Regards.
--- Rich Adamson [EMAIL PROTECTED] wrote:
Have you tried:
cd /usr/src/zaptel
make update
make install
No, but I tried the SVN version and it compiled just fine.
Thanks
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--- Bill Maidment [EMAIL PROTECTED] wrote:
Hi
I've just tried to compile the zaptel-1.2.9 release and I get the
following error:
Same here, using CentOS 4.4 kernel 2.6.9-42.0.2.ELsmp, got these errors when
compiling zap:
make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command
--- Nigel Godfrey [EMAIL PROTECTED] wrote:
The work around is at:
http://www.sineapps.com/news.php?rssid=1496
Thanks, I'll give it a try.
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Hi,
I know that this is not the right place, but Im not aware of any alternative.
I have some VOIP equipments I would like to trade-up
Digium IAXyS101i
Used for 2 days
VoipSupply page: http://www.voipsupply.com/product_info.php?products_id=772
VoipSupply Price: $90
Linksys WBP54G 802.11G WIFI
--- Erick Perez [EMAIL PROTECTED] wrote:
A customer is asking for a manual. He's not talking about a How-To.
He's talking about a PDF/DOC that shows what files do what and what
parameters can be used and the syntaxis.
I think Asterisk Business Edition has one that comes with the box.
For
Hi,
I'm going to send a Sipura SPA-3000 to one of my friends in Egypt.
Does anybody has experienced any difficulties configuring the SPA-3000 to meet
the Egyptian PSTN network norms.
Appreciate your help.
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--- Graziano Poretti [EMAIL PROTECTED] wrote:
any idea where i can find the sip client to embed in my website ? (c# - java
or whatever)
SIP:
http://www.vaxvoip.com/WebDemo/Softphone.HTM
http://www.microappliances.com/site/html/index.php
http://www.etntalk.com/callto/loginany/
--- Tomá¹ Komárek [EMAIL PROTECTED] wrote:
Hello,
I am trying to setup a2billing system for asterisk. I have installed
it
corectly, but I have not found any users manual. I do not understand
the
whole structure. How do the parts like calling cards and sip friends
cooperate together?
Hi,
For some reason I need to do a reload every 15 minutes, the result is
a very large log file.
Does anyone know a way to prevent the logging of a reload verbose.
This is what I have so far:
echo /var/log/asterisk/tmp
mv /var/log/asterisk/full /var/log/asterisk/full_tmp
mv
Mark,
1. Make sure that SIPGetHeader application is registered
CLI show application SIPGetHeader
if it is registered you'll get
-= Info about application 'SIPGetHeader' =-
[Synopsis]
Get a SIP header from an incoming call
[Description]
SIPGetHeader(var=headername):
Sets a channel variable to
Amir,
1. What hardware do I need for the server to accept incoming and
outgoing analog calls.
http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TDM400P
With at least one FXO module
2. What books, guides or companies or individuals can help me setup.
a-
--- Jonathan Lin [EMAIL PROTECTED] wrote:
you get ping time in the status page if your extension.conf has
qualify=yes
Setup
# Device Location options
200 Sipura local
210 Sipura remote nat=yes qualify=yes
310 eyebeam remote nat=yes qualify=yes
sip show peers
Name/user Host
--- Goran Skular [EMAIL PROTECTED] wrote:
They do not have NAT option.. and they do not have qualify...
Ext 310 HAS nat=yes AND qualify=yes
# Device Location options
310 eyebeam remote nat=yes qualify=yes
sip show peers:
Name/user Host Dyn Nat Status
310/310 71.180.126.60 D
Hi,
Is there anyway to pass a variable from one context to another (NOT
macro and NOT global)
Regards
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--- Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote:
That is the default. Once you set a variable it should exist for the
life of the channel. Now, if you are wanting to access that variable
when one channel spawns another channel (like chan_local does), then
prefix the name of the
Hi,
I have 3 SIP extensions, setup as follows:
# Device Location options
200 Sipura local
210 Sipura remote nat=yes qualify=yes
310 eyebeam remote nat=yes qualify=yes
This is the result of sip show peers:
Name/user Host Dyn Nat Status
200/200 192.168.1.150 D Unmonitored
--- Sergey Okhapkin [EMAIL PROTECTED] wrote:
Are the devices at 200 and 310 set up to register with your asterisk?
Yes, they are registered and I can call them
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--- Sergey Okhapkin [EMAIL PROTECTED] wrote:
Hmm.. What is the output of sip show users and sip show peers?
sip show users
Username Def.Context ACL NAT
200 from-internalNo No
210 from-internalNo Always
310 from-internalNo Always
sip show peers
Name/user
--- John Millican [EMAIL PROTECTED] wrote:
Hello all,
Okay when you are done laughing at the simplicity of
this question could
someone show me please what I have wrong in the
following statement?
GoToIf($[${numdial} != [1-9] ]?15:3);
What this is supposed to do is if numdial is not a
--- Nate Kapi [EMAIL PROTECTED] wrote:
I've been having a lot of problems with Broadvoice
lately. Anyone else
been without service for extended periods of time
this week?
Service is down right now
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--- Blake Krone [EMAIL PROTECTED] wrote:
What is the best solution? I dont want to have
modify firewall's at all or
do port fowarding. Ideally I would like a solution
that with either a
softphone or wireless hardphone one could connect
via friends, family, or
hotspots without reconfiguring
Hi,
We have a Data/Voice service supplied through an
integrated T1.
Does anyone know if Digium T1 card will support the
splitting of the Voice and Data?
Regards.
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--- Kevin P. Fleming [EMAIL PROTECTED] wrote:
It _IS_ true, the code that does that work was even
recently improved.
Keven,
True, my mistake. I had 2 asterisk boxes, one running
ver 1.0.7 and the other running 1.2.0-beta1.
The log I posted before was from the 1.0.7, when I
looked at the log of
Hi,
Is there anyway to eliminate the color coding (for
example [1;36;40m) to be stored in asterisk log file?
Regards.
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--- Kevin P. Fleming [EMAIL PROTECTED] wrote:
Asterisk already strips the color codes before
putting the output into
the log files.
Kevin,
This is NOT true, bellow is some of my asterisk log
file:
Oct 8 16:41:49 VERBOSE[4016]: -- Executing
Anders,
There are 2 ways to acomplish this:
1. Keep your Asterisk box behinde your router. In this
case you need to do this:
- Port Forward to Asterisk Box
UDP 5060
UDP 1-2
- sip.conf (Change as needed)
externip=xxx.xxx.xxx.xxx
localnet=192.168.1.0/255.255.255.0
-
Can externalip be a dns-address?
Yes, as long as you can ping any address in the
Asterisk box (ping yahoo.com)
How do I configure the Incoming settings in the
siptrunk? I can call out
using the trunk but get busy tone when I try to dial
in. Use AAH
I don't know your provider, but generaly
Rene,
Try this registration string
1234567890:[EMAIL PROTECTED]/200
Where:
1234567890 = Your BroadVoice number
mypassword = Your BroadVoice password
200 = Your internal extension that will recieve the
incoming calls
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Mark,
One thing I know for sure (It was the reason why I
upgrade it to HEAD) is SipGetHeader application. I use
it to get the BroadVoice Distinctive Ring Information.
It is not available in 1.0.9
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--- Mark Phillips [EMAIL PROTECTED] wrote:
Can I get the 1.0.8 code.
Yes, but the SipGetHeader application is not available
in ANY of the production releases.
You can try to patch the 1.0.9 with
http://bugs.digium.com/bug_view_page.php?bug_id=0002838
And / Or
Andreas,
You may like to take a look at
http://mundy.org/blog/index.php
The part of Call Out feature is near half page, the
paragraph heading is Phone Home (with ET Image !!!)
I tried it and it work great.
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Hi,
This page
http://linuxcommand.org/man_pages/logrotate8.html
has a sample config file with a file size option
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Tommy,
If you meant VoicePulse, here is how to set it up
http://asteriskathome.sourceforge.net/handbook/index.html#Section_3.3.2
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Hi,
Did anyone successfully installed and setup Astaro
Security Linux V6 SIP Proxy with Asterisk behind the
Astaro and clients bedinde another NAT?
Regards
Start your day with Yahoo! - make it your home page
Hi,
I patched Asterisk 1.0.9 with patch 0002838
(SIPGetHeader Application), when running make, it
reported this error:
chan_sip.c: In function `sip_getheader':
chan_sip.c:9119: structure has no member named `cid'
Any ideas?
Regards
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Hi,
Does anyone know if Patches 0002838 and 0002924 are
included in Asterisk 1.0.7 ?
Regards.
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Somehow, Asterisk log files are consuming all the
space that I have in
my hard disk... They've already eaten 14GB and are
still hungry!! What
shall I do? I'm not even logging anything in
verbose mode!!
Leo,
This can be done through editing of
/etc/logrotate.d/asterisk. This file
Here is my setup:
Asterisk box on a Dynamic Public IP with 2 NIC, the
same box has a firewall (Shorewall) and a DHCP server.
Sipura 2000 connects remotely with NAT Keep Alive
Enable, qualify=no on Asterisk
No problems so-far, Iâm planning to send the Sipura
to Egypt so they can connect to my
Hi,
Iâve an Asterisk box with G729 Codec, a Broadvoice
trunk, IAXy connected locally and a Sipura 2000 (G729
as a preferred codec) Connected remotely. When Iâm
dialing out from the Sipura using the Broadvoice
trunk, everything is OK but if I dial the IAXy
extension, It keep ringing and I
Hi,
I've an Asterisk box acting as firewall with
Shorewall, yet I can't get a SIP client (Sipura 2000)
to connect remotely (behind a firewall). My Shorewall
Config as follows:
interfaces
#ZONE INTERFACE BROADCAST OPTIONS
net eth0 detect
dhcp,routefilter,norfc1918,tcpflags
loc eth1
I am looking for a IAX2 analog telephone adapter,
just want to ask your
views on which ones are bad, good and the best.
I have a Digium IAXy S101I, it works fine. The only
problem is that it has to be setup with the Asterisk
Server's IP Address, it wouldn't accept a domain name.
Hi,
I'm planning to get my Asterisk box out of the LAN,
get rid of my router and make the box acts as a
Router, Firewall, DHCP Server (with Shorewall).
I'll do that to be able to use some SIP clients
remotely.
Does anyone doing the same with the Asterisk box, is
it a good idea, is there any
It really depends on what kind of load that cpu is
going to have. There's no
technical problems with doing the above. Except I
don't see the point with
having a dhcp server, unless you are an ISP.
Steve,
Thank you for the valuable advice, I'll do exactly
what you are suggesting, No DHCP
Go for it!
I will Rod, wish me luck.
Thanks
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Asterisk-Users
Sorry I'm late. How about a shameless plug for my
distro - AstLinux.
Kris,
I was taking a look at your site yesterday, great
work. One day I'll get a Soekris and try it out. I'm
downloading the PC distro now to give it a try.
Appreciate your work
Samy
ps. Why you are not listed in
Can any one tell me what the mysql password, no it's
not password..
Try passw0rd with a disgit ZERO not o
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Hi,
I'm looking for a tutorial or installation guide for
SER to be used with asterisk to solve the remote SIP
agent problem. All the documents available are for
large scale installation.
Any help is highly appreciated.
Regards.
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Hi,
I'm looking for a tutorial or installation guide
for
SER to be used with asterisk to solve the remote
SIP
agent problem.
Which problem?
--
Cheers,
Matt Riddell
If the client is behind NAT and Asterisk Server is
behind another NAT
Hi,
Is it possible to have the server entry on the
Provisioning file a domain (e.g. sip.mydomain.com)
instead of an ip address?
Thanks in advance
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--- Luis Diaz [EMAIL PROTECTED] wrote:
Hi all, im Luis from argentina
i started to setup asterisk in my network but i just
cant...
so i have a few questions, if some one can help me
with
examples and or explanations would be great!
My Setup: Asterisk 1.0.7 (on 10.0.0.254) Gentoo
Hi
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