RE: [asterisk-users] Call Back

2007-03-13 Thread Samy Antoun

--- 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

http://samyantoun.50webs.com/asterisk/athome/callback.htm



 

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[asterisk-users] Connect a Skype adapter to TDM400P

2007-01-20 Thread Samy Antoun
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
http://www.dlink.com/products/?pid=466
http://www.usr.com/products/voip/voip-product.asp?sku=USR9620

to a TDM400P, so that the Line port of the adapter connects to the FXS port of
the TDM400P and the Phone port of the adapter connects to the FXO port of the
TDM400P

Appreciate any help

Regards.



 

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[asterisk-users] Asterisk 1.4.0 B4 Sounds Directory

2006-12-16 Thread Samy Antoun
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.




 

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Re: [asterisk-users] Zaptel-1.2.9 compile error

2006-09-10 Thread Samy Antoun
--- 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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Re: [asterisk-users] Zaptel-1.2.9 compile error

2006-09-09 Thread Samy Antoun
--- 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 not found
make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not found
make[3]: *** No rule to make target
`/usr/src/zaptel/wct4xxp/../oct612x/include/oct6100api/oct6100_api.h', needed
by `/usr/src/zaptel/wct4xxp/vpm450m.o'.  Stop.
make[2]: *** [/usr/src/zaptel/wct4xxp] Error 2
make[1]: *** [_module_/usr/src/zaptel] Error 2
make: *** [linux26] Error 2

Hope someone has a workaround for this problem


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Re: [asterisk-users] Zaptel-1.2.9 compile error

2006-09-09 Thread Samy Antoun
--- 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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[Asterisk-Users] VOIP equipment trade-up

2006-05-26 Thread Samy Antoun
Hi,

I know that this is not the right place, but I’m 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 Dongle
Never Used
VoipSupply page:  http://www.voipsupply.com/product_info.php?products_id=1094
VoipSupply Price: $45

D-Link DI-102 VOIP QOS Adaptor Packet Prioritizer
Never Used
VoipSupply page:  http://www.voipsupply.com/product_info.php?products_id=1168
VoipSupply Price: $80

I would like to trade the three of them with
Linksys SPA-942 SIP Phone Dual RJ45, PoE, Inc. AC P/S
VoipSupply page:  http://www.voipsupply.com/product_info.php?products_id=1334
VoipSupply Price: $190

If interested, kindly email me off the list
[EMAIL PROTECTED]

Thanks


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Re: [Asterisk-Users] Asterisk Manuals

2006-05-25 Thread Samy Antoun
--- 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 the recent *not ABE* stable release will be nice.
 
 Comments?

Eric,

Here is a very good set of documents:
http://www.asterisk.org/doxygen/



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[Asterisk-Users] Sipura SPA-3000 in Egypt

2006-03-05 Thread Samy Antoun
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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Re: [Asterisk-Users] click to talk

2006-02-04 Thread Samy Antoun
--- 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/
http://www.worksoutsoft.com/products/intellIPhoneSDK.aspx

IAX:
http://www.silicontechnix.com/webtelefone/start.html


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Re: [Asterisk-Users] a2blling billing system

2006-01-11 Thread Samy Antoun
--- 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?

Tomas,

The calling card application can work in 2 ways:
1. Customer calls a number (At you * server), the system will prompt
for a pin number, after authenticating, a balance is anounced, customer
is ready to make a call.
2. Customer will use a SIP or IAX phone (Software or Hardware) to
register, in this option you need to create a SIP or IAX Friend.

I've created a page on how to install it, but it's based on
[EMAIL PROTECTED]:

http://samyantoun.50webs.com/asterisk/athome/a2billing/



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[Asterisk-Users] Prevent Logging reload verbose

2005-12-14 Thread Samy Antoun
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 /var/log/asterisk/tmp /var/log/asterisk/full
asterisk -rx logger reload
asterisk -rx reload
mv /var/log/asterisk/full /var/log/asterisk/tmp
mv /var/log/asterisk/full_tmp /var/log/asterisk/full
asterisk -rx logger reload

I'm not quite satisfied with this solution, I would appreciate if
anyone has a better solution

Regards.


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Re: [Asterisk-Users] Detect alternate line in Broadvoice inbound context

2005-11-21 Thread Samy Antoun
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 the content of a SIP header
Skips to priority+101 if header does not exist
Otherwise returns 0
If not,
Your application(s) is (are) not registered

If the application is not registered, I can't recommend anything for
you, I had an Asterisk system with ver 1.0 (no SIPGetHeader) and I
tried to patch it with any of the following with no luck:
http://bugs.digium.com/bug_view_page.php?bug_id=0002838 
http://bugs.digium.com/view.php?id=2924

If you have it registered, here is a sample of my setup:
[bvdr]
exten = s,1,Answer 
exten = s,2,DigitTimeout(5) 
exten = s,3,ResponseTimeout(10) 
exten = s,4,SIPGetHeader(Var_Alert=Alert-Info) 
exten = s,5,GotoIf($[${Var_Alert} =
http://127.0.0.1/Bellcore-dr3]?ext-local,320,1) 
exten = s,6,Goto(ext-local,200,1)

This setup for ONE Distinctive Ring only (Bellcore-dr3), if you have
more than one, you can use sip debug to retrieve the header
information

The BEST reference for this subject is:
http://voxilla.com/PNphpBB2-viewtopic-t-3935-highlight-dring1.html

Hope this helps





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Re: [Asterisk-Users] help and guidance needed from gurus

2005-11-15 Thread Samy Antoun
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- http://sourceforge.net/projects/asteriskathome/ (heighly
recommended)
b- http://www.voip-info.org
c- This List

 3. I need scalable hardware.
If by hardware you mean VOIP hardware, take a look at
http://www.digium.com

 4. What do I need in terms of technology and hardware to route calls
internationally.
You need to specify the location of the three elements:
a- Asterisk Server
b- Clients
c- Trunks
for example:
Asterisk Server Located at Japan, has some BroadVoice trunks from the
US, Serving 10 Software Clients (xten) located at South Africa

In case of Analog Trunks, Asterisk server has to be in the same
location as the trunk

If you need further help, please email me off the list
 




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Re: [Asterisk-Users] sip show peers

2005-10-16 Thread Samy Antoun
--- 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  Dyn Nat Status
200/200   192.168.1.150  D  Unmonitored
210/210   84.36.36.140   D   N  OK (305 ms)
310/310   71.180.126.60  D   N  Unmonitored

As you can see, ext 310 has qualify=yes and and Unmonitored Status !!!




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RE: [Asterisk-Users] sip show peers

2005-10-16 Thread Samy Antoun
--- 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   N  Unmonitored





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[Asterisk-Users] Pass variable to context (NOT macro)

2005-10-16 Thread Samy Antoun
Hi,

Is there anyway to pass a variable from one context to another (NOT
macro and NOT global)

Regards





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Re: [Asterisk-Users] Pass variable to context (NOT macro)

2005-10-16 Thread Samy Antoun
--- 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 variable by _ or __ .  I'm sure it's
 documented 
 somewhere, but I have no idea where.  The underscore and double 
 underscore prefix feature is NOT available in 1.0.x, only in CVS-HEAD

Eric,
If I have two contexts:

[context1]
exten = s,1,Answer
exten = s,2,SetVar(MYVAR=1)
exten = s,3,Goto(context2,s,1)
[context2]
exten = s,1,NoOp(${MYVAR})

The NoOp in context2 will return 1?





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[Asterisk-Users] sip show peers

2005-10-15 Thread Samy Antoun
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
210/210   84.36.36.140   D   N  OK (305 ms)
310/310   71.180.126.60  D   N  Unmonitored

Does anyone know why ext 210 the only one has a ping
status OK (305 ms) and the others are Unmonitored

Regards




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Re: [Asterisk-Users] sip show peers

2005-10-15 Thread Samy Antoun
--- 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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Re: [Asterisk-Users] sip show peers

2005-10-15 Thread Samy Antoun
--- 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 Host  Dyn Nat Status
200/200   192.168.1.150  D  Unmonitored
210/210   84.36.36.140   D   N  OK (305 ms)
310/310   71.180.126.60  D   N  Unmonitored





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Re: [Asterisk-Users] match a set of numbers in GoToIf against a variable

2005-10-14 Thread Samy Antoun
--- 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
 single digit from 1 to 9 
 inclusive goto 15, if it is a singledigit from 1 to
 9 inclusive goto 3.  
 Should be pretty simple but not working for me,
 always goes to 15.  from what 
 I read on the wiki this should work, but obviously I
 must have read it wrong.
 
 I can get this to work:
 exten = 1,2, GoToIf($[${numdial} = 1 ]?15:3);
 exten = 1,3, GoToIf($[${numdial} = 2 ]?15:4);
 exten = 1,4 ,GoToIf($[${numdial} = 3 ]?15:5);
 ...
 exten = 1,15, do other stuff
 but I don't want to put this in 9 times.
 
 Thanks in advance,
 John M

John,

Will this meet your needs:

exten = _Z,1,Will match any single digit from 1 to 9
exten = i,1,will match enything else (invalid)




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Re: [Asterisk-Users] Broadvoice Outages?

2005-10-13 Thread Samy Antoun
--- 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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Re: [Asterisk-Users] SIP behind NAT to pub Asterisk, best solution?

2005-10-13 Thread Samy Antoun
--- 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 their devices.
  What are people using? STUN? SER?
  Thanks in advance!

This is my setup:
http://samyantoun.50webs.com/asterisk/firewall/firewall.htm

It works great, it's been up and running for the last
year with couple of local extensions, couple of remote
extensions within the same state and couple of remote
extensions overseas.





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[Asterisk-Users] Integrated T1

2005-10-12 Thread Samy Antoun
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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Re: [Asterisk-Users] Asterisk Log Color Coding

2005-10-11 Thread Samy Antoun
--- 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 the other box (1.2.0-beta1), it
was perfectly fine.

Thank you for pointing this out.





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[Asterisk-Users] Asterisk Log Color Coding

2005-10-08 Thread Samy Antoun
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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Re: [Asterisk-Users] Asterisk Log Color Coding

2005-10-08 Thread Samy Antoun
--- 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
SetCallerID(IAX2/[EMAIL PROTECTED]/1,
Samy Antoun 8184281334) in
new stack
Oct  8 16:41:49 VERBOSE[4016]: -- Executing
SetGroup(IAX2/[EMAIL PROTECTED]/1,
OUT_3) in new stack
Oct  8 16:41:49 VERBOSE[4016]: -- Executing
CheckGroup(IAX2/[EMAIL PROTECTED]/1,
2) in new stack
Oct  8 16:41:49 VERBOSE[4016]: -- Executing
SetVar(IAX2/[EMAIL PROTECTED]/1,
DIAL_NUMBER=19056661572) in new
stack
Oct  8 16:41:49 VERBOSE[4016]: -- Executing
SetVar(IAX2/[EMAIL PROTECTED]/1,
DIAL_TRUNK=3) in new stack
Oct  8 16:41:49 VERBOSE[4016]: -- Executing
AGI(IAX2/[EMAIL PROTECTED]/1,
fixlocalprefix) in new stack
Oct  8 16:41:49 VERBOSE[4016]: -- Launched AGI
Script /var/lib/asterisk/agi-bin/fixlocalprefix
Oct  8 16:41:50 VERBOSE[4016]: -- AGI Script
fixlocalprefix completed, returning 0
Oct  8 16:41:50 VERBOSE[4016]: -- Executing
SetVar(IAX2/[EMAIL PROTECTED]/1,
OUTNUM=19056661572) in new stack
Oct  8 16:41:50 VERBOSE[4016]: -- Executing
Cut(IAX2/[EMAIL PROTECTED]/1,
custom=OUT_3|:|1) in new stack




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Re: [Asterisk-Users] NAT

2005-09-29 Thread Samy Antoun
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

 - Extensions
   nat=yes
   qualify=yes

2. Configure your Asterisk box as a router. In this
case you need:
 - Asterisk box must have 2 NIC's
 - A network switch
 - Setup instructions can be found at:
http://samyantoun.50webs.com/asterisk/firewall/firewall.htm

Hope this helps.





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RE: [Asterisk-Users] NAT

2005-09-29 Thread Samy Antoun
 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 you need to
setup the trunk in AAH so it will be suffixed with
/YourNumber, you can find an example at:
http://samyantoun.50webs.com/asterisk/athome/setup/trunkbroadvoice.htm

Also, you need to have a DID Route (Inbound Routing in
ver. 2) with your number.




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Re: [Asterisk-Users] Broadvoice inbound issues

2005-09-29 Thread Samy Antoun
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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Re: [Asterisk-Users] Does the 1.0.9 release contain the Broadvoice patches?

2005-09-28 Thread Samy Antoun
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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Re: [Asterisk-Users] Does the 1.0.9 release contain the Broadvoice patches?

2005-09-28 Thread Samy Antoun
--- 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
http://bugs.digium.com/view.php?id=2924

I tried to do this before with no luck



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Re: [Asterisk-Users] Make asterisk call out

2005-09-11 Thread Samy Antoun
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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Re: [Asterisk-Users] Syslog file size

2005-09-11 Thread Samy Antoun
Hi,

This page
http://linuxcommand.org/man_pages/logrotate8.html

has a sample config file with a file size option






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Re: [Asterisk-Users] Asterisk and AMP installed now what?

2005-09-11 Thread Samy Antoun
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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[Asterisk-Users] Astaro SIP Proxy

2005-09-01 Thread Samy Antoun
Hi,

Did anyone successfully installed and setup Astaro
Security Linux V6 SIP Proxy with Asterisk behind the
Astaro and clients bedinde another NAT?

Regards





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[Asterisk-Users] SIPGetHeader

2005-09-01 Thread Samy Antoun
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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[Asterisk-Users] Patches 0002838 and 0002924

2005-08-11 Thread Samy Antoun
Hi,
Does anyone know if Patches 0002838 and 0002924 are
included in Asterisk 1.0.7 ?
Regards.

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Re: [Asterisk-Users] How to prevent log files from eating my hard drive?

2005-07-05 Thread Samy Antoun
 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 contains a script
of log rotation. This is a sample file:

/var/log/asterisk/*log {
   missingok
   rotate 5
   weekly
   create 0640 asterisk asterisk
   postrotate
   /usr/sbin/asterisk -rx 'logger reload' 
/dev/null 2 /dev/null
   endscript
}

As you can see, the rotate 5 means that it will keep
5 old files, you can reduce that number to 2 or 3 to
save some space.

Another thing you can do to save a LOT of space is to
change weekly to daily, this way you will keep
only ONE DAY of logging data in the file.

Third thing you can do is to add compress statement
to the script, this will compress the old files.

I think if you'll combine those 3 options, you going
to have a log files that is almost 10% of what you
have now.

Regards.

Samy



 
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Re: [Asterisk-Users] How do you handle NAT?

2005-06-29 Thread Samy Antoun
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, I’m planning to send the Sipura
to Egypt so they can connect to my Asterisk box,
that’s why I’ experimenting with G729 due to the
fact that the band width will be an issue.



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[Asterisk-Users] IAXy G729 Translation

2005-06-20 Thread Samy Antoun
Hi,
I’ve an Asterisk box with G729 Codec, a Broadvoice
trunk, IAXy connected locally and a Sipura 2000 (G729
as a preferred codec) Connected remotely. When I’m
dialing out from the Sipura using the Broadvoice
trunk, everything is OK but if I dial the IAXy
extension, It keep ringing and I can’t hear the
voicemail prompt (Though the CLI showing that it plays
the voice prompt). Odder, if I disconnect the IAXy and
try to call that extension from the Sipura, I can hear
the voicemail prompt.
Anyone has an explanation for this behavior?
Regards.


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[Asterisk-Users] Shorewall Configuration for Asterisk Box

2005-06-11 Thread Samy Antoun
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  detecttcpflags

zones
#ZONE DISPLAY COMMENTS
net   Net Internet
loc   Local   Local Networks

policy
#SOURCE DEST POLICY LOGLEVEL
loc net  ACCEPT
fw  net  ACCEPT
net all  DROP   info
all all  REJECT info

rules
#ACTION SOURCE DEST PROTO DESTPORT
ACCEPT  fw net  tcp   53
ACCEPT  fw net  udp   53
ACCEPT  locfw   tcp   22
ACCEPT  locfw   icmp  8
ACCEPT  netfw   icmp  8
ACCEPT  fw loc  icmp
ACCEPT  fw net  icmp
ACCEPT  netfw   udp   1:2
ACCEPT  netfw   udp   5060
ACCEPT  netfw   tcp   5060
ACCEPT  netloc  udp   5060
ACCEPT  netloc  tcp   5060
ACCEPT  netfw   udp   4569
ACCEPT  netfw   tcp   4569
ACCEPT  locfw
ACCEPT  fw loc
DNATnetloc:192.168.1.10 tcp http

masq
#INTERFACE SUBNET
eth0   eth1

Also, I'm trying to find any documentation for
shorewall logwatch command.

Any help is heighly appreciated.
Regards.

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Re: [Asterisk-Users] IAX2 analog telephone adapter

2005-06-01 Thread Samy Antoun
 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.

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[Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server

2005-06-01 Thread Samy Antoun
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 other solution for the
SIP emote Clients.

Regards.




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Re: [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server

2005-06-01 Thread Samy Antoun
 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

Regards.

Samy

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Re: [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server

2005-06-01 Thread Samy Antoun
 Go for it!

I will Rod, wish me luck.

Thanks



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Re: [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server

2005-06-01 Thread Samy Antoun
   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 distrowatch.com




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Re: [Asterisk-Users] asterisk@home

2005-05-30 Thread Samy Antoun
 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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[Asterisk-Users] SER Help

2005-05-25 Thread Samy Antoun
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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Re: [Asterisk-Users] SER Help

2005-05-25 Thread Samy Antoun
  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




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[Asterisk-Users] IAXy Provisioning

2005-05-23 Thread Samy Antoun
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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Re: [Asterisk-Users] Basic newbie questions

2005-05-23 Thread Samy Antoun

--- 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 Luis,

 1) Can i have LOCAL users on my lan?
YES, it's better to identify the users as Extensions,
you can have as many Extensions as you like.

 2) what should i use AIX or SIP
If all Extensions (Users) will connect from the same
network (10.0.0.10. 10.0.0.25 ... etc.), you can use
SIP. If the Users will connect remotely (Anywhare at
the Internet) IAX is the answer.

 3) can my local users have a phone number?
They HAVE to. Any user MUST have an extension (Phone
number)

 (lets say im 1112 and i can call to my neighbour on
 phone number 1113)
Yes, in this case you will dial 1113 to talk to your
nabour.

 4)what soft phone is the most compatible with
 asterisk
A lot of phone are Asterisk compatible, most commonly
used SIP phone is xten (www.xten.com), IAX phone is
Firefly (www.virbiage.com)


IMPORTANT NOTE:
I would Hieghly recommend [EMAIL PROTECTED]
(http://sourceforge.net/projects/asteriskathome), it's
an ISO image of Asterisk with a very nice, easy to use
web interface.

Hope this helps




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