Displaying 20 results from an estimated 4000 matches similar to: "H.323"
2007 Jul 23
2
Upgrade and keep the configuration
Hi List;
How to upgrade the Asterisk, Zaptel and LibPri and
keep the configuration the same? I do not need to
remove current asterisk, zaptel and libpri and
download new one and write new configuration.
Regards,
--------------
ITS
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 00965 9849460
2007 Jul 13
3
Macro: s-NOANSWER, _s-.
Hi List;
I have this example for Macro and I am not able to
understand some line, if any one can help me plz :)-
[macro-voicemail]
exten => s,1,Dial(${ARG1},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten => s-NOANSWER,2,Goto(incoming,s,1)
exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten => s-BUSY,2,Goto(incoming,s,1)
exten
2007 Jul 01
4
Not able to find the file zaptel.conf after compiling asterisk and zaptel
Hi List;
I compiled Zaptel 1.4 and Asterisk 1.4 after
downloading them using svn, but when I checked the
file zaptel.conf under etc/asterisk, I did not find
this file. Any help?
By the way: How can I know the asterisk and zaptel
version extactly that I compiled them? In other words,
asterisk 1.4.... and zaptel 1.4.... ?
Regards
-------------
ITS
IP Telephony and Contact Center Engineer
Eng.
2007 Aug 02
4
Receiving SIP calls without registeration and dynamic IP address
Hi List;
How can I configure asterisk to receive a call from
SIP end point without being registered at asterisk and
its IP address is dynamic, and authentication to be
based on the username and password or any other
string?
I know that if I place the host with static IP then no
need to register, but what if the voip gateway was
having dynamic IP and I do not need to register on
asterisk, but I
2007 Jul 01
1
How can we block the calls for specific code
Hi List;
What is the command and where I can write it to block
specific code from calls (then no one will be able to
place call for any number start by that code)?
---------------
Regards
ITS
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: + (965) 9849460
Yahoo ID: bilmar_gh at yahoo.com
MSN ID: bghayad at hotmail.com
2007 May 01
10
Digital Phones
Hi List;
Asterisk does not have any kind of cards that can work
with it to be used with Digital Phones (digital phones
differ than analoge phone and differ than IP Phones).
Anyone can advise about this as I did not find this on
Diguim
Regards
Bilal Ghayad
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2007 Aug 27
4
Prepaid Billing: A2Billing, AstBill, ASTCC
Hi List;
I need to use an prepaid billing system with Asterisk,
and I do not know which one is more stable and
integrated with Asterisk?
A2Billing or AstBill or ASTCC?
Also, from where I can download it and ready about its
configuration?
Regards
ITS
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 009659849460
2007 Aug 23
3
Asterisk Prompt
Hi List;
I read the following sentence:
"The CLI prompt is set with the ASTERISK_PROMPT UNIX
environment variable"
In the following link:
http://www.voip-info.org/wiki/index.php
page=Asterisk+CLI+prompt
The question is: what is the ASTERISK_PROMPT UNIX
environment variable and where I can access it to
change it? Also where I can find information about it?
Regards
Bilal Ghayad
2007 Sep 09
3
canreinvite
Hi List;
If I need traffic to be directly between the
endpoints, then I have to set the canreinvite = yes?
If I did not configure the canrenvite at all, then by
default it will pass the traffic via Asterisk and not
directly between the endpoints?
What if one endpoint was SIP and configured with
canreinvite=yes while other endpoint was IAX2 and
configured with canreinvite=yes, then they can send
2007 Mar 30
1
Which IP Phones have buttons can be assigned to functions with Asterisk
Hi List;
Can someone advise me which IP Phone model that has
buttons that can be assigned to do specific
functionalities (call pickup, call formward, call
appearance) and a transfer button and hold button?
Which is the best of the following (that has buttons
can be assigned to specific functions):
Cisco 7970 or 7960
Polycom 501
Grandsream IP Phone Budge Tone 1001 or 1002
Linksys SPA 942 or 922
2007 Jul 13
1
Media Proxy Mode in Asterik: SIP and
Dear Alex;
Thanks for your kindly reply.
Please explain for me what do u mean exactly in "a la"
in the following sentence u wrote it below?
" in SIP, this can be done via
"re-INVITEs" a la the canreinvite= option for SIP
peers in sip.conf"
Another thing, do u mean that it is easier (better) if
we need H.323 endpoint to talk with SIP endpoint then
we use full
2007 Jul 13
1
Media Proxy Mode in Asterik: SIP and H.323
Hi List;
All we know that in voice, there are a type of
communications between endpoints, for example: in some
communications we do a proxy for media and signaling
while other communications we do a proxy for only
signaling.
Where I can determine these things in Asterisk if I am
using SIP and if I am using H.323?
Regards
--------------
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
2007 Sep 04
11
stop log/debug messages into /var/log/messages
Hello good ppl,
Any way of stopping asterisk writing into syslogs or any other file, if
I set verbose 6 on the console?
All I want is the verbose output only on the console, nowhere else.
My logger.conf says :
console=> notice,error
;messages => notice,warning,error
Thanks in advance.
- Benjamin Jacob.
EMAIL DISCLAIMER : This email and any files transmitted with it are confidential
2004 Feb 27
6
Video Conference
Is Asterisk capable of handling video conference? I am wondering if there is anybody in the list who tried it with NetMeeting(s). If it is possible, is the * required to register in the GK for this purpose? or making it as h323gw only is enough.
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2008 Oct 27
11
Fring: Open VPN client to be installed on the mobile, which mobile?
Hi All;
I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile:
Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server?
Regards
Bilal
2007 Jul 24
10
What is the best softphone work with Asterisk
Hi List;
I need to configure a softphone to be client and use
it with Asterisk, which is the recommended one? Is it
iax2?
Regards
Bilal
____________________________________________________________________________________
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2009 Jul 05
1
SIP IP-Trunk to be authenticated based on username and password, not IP address
Hi List;
How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed
2009 Jun 18
3
asterisk-gui: read/write in the conf files or db
Hi Danny;
Really I did not understand how I can determine if the IO will be DB or conf files? Is it from the Asterisk manager?
Regards
Bilal
-------------------------
It depends on how you are configured. The gui interfaces using Asterisk
Manager, so you get the Same IO from the gui that you would get from a
native manager session.
-----Original Message-----
From: asterisk-users-bounces at
2008 Dec 21
6
Asterisk and Dabatase
Hi All;
Anyone knows if there is an Asterisk version that setup can be stored in Database instead of the configuration files (.conf)?
Any advise?
Regards
Bilal
2007 Jul 08
2
asterisk is not sip proxy
Hello Asteriskers,
I'm confused about why Asterisk is not a SIP proxy and why exactly
this can affect the performance of a large Asterisk system.
I know that Asterisk acts as a useragent endpoint, but my doubt is why
exactly Asterisk could overload the call flow if the RTP voice stream
goes from the caller to the called party.
Does someone know how many calls or pencentaje that could handle