Displaying 20 results from an estimated 7000 matches similar to: "Media Proxy Mode in Asterik: SIP and"
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 Aug 02
1
H.323
Hi List;
Did any one tried the H.323 module? How much it is
stable and work fine?
Regards,
------------
ITS
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 00965 9849460
____________________________________________________________________________________Ready for the edge of your seat?
Check out tonight's top picks on Yahoo! TV.
http://tv.yahoo.com/
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 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 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 Jul 13
1
Can Asterisk hear on two IP addresses? And can I do routing for calls from private to public or public to private IP addresses
Hi List;
Can asterisk hear (receive) calls on two IP addresses?
How?
If yes, then:
If I have a VPN router, and my Asterisk server
connected to two network cards, one has a private IP
address (192.168.0.2) connected to the VPN router
(192.168.0.1) and another network card has a private
IP address (193.111.196.249) connected directly to the
outside default gateway (193.111.196.240), where the
VPN
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 03
5
Determining the used codec for the IP Trunk (SIP Trunk)
Hi List;
Where I determine the codec to be used for the SIP
Trunk (between Asterik and another SIP softswitch)?
Regards
Bilal
____________________________________________________________________________________
Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out.
http://answers.yahoo.com/dir/?link=list&sid=396545433
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 Jun 21
2
mediant 2000 with asterik configuration
Dear all
anyone have idea about connect asterisk with mediant 2000 audiocode configuration ... anybody have configuration about it
---------------------------------
Get your own web address.
Have a HUGE year through Yahoo! Small Business.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2013 Jul 24
1
Mysql Support int Asterik-11
Hi,
I was having question about mysql driver support ( not odbc).
Do we still need the asterisk-add-on to be installed for mysql support.? If yes, Which version should be used and from where I should get it?
Thanks in adavance.
----
Thanks & Regards,
PrashantAbhang
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2007 May 10
1
call transfer to asterik.. asterisk as an end point
Hello All.
I am having some trouble with call transfers when asterisk is the 2nd party called and I hope to benefit from your experience.
I want to use asterisk for call park/pickup and have configured openser
to relay calls made to ruri 700-720 to asterisk running on
localhost:5069
Call flow:
phone A calls phone B (both phones are polycom)
Phone B answers
then phone b
2007 Aug 26
1
Is it possible to register without sending the password?
Hi List;
I noticed that if I disabled secret in the context by
placing ( ; ) before it, then at the asterisk the log
will be:
-- Registered SIP 'bilal_sip' at 0.0.0.0 port 5060
expired
The IP address of the endpoint was not captured!!!
Why?
If secret enabled, then some endpoints can not
register (maybe due to compatibility in reading the
negotiation packets), so what is the solution?
2007 Aug 23
0
ASTCC and IVR
Hi list;
ASTCC supports IVR or there is a separate module for
IVR?
Can someone advise me a link to start download and
ready about ASTCC to do the configuration?
Regards,
---------
ITS
IP Telephony and Contact Center Engineer
Bilal Ghayad
Mobile: 00865 9849460
____________________________________________________________________________________
Boardwalk for $500? In 2007? Ha! Play
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
2001 Feb 08
2
Installling more than one wine version (fwd)
On Thu, 8 Feb 2001, Rick Moulton wrote:
> It seems that Corel Office 2000 for linux will not run on any version of
> wine later that the 1022999 ver.. Seems Any newer version causes an
> "ERROR no 6 and stuff about not being able to find "wordperfect" or
> quattropro executuables, and also saying it is a wine error.
> My question, is there a way to install
2017 Jun 29
2
The undef story
On Wed, Jun 28, 2017 at 11:53 PM, Peter Lawrence via llvm-dev <
llvm-dev at lists.llvm.org> wrote:
> Philip,
> email responses are varied, some say what you do, but
> others say give the guys a chance and listen to what he has to say.
>
> I say that I have a mild personality disorder such that I can’t say
> things in politically correct style, and that this is a
2011 Sep 29
2
[asterik-users] Installing PRI card
Hi,
We have got a new PRI card at one of our Office locations and now I need to
install the the device on a remote server. Is there any way to know if the
device is loaded already.
When I give " cat /proc/zaptel/* " it returns the following.
# cat /proc/zaptel/*
Span 1: WCT1/0 "Wildcard TE122 Card 0" (MASTER) B8ZS/ESF RED
IRQ misses: 2
1 WCT1/0/1
2007 Oct 19
1
SIP to H323 translator
Hi All;
If I installed H.323 on asterisk, and the caller phone
was SIP endpoint while I need to route the call for a
destination via an H.323 trunk, so Asterisk will do
that SIP to H.323 translation automatically or I have
to do also a configuration to SIP to H.323
translation?
Regards
Bilal
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the