Displaying 20 results from an estimated 700 matches similar to: "Regarding asterisk"
2011 Jan 25
2
regarding quit, exit and stop now in asterisk
Hi all,
i am running asterisk by using command asterisk -r, asterisk -vc
............ when i want to come out of asterisk it not getting exit or quit
from the shell....
i have tried soo many options like...
stop now
stop gracefully
exit
stop
quit
its not working stillll can any one tell me what would be the problem with
this?
please help me ... :(
with regards,
viswavardhan
2011 Jan 26
5
Regarding error in Asterisk dail plan:
Hi all,
i am doing my master thesis on server perfromance evaluation i am
using asterisk as sip proxy server and sipp tool as traffic generator...
i have run basic testing of asterisk like as shown in website
http://sipp.sourceforge.net/wiki/index.php/Howto_test_an_Asterisk_server_using_SIPp
when i have copied sip.conf and extensions.conf from the site and run the
client and server i
2008 Feb 14
1
Ser, asterisk and ip2ipgw
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
<font size="-1">Hi,<br>
<br>
i use a ser, as proxy sip server(authentication), then a cisco router
as sip2h323 gw(authorization and accounting). i want to start asterisk
as sip statefull
2008 Jan 04
3
b2bua
Is there a way to disable the b2bua feature in asterisk.
I would like asterisk to work as a sip server and not be involved in the RTP path between phones.
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2009 Jul 21
2
best practices for running asterisk as SIP B2BUA
Hi,
What are the current best practices for running asterisk as SIP B2BUA?
Are there any sample configs online or the books that detail this
configuration for the newbies? I'm going to run it behind 1:1 NAT for
the clients in the public internet so I will use the externip, localnet,
and nat settings. Thanks,
Andrew
2004 May 17
4
Asterisk Proxy Type
Perhaps stupid question but, is Asterisk a statefull or stateless proxy?
Ignace
2004 Jan 09
1
* as sip b2bua?
Hi everyone,
any chance * could be used as a b2bua without forcing the media stream
through the same box? I would love to do some computing on incoming
calls, do things like setting another callerid and the forward the call
to another sip UA - all without any audio traversing the * box. Any
ideas?
Thanks,
Thilo
2001 Dec 27
1
Reg. Samba configuration
Hi All
I am facing a lot of problem for connecting my Linux server as a member
server in my Windows NT 4 based network. Kindley help me overcome it.
Details about my current setup
Linux Server
-------------
Server : RH Linux 7.1
Kernal : 2.4.2-1
Samba Ver.: 2.2.2
Network Details
----------------
Windows NT 4.0 Details
NetBios Names of the NT Servers
2012 Jun 04
3
HP DL360 G5 better than HP DL360 G7 ?
Any tips on solving the following performance conundrum:
Asterisk 1.8.12.2 running on HP DL360 G5 and G7s
tcpdump running to capture UDP 5060/SIP signaling to .pcap files
All calls are ultimately B2BUA client -> asterisk -> PSTN
Media stays on Asterisk at all times
AGI script has exit handler that connects and updates an external
database upon BYE from either side.
I know that if exit
2005 Jan 17
3
FW: Radius on *
Hello all,
It's my try to make some 'emulation' of vovida's b2bua using asterisk.
I was in rush while writing it, so I sure there is much code that can
be cleaned, great that not too much. :)
http://dslmax.boom.ru/asterisk_b2bua_v0.1.zip
cdrradius and agi script inside.
__
Mike Tkachuk
2009 Feb 01
1
asterisk-users Digest, Vol 54, Issue 109
Sorry, but why u r using the Radius with the CDR? Not enough to access the CDR in the /var/log/asterisk/cdr-csv/Master.csv?
Also, what kind of Radius u r using? Any suggested link?
Regards
Bilal
>
> Hello list.
>
> I'm having some problems with the CDR Radius in my
> Asterisk 1.4. I'm
> using two TC400B cards for transcoding. When I reach
> nearly 100
>
2013 Apr 12
3
Network based transcoding
Hello Everyone,
We are looking for solutions where the transcoding is abstracted away
from our * box (i.e., to the network layer) using some carrier grade
gateway, or router.
The reason for my post is to know about solutions people have used in
the past, and how it fits into their overall architecture. Our
transcoding needs consists mainly of u/alaw <-> g729, and gsm would
also be good....
2004 Oct 25
4
enquiry on shorewall functions
hi all,
shorewall claim that support stateful connection. But I read the
document, I can''t found any configuration on it like in iptables e.g.
-m -state NEW, ESTABLISHED
something like like.
Is shorewall by default is staeful connection for any connectione.g. web, http
2007 Aug 23
1
[Serusers] why combine ser with asterisk
Asterisk is an excellent PBX system, and makes a very good endpoint in
the SIP chain for all sorts of things -- IVR systems, voicemail
applications, automated messages, etc.
It has an extremely well-written CDR engine, so many people mesh it with
billing applications to produce accurate accounting information. It also
is fully aware of the media stream, which means it's capable of cutting
2005 Jan 18
2
MFCR2 - LIBUNICALL - Asterisk Problems
Dear Steve and *.* e1r2 developers and users,
now MFCR2 is successfully installed! many thanks for
your help.
I'm living in Argelia (north africa). I have configure
my MFCR2 according argentina R2 settigs :
the test call run perfectly (only warnings and I think
that is just debug).
but I have many problems and when I run
Asterisk-MFCR2.
generally in the begging no errors occur.
after
2010 Apr 29
2
No change in payload. (SDP)
re-posting the question.
-----------
use case:
when some one in my pbx calls 100.200, I have translations well defined, Media also (media via asterisk) --Works.
when some one calls bob, or for any names I am adding Domain and call is been sent to the other party -- Works, no media...
For the cases when it is talking to the external work,
I want Astersik not to do anything with the SDP/payload.
2004 Dec 13
7
Dialing out to 2 clients simultaneously
Hi
When I register a SIP or IAX client to asterisk and I dial to it from
another UA then there is no problem at all
But, when I register two or more clients to the SAME peer (with the same
user/pass) and I call to this peer.. Then only the UA which registered
the last will ring.. Others don't ring...
What can I do about this??
I would like to register for example 10 UA's to the same
2007 Nov 14
4
Hardware Requirements for qdisc htb/sfq
I am planning to replace our cisco 7200 core router with Linux. We
currently serve around 1500 (3/4 DSL - different router) customers with
probably half of them being concurrent at any given time.
We have a fiber network and customers currently aren''t managed as far as
how much bandwidth they can use at anytime. Therefore I have constructed
a working tc qdisc Linux router as a test. It
2008 Jan 04
1
asterisk as sip server
I am trying to setup asterisk as a registrar and sip server only.
Currently When I make calls all my rtp traffic is going through the asterisk server as a B2BUA.
Is it possible to turn off this feature and have all my calls RTP traffic going directly to the SIP UA?
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2009 Feb 19
2
Managing SIP hardphones call history
Hi,
I've been asked sometimes to tailor call history features embeded in SIP
hardphones.
For example, a cutomer wanted internal call to be taken out.
Another wanted calls to sorted according specific criteria.
1. Have you identified a phone offering the possibility to display as Call
History, an XML list produced on a distant web server ?
With this feature, you would simply have to tell the