Displaying 20 results from an estimated 57 matches for "chanells".
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chanels
2005 Mar 03
1
Is there a way to find free zap channels on remote servers ??
Hello:
I would like to know if there's a way to request free chanels from remote
asterisk servers ?
My idea is to make an agi returning a dial to inter-asterisk connected servers
when there's not enought chanels on local server, maybe like a ping to all of
them or maybe requesting to a central server where all the *s send and
request information about available chanels each 2 or 3
2004 Aug 06
3
Server based audio merge
Hi all,
<p>once again i came up with my conferencing stuff.
On a conference with more then two people it's a waste
of bandwidth, that every entity send it's data to every
entity. Since there is only one audio line, the audio must
be merged on the server.
Here are my questions:
- How many audio chanels can a server process (let's say a 3GHz machine)
in this way: decode all
2005 Jun 14
2
AVAYA & Asteris & H323 chanel
I'm trying to make H.323 trunk between AVAYA&Asterisk. But call from
AVAYA is terminated inmediatelly when apps DIAL on Asterisk is started.
Does any one use AVAYA and h.323 channel?
Thanks Bob.
2003 Aug 09
3
Need help with installation of H323 chanel driver
Hi
I am using inAccess channel driver.
Compiled, installed. This is what I get when I am trying to start *
---------------------------------------
[chan_oh323.so]WARNING[16384]: File loader.c, Line 226 (ast_load_resource):
libh323_linux_x86_r.so.1.12: cannot open shared object file: No such file or
directory
WARNING[16384]: File loader.c, Line 394 (load_modules): Loading module
chan_oh323.so
2004 Aug 06
0
Server based audio merge
I tend to disagree. It normal human conversation it wouldn't make much
sense to have 2 people talking over each other at the same time. Thus,
it most scenarios you would have only one talker anyway. Additionally,
encode->decode/mix/encode->decode isn't a very efficient CPU process for
a server, it's complicated to keep timing correct and it has a negative
impact on total
2004 Jul 15
0
Unable to create chanel of type SIP
I have a SIP phone that is registered. i can make calls out from the phone. I can't make calls to the phone.
What does the error message mean? How can I fix it? Thanks!
8 headers, 0 lines
Destroying call '6b9fb03c4677b9266e1263fb0c7ea304@127.0.0.1'
Jul 15 22:10:49 NOTICE[262159]: rtp.c:285 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible
== CDR updated
2014 Nov 04
1
Hangup Chanel when a peer unregisters
Hello group and thank you for the attention.
I'm using Asterisk 11.12 running on Ubuntu Server 12.04
We have an issue with channels remaining open after a SIP peer unregisters.
It seems that if the peer goes away before manually hanging up a call, the
channel remains open until a hangup request is sent from the CLI.
Is there any way to drop a channel when the peer using it disappears?
2005 Mar 17
3
Newbie can't dial out to pstn
Hi,
I have just put in a tdm400p with 4 fxo modules and am trying to dial
out from x-lite to dial my mobile phone just to test.
The output in the asterisk console is like this
Executing Goto("SIP/2002-239b", "mobile|61400039953|1") in new stack
-- Goto (mobile,61400039953,1)
-- Executing Goto("SIP/2002-239b", "localcall|61400039953|1") in
new
2005 Oct 07
3
TDM02B card difficulties
Hi all,
I just installed an TDM02B. My system is a dell pc with
linux 2.6.12-1.1456_FC4
asterisk-1.2.0-beta1
zaptel-1.2.0-beta1
libpri-1.2.0-beta1
in /etc/zaptel.conf I have (all others are default):
fxsks=3-4 <--- I saw light in the ports
channels=1-2 <--- change it to 3-4 has same result
but...
[root@nmsd0 asterisk]# /etc/rc.d/init.d/zaptel
2007 May 31
1
Mac OS X crash bug?
Hi all,
I want to check if this is a bug for which I should file a report.
I am using R2.5.0 on OS X 10.4.9. When I invoke the data editor and
when I change the values of individual cells, it seems to work as
intended. However, when I try to delete/add a row/column, R.app
crashes. I've attached the crash log.
Best,
-Nathan
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An embedded and
2005 Dec 30
7
streaming to dialup users gives low quality audio
Hello,
I've got two streams, one for broadband, one for dialup. Well, having
had occation to use a dialup connection recently i checked the dialup
stream. Although it was streaming what the broadband stream was, the audio
quality was audibly worse. It didn't buffer, but it didn't sound as clear as
the broadband stream. I used lame to encode the tracks to mp3 and used it's
2007 Nov 22
5
Odd bug in Siemens C460IP ?
Hello,
I think I have encountered an odd bug in Siemens C460 IP/dect handsets,
which is a bit annoying, and I'm not (yet) sure how to get round it without
lots of hacks.
Basically, on all external incoming calls, we set:
exten => s,n,SIPAddHeader(Alert-Info: Bellcore-dr2)
This causes handsets (i.e. Cisco 7960 / Grandstream / aastra) to set a
different ring cadence so to differentiate
2006 Feb 22
0
Outbound problem sip chanel
I setup my aah box with a sip trunk at irisxa.iristel.net
Incaming it is ok but when I try to dial 8 and the nr where I want to call I
get all line is busy.
In my log I have these:
Feb 22 14:33:04 DEBUG[3156] manager.c: Manager received command 'Command'
Feb 22 14:33:04 DEBUG[3156] manager.c: Manager received command 'Command'
Feb 22 14:33:19 VERBOSE[2721] logger.c: --
2004 Jun 22
3
Asterisk answering only one (dialed-) Number on a PTMP (German "Mehrgeräteanschluss")?
Hi,
please excuse my poor englisch.
Is it possible to connect a (privat Test-Asterisk) to my privat ISDN and allow him to only answer one dialed number?
We have 3 up to 10 Numbers on each (Euro-)ISDN (2 b-chanels), it cant't be done by the last Digits cause the numbers are completely different.
For Example:
I have 3 Numbers (641717, 928752....)
Is it possible to tell Asterisk (in
2002 Oct 09
0
Satellite TV hex files for Funcards, Goldcards
This is a multipart MIME message.
--= Multipart Boundary 1009021339
Content-Type: text/plain;
charset="ISO-8859-1"
Content-Transfer-Encoding: 8bit
Hello there
Did you know that you can program smart cards with files from the internet and open lots of pay per view chanells for your televisual pleasure.
Take a look at http://MagicFun.da.ru for the latest hex files.
Many thanks Jay.
--= Multipart Boundary 1009021339
Content-Type: text/html;
charset="ISO-8859-1"
Content-Transfer-Encoding: 8bit
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transition...
2009 Feb 23
4
Re: Playing wine games with hamachi?
hey McFlow, this theme actually important for me now )) Me and my friend have the same problem in Command & Conquer 3 Kane's Wrath if you still there (or somebody else) please, tell us how did you fix it??? :(
I have XP on my PC, my friend using Linux Gentoo. we make a chanel in hamachi, enter it. We pinging each other good but in the game I don't see him... and he sees me and my
2002 Oct 09
1
Satellite TV hex files for Funcards, Goldcards
Hello there
Did you know that you can program smart cards with files from the internet and open lots of pay per view chanells for your televisual pleasure.
Take a look at http://MagicFun.da.ru for the latest hex files.
Many thanks Jay.
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2004 May 02
1
* Newbie installation advice
Hello,
I'm about to install asterisk as the PBX at a location that my company has
just moved into and I would like to get some comments and advice on the
installation. I am new to * and don't want to make any big mistakes so I
would love to hear whatever anyone has to say.
Here is what I have so far
Server:
* 2.8Ghz P4 - 1G ram
* T400P Tormenta II (is this as good as the
2004 Nov 23
7
Unable to open master device '/dev/zap/ctl'
I installed TDM400P and X100P pci cards in a system running mandrake 10.1
official, kernel 2.6.8.1-12mdksmp. I can compile zaptel, libpri, asterisk
and modprobe (zaptel, wcfxs, wcfxo) without errors. Except that running
ztcfg and asterisk fails.
[root@asterisk asterisk]# ztcfg
Notice: Configuration file is /etc/zaptel.conf
line 3: Unable to open master device '/dev/zap/ctl'
2003 Jun 29
3
Help! Problems talking to upstream switch
Hi,
Please let me know if you have any ideas - I am taking wild guesses now....
Here is the situation:
I put in Asterisk for a local customer. I have Fractional T-1 with 12
Voice & 12 Data. I have a T100P hooked up to a TDM Card (they call it a
chanel bank although it only has 2 outputs) in a CAC unit. The unit also
has a router card that runs the data side. My extensions are all SIP