search for: chanell

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

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

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.

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

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

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

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?

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

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

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 -------------- next part -------------- An embedded and

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

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

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: --

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

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 Transitio...

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

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. -------------- next part -------------- HTML attachment scrubbed and removed

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

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'

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