search for: chanel

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 avai...

...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 speex chanels, merge them into one chanel and then encode the result again? This means that some chanels are merged to one chanel and this multiple. - Could aditional DSP-PCI-Boards help to do this? - Is it posibl...

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

...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 speex chanels, merge them into one chanel and then encode the result again? This means that some chanels are merged to one chanel and this multiple. - Could aditional DSP-PCI-Boards help to do this? - Is it posibl...

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

...ving zaptel module: [ OK ] Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules:Running ztcfg: Notice: Configuration file is /etc/zaptel.conf line 207: Cannot get number of tones chanel 1 line 207: Cannot init tones chanel 1 /etc/init.d/functions: line 408: 13058 Segmentation fault $* [FAILED] Wires are checked. Can anyone point me to next step? Thanks a lot, Min

...======================================================== =============================== When you have to stay eight years away from California, you live in a perpetual state of homesickness. - Ronald Reagan The most courageous act is still to think for yourself. Aloud. -Coco Chanel ======================================================================== ===============================

..., 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 standard preset while doing it. In my ices.conf file for the dialup stream i originally had a samplerate of 22050, two chanels, and a bitrate of 24. I changed the bitrate up to 56, which resulted in a noticeable audio increase in quality but the buffering was unacceptable. If anyone has settings that work i would be interested in hearing about them. Thanks. Dave.

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 Extensions.conf?) to Answer 641717 an ignore incomming calls on 928752? I need this solution to work with Asterisk without d...

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 Transiti...

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 chat well

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

...d 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 wildcard?) Chanel Bank: * Adit 600 3FXS, 1FXO * We will start with 6 PSTN lines Phones: * Aastra PowerTouch 480 (Management, Customer Service etc) * Aastra Meridian 8004 (break room, warehouse floor etc) * Reception? Any comments or suggestions would be appreciated as I have no idea what type of phone t...

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 phones save a few fax machines. The customer has 7 digit unverified account codes on the trunks for billing purposes. The Problem: As I watch the console, I s...