similar to: Interconnection of LDK to an Asterisk server

bounjour je dispose de differents commutateurs de LG (IP LDK) sur differents sites. je voudrais savoir comment je pourrais interconnecter ces differents IP LDK a un serveur Asterisk via IP ( ceci sous entend que chacun de ces commutateurs dispose d?j? d'une carte VOIBE). Mecri d'avance pour l'aide _________________________________________________________________ MSN Messenger :

hi, How to make to authorize a user samba to be connected from one or two particular machine of the network only and not of the others? thanks _________________________________________________________________ MSN Hotmail : antivirus et antispam int?gr?s http://www.msn.fr/newhotmail/Default.asp?Ath=f

hello I try to manage a printer (Apple Lawerwriter 16/600 ) with cups but without succes !!! with cups web interface : I "add a printer": device = LPD/LPR Host or Printer URI= lpd://mymachine/lp Type= Apple Model = Apple LaserWriter16/600 Foomatic/Postscript (Recommended)(en) I try to print a testpage but without succes - 'mymachine' is either in my dns and my /etc/hosts

Hello, I wish to limit acces of a user samba to his own machine and a pool of common machines only. If it is connected starting from the machine of the neighbor, it reaches all its local files!!! Does somebody have an idea? Thank you in advance for your assistance _________________________________________________________________ MSN Hotmail : antivirus et antispam int?gr?s

hi all, this message is destinate to users who rub libunicall on asterisk succefuly ... my first question is: what is the version of linux you use to make asterisk run with the new libunicall? and what is the version of asterisk? my problem is as follows: with version FC3 I have success to run libunicall without problem, the testcall run perfectly well. askterisk run without problem on the

I am using Aserisk as a SIP server to interconnect differents PBX in differents sites. I am now looking for a tool that can test the performance of this solution: I mean is there a tool that enables me to test the capacity of this SIP server in terms of simultaneous calls that could be treated, the comsuption of bandwidth.. or any thing like this? I am in urgent need to such a tool, If anyone

HI ALL; Is astersik enable to translate between different codecs. I have couple of SIP-UA , one with (a-law) and the other with (g729), registered with my astersik box.Can astersik translate between alaw-g729 and vice varsa. Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Chris, Your were totally right with your suggestion. I have implemented the code that : a) does not merge multiple TokenFactor nodes in the DAGCombiner::visitTokenFactor(), if the resulting TF node would contain more than 64 operands. b) produces a bunch of TokenFactor nodes with at most 64 operands, instead of one huge TokenFactor in the SelectionDAGLowering::getRoot(). If we have n

Hi Dan, Thanks for commenting on this topic. See my comments in-line. ----- Ursprüngliche Mail ---- > Von: Dan Gohman <gohman at apple.com> > An: LLVM Developers Mailing List <llvmdev at cs.uiuc.edu> > Gesendet: Mittwoch, den 30. April 2008, 21:38:26 Uhr > Betreff: Re: [LLVMdev] FoldingSetNodeID operations inefficiency > > > On Apr 28, 2008, at 6:21 AM, Roman

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 Good day! - From Nouveau feature matrix page: >SLI or even multicard setups are very rare among developers. >You should start hacking with us, if you have such a setup. I have two GTX 770 cards running in SLI[1] and would like to help bringing SLI capability into Nouveau. I would gladly write code for it and I am a competent C programmer

This is from nanog; I presume there is significant interest from readers here not also on nanog.... I've edited it to only the interesting part... -JimC -------------- next part -------------- An embedded message was scrubbed... From: Daniel Golding <dgolding@burtongroup.com> Subject: Re: New VOIP Peering/Interconnection Mailing List Announcement Date: Fri, 14 May 2004 15:09:00 -0400

Has anyone had any luck getting a SIP trunk up and working between Callmanager and Asterisk? If so were there any steps you had to take that were not in the documentation on wiki? Blake

Hi, Has anyone on this list connected with Verizon's SIP product? We are currently undergoing interop testing with Verizon, and honestly, it seems like the most convoluted process. I'd be interested in talking with someone else who has gone through this and run a few things past you. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, My colleague installed a Asterisk home as company's SIP server and I would like to integrate the Quintum gateway (SIP) but the calls don't get through. Bellow is are the configurations on each side: Quintum ******** Primary Registrar = 202.69.190.244:5060 Primary Registrar User Name= sipquintum Primary Registrar Pwd= sipquintum Primary Proxy =

dear all, we need help for integration asterisk (sip) with mera we have configure for sip.conf and extentions.conf sip.conf [mvts] context=mvts type=friend host=10.10.0.2 dtmf=rfc2833 in extentions.conf [mvts] ; ; mvts exten => _01162.,1,SetCallerID(mvts) exten => _01162.,2,SetCIDName(to mvts) exten => _01162.,3,Dial(SIP/${EXTEN:3}@mvts) i need if i dial 01162 in mera replace with

HI; Thanks for your reply. The reason for why I am going through asterisk in such case is just "using asterisk voicemail service" I mean: ATA1 calls ATA2, suppose ATA2 is unreachable or he is not at the office, then the call reroute (my GK is able to reroute calls if the first route is not valid) to atersik for voicemail service. Do you think I can handle it with asterisk native

Hello, I am writing a program based on Astersik Manager which needs to put calls on hold and to redirect them to others extensions. I haven't funded any action able to do this. Is there a way to place calls on hold using Asterisk Manager Actions? Amaury -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a "Failed to

My fellows, We have Asterisk@home installed and we want to interconnect it with our existing quintum gateways.. any idea how to config that? Your time is very much appreciated.. Cheers, Jessie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050220/2518797c/attachment.htm

Hello, Asterisk Real time database worked on astersik 1.6.2.0 but now i am working on Asterisk to latest version which is 1.6.2.2 ,there is a a warning [Feb 24 16:26:14] WARNING[4053]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available [Feb 24 16:26:14] NOTICE[4053]: chan_sip.c:21500 handle_request_register: