similar to: Re: Some questions about Asterisk and reliability

Hi, I have some experience in sending SMSs using smsclient. I call the german Vodafone SMSC (01722278020), and smsclient takes approx 20 secs to send a SMS. The hardware is an Sedlbauer ISDN card. Now, I want to do the same using asterisk and a digium PRI card. I dialed using the manager with: action: originate channel: Zap/g4/01722278020 ... I assumed, the call will fail, because the remote

Hi, Whenever i sync certain files in ( like /etc/tnsnames.ora ) from our hostA to hostB, rsync always print out /etc/ it prints out the files modified. it does that for every directory name in the path of files i update... Is there a way to suppress that output, since it is not modifying that directory ? Thanks Daniel Bourque Unix Systems Administrator BTS Platform Services Learjet Inc.

Hi, I'm sharing an NFS mounted directory via samba. When a client connect to the share, he can browse it fine, but any attempt to read data, like copying or looking that the file's properties results in a lockup on the client, and an eventual time out. On the server, a lock ( DENY_NONE 0x120089 RDONLY NONE ) is created an never goes away, even after the client as

I'm first to buy 5 pack. Even for > $30.

Whicch version of zaptel and Zapata should I use with 1.0?

Can someone point me for good VRRPD (rfc2338) implementation on linux. Some stable and live project Thanks _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/

Anybody sucessfully got stable 1000Hz clock without Digium harware and kernel 2.6? We need to consult some peoples how to clock asterisk stable with exactly 1000 Hz without much kernel/drives patching/tweaking. Some test results we made so far: 2.6 with digium card - stable 1000 Hz. 2.6 with ztdummy - uses RTC and the clock is 1024, not 1000. 2.6 with some Realtime kernel patch - provides stable

Has anyone succesfully integrated * with a cisco voice gateway ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031214/3b1ba7b3/attachment.htm

Just came by this link So I'm posting to keep the community informed. I don't use or endorse this product. I'm just letting people know about it. http://www.indosoft.ca/audioconferencesystem.htm Audio Conferencing System & Teleconferencing Solution that connect seamlessly over TDM and IP networks. This audio conference system include a comprehensive set of features and easily

hello, i have some questions, tryin load KX-TDA100 console(ATS programming) but have window: "This model of PBX not supperted ...etc" and in xterm : fixme:exec:SHELL_execute flags ignored: 0x00000004 fixme:imm:ImmReleaseContext (0x10020, 0x1270b8): stub tryin XP/2000/ emulation not working :( what i must do to load this? thanks

I've reading this lists few months. We are small company, that makes some system intregration, development and deployments in VoIP scene. Completely under linux. Today i have 6 machines with asterisk, huge test base - including devices like AS5350, Audiocodes gateways, ATAs, IP phones ... Now is time to make a decition for including * in our future projects. Main goal for us is the Stability.

Need to buy 2-3 channel banks for some asterisk deployments...

Must accepts wire transfers and ships to Sofia. Thanks

Yes, i know that there are many ISDN card on the market. But when i spend money for ISDN card, i prefer to be Digiums, to get all support and help Asterisk :).

I just got my Access Bank 2 (of course i'm happy now:)). Just need the wiring scheme Thanks

I just need to buy 5-6 ADSI Phones. I wondering which of these models to choose - aastra 390 - I don't know is this an ADSI phone at all. Is there versions with and without ADSI - aastra 350 - I'm sure that it have ADSI. If there is some other good working model, it will be great, if someone points me. Thanks

Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco

Just spended ~ hour googling - all boards are based on GC-XX or I750X Chipsets - all for Xeons. There also some boards for Pentium 3. Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4 with 800Mhz FSB. Thanks

sorry... typo.... the problematic phone has the 192.168.0.13 the asterisk has 192.168.1.211 when i connect a snom phone on the cable that was in the soundstation 6000 before and configure the phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP... it would be helpful if someone, that has a running soundstation ip 6000 could send the configuration... :-/ regards, yves Am

Hello, I am doing a confirmatory factor analysis with package Lavaan. My model consists of 5 factors and I would like to calculate the factor-reliability and the average variance extracted (AVE) for every factor. What function can I use to receive this measures? The Lavaan Package doesn't have this measures. Best regards, Ina [[alternative HTML version deleted]]