similar to: Zaptel card AND Ztdummy together?

Can someone who is knowledgable in the traditional telco space please give me a layman's explanation (or point me to an appropriate url) of LBO as per the zaptel configuration file? # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1)

If you see a wj0011 version of firmware for Zyxel Prestige 2000W floating around (I found it in a German forum), KEEP AWAY. It completely trashed the wireless networking in my phone. -- ========================================== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237

P.S. I _had_ read the other posts that suggest changing to unicast sntp mode. This didn't help. I eventually setup a new ntp server on my LAN, and used it as a sync source. Everything seems OK now. Obviously a problem with my other ntp server. Cheers. -------- Original Message -------- Subject: Cisco 7940 - Disappearing Clock Date: Thu, 28 Jul 2005 11:50:35 +1000 From: Rod Bacon

Let me know if I should post this on the asterisk-dev list instead. I am building a Linux-Vserver (http://www.linux-vserver.org) host system that will have several guests running Asterisk. Since the guests can't load kernel modules or do other dangerous stuff, but can access them I built zaptel 1.4 and it is now loaded by the host. The issue I see is there will be no "Zaptel"

I was wondering if anyone had experienced the following with asterisk stable. After a period of time (can vary), If I stop asterisk and try to run ztcfg -v to reinitialise my quad e1 card, the server will lock up. Sometimes it's a complete lockup, where it won't even return pings, other times it seems to be "partially screwed". -- ==========================================

I am wanting to front-end a legacy PBX with an asterisk box. I have done plenty of asterisk work over the last 6 months to PRI circuits, but not with a PBX being involved. I know I can use asterisk and digium cards in this manner, but do I need separate cards for the PRI -> Asterisk side to the Asterisk -> PBX side, or will a 4-port PRI card do the job? (I already have a spare one of

I posted something on this a week ago, at which time I was told that this was an 'old' issue. Since then, I've spent hours looking, but can't find the answer. For some reason, some of my CDRs (both to CSV and MySQL) are being written with the wrong callerid. As best as I can determine, they are being written with the CLID of the _last_ caller to access the specific ZAP channel

I've posted this to SNOM, but was wondering wheter anyone here has issues with SNOM 190 phones not showing the correct DST adjusted time (using the latest firmware). -- ========================================== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 ? ?Fax: +613 99401650 FWD: 512237 ? ? ? ? ? ? ? ? ?

When I dial a PSTN (via * with Digium Quad-E1) number, I hear two sumultaneous ring tones. One is coming from *, the other I assume is from the ATA. Is there an intelligent way to get around this?

I have googled until blie in the face, WiKi'd until physically exhausted and searched through every Asterisk repository that I can find, all to no avail... No matter which version of SpanDSP I use, with which version of libtiff, Asterisk, ... I simply cannot send faxes. I can receive faxes *perfectly* (sending from standard fax macine on POTS) into tiff file (via Digium analogue card),

Will the CVS HEAD version of the Zaptel drivers work with the STABLE branch of *? -- ========================================== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270 ==========================================

I'm new to H.323 and I have noticed that there are two separate channel drivers for * available - the inbuilt one, and oh-323. I had trouble compiling oh-323 with the current cvs stable, so I tried the inbiult one (with specifiec recommended versions of openh323 and pwlib). It compiled cleanly but I am told that it is not recommended (unstable?). Can someone with first-hand * H.323

The one I like: http://www.rhetorical.com/cgi-bin/demo.cgi is toast. I think they went broke or got aquired by someone. Also, is there a Festival voice that sounds as good as Rhetorical or the AT & T stuff? The default one is barely legible. Since Festival is a little brutal to configure, I'd like to get someone's recommendation then go through the pain of reconfiguring it only once.

Do broadvoice limit the number of concurrent calls that any given sip registrant can make? What about other similar providers?

Does anyone have any recommendations for a phone that has easy to understand/implement central provisioning? I've used CISCO 79XX phones, and they're great (but too expensive). I like Grandstream phones, but their provisioning sucks. What is everybody else using in large environments where individual config is not an option? ---------------------------------------- Rod Bacon

We have just had an ISDN PRI service commissioned here in AU (using Powertel as provider). I have called them and ensured that we have the ability to set Caller ID on our service to any number in our 100 number block, and I have been assured that everything is OK from their end (unlikely). Every time I drop a sample.call in to /var/spool/asterisk/outgoing with the Callerid: option set, the

I know that this has been a hot topic in the past, and that there are a number of views about the overall suitability of RADIUS, which billing methodology is best, what is the meaning of life and such... What I would like to know is has anyone found an open-source billing platform that performs basic billing functionality (pre/post) from RADIUS and/or Asterisk CDR and is written

I know this is casting a wide net, but If you were charged with building a large, public VOIP network with multiple PSTN gateways, the capacity to carry a lot of traffic and bill clients accurately, what pieces (brands, makes, models) would you use to assemble the solution? Assume that $$$ is not an issue. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've managed to build the zaptel modules including ztdummy; ztdummy is installing fine in the modules list and the relevant device structures are present. lsmod | grep ztdummy gives:- ztdummy 5160 0 zaptel 186916 1 ztdummy rtc 12372 1 ztdummy Where I'm stuck is I am now at a loss as to how to configure my /etc/zaptel.conf and

Hi there I am currently using Asterisk with Meetme on a 2.4 linux kernel. I am using Ztdummy with usb-uhci driver for timing. It seems to work ok, although I haven't tried it with more than 5 users. However, I am now looking to move into a production environment and some people have said that it is better to have a Zaptel card. I would be installing Asterisk on a 2.6 linux kernel for the