similar to: Zaptel HEAD with * Stable?

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

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 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 ? ? ? ? ? ? ? ? ?

I have several * servers, and have noticed something most annoying on recent builds. On an old server, running a CVS 1-0 checkout from 10th May, incoming zaptel calls (ISDN PRI) start processing the dialplan immediately, and look something like this... "Accepting call from 'XXXXXXXXXX' to 'XXXXXXXXXXXX' on channel 0/2, span 1" My other servers are running either the

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)

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 anyone has any hesitations in upgrading their 405P (or 410P) to V2 of the firmware, read below; I installed one today (turnaround time around 2 weeks to Australia, inc. economy freight in both directions... impressive!) and have noticed immediate, significant improvements. Audio levels are better (have set tx and rx gains back to 0.0) and missed frames have gone (popping, clicking,

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.

I'm getting these messages every 7-10 seconds. -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP '532' at x.x.x.x port 56988 expires 60 -- Registered SIP '529' at x.x.x.x port 51444 expires 60 -- Registered SIP '529' at x.x.x.x port 64044 expires 60 -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP

Nobody has been able to answer this. Not even Digium at this stage, but I'm hoping someone here, smarter than I, will be able to. We are running some TE406P (upgraded 405Ps) cards performing mainly PRI bridged calls. After a server is brought up, calls sound absolutely perfect. Over time, delay (latency) creeps into the calls. What is really weird about that is apparently with the new

I am having trouble getting trunking to work between a couple of my servers. All servers are running 1.0.9 stable version, and are working perfectly. All have a Zaptel card of some description, so timing is not a problem. Each server has a definition for each other server, using RSA auth, and qualify=yes (just so I can see ping times). trunk=yes is on globally, AND in each peer/friend

Hi all ! What is the current status of the Dialogic channel driver ? Is it available ? Is it commercial ?? Any info ? Thanks

I have asterisk connected to a PSTN line with FXO interface cards from Digium. When I first start Asterisk it sounds fine.... after a few days of running I begin to hear alot of echo on calls... restart asterisk and it is fine again. Any thoughts what this might be?

I have looked through the archives, and can only find old references to this problem that appear to be no longer relevant, so I thought I'd ask again. I am having a problem with periodic breaks in audio over an IAX trunk. The interruption only happens in one direction, and (I think) only with clients built on the open source libiax. Codec is irrelevant, and jitterbuffer on/off seems to

I just wanted to post here and let everyone know that the TE406P (quadspan T1/E1 with hardware echo can) kicks some serious ass. We've been running a PRI now for over a year with Asterisk (every single call in and out is through two Asterisk boxes, including faxes) and while the software based echo cancellation is more than adequate, we'd get the occassional "edgy" echo and

I am running ntpd on the same machine as asterisk in order for the grandstream phones to display the time. After a while the time display fails until the phone is re-booted. Has anyone run into this problem before? Is it simply a bug in the GS firmware? Sean

I have a number of servers with TE405P cards. The servers are DELL 1850's (which I _NOW_ see are listed on the digium "not recommended page" because of the ethernet interface). The problem I have is only during bridged calls. If I place a call into a service hosted on the box, or out to a VOIP phone, audio is crystal clear. If place a call "through" the box (a bridged

Can anyone tell me if Asterisk would speficically benefit from the reduced latency of a preemptible Linux Kernel? I know it was recommended against in the early days, but I'm wondering if there are any updated opinions? -- ========================================== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600