similar to: Not enough lines available for Asterisk implemetation

I am so fed up with Digium cards. My company first owned a TE410P, I installed it in a Dell server and "enjoyed" its instability (we bought it months before Digium warned about the incompatibility issues). Then we switched to a TE411P for the hardware echo cancellation. Now we want to receive fax (< 20/day) on it and guess what ? Since April 2006 (again a few months after we bought

Hi, What driver should i use with APC BK-300C ? Michale Plourde Digicom Technologies

I have a Digicom TE110p card installed in our exchange. I have compiled and installed libpri, zaptel and recompiled and installed asterisk. I have configured udev as I am running Fedora Core 3. The problem that I have is that when zaptel is not running everything works fine. However when I start zaptel (service start zaptel) then I can make normal calls ok but the 'Playback()'

First, Something seems to be wrong with the list. I'm not the only person who has expressed seeing their messages either arrive late, or not at all. With that out of the way.. Is anyone aware of any type of failover device for PRI on asterisk? I've found the ISDNGuard, however it is currently not made in the U.S., nor does it run on U.S. power. Is anyone aware of a device that will

Hi all. I found some issue in implementation of finalizeObject function in MCJIT.cpp. If you look at the source code of the function, you can notice that machine code for second "owned" module will never be generated if it doesn't depend on the first one. More over it will cause a crash if entry point isn't in first module. Implementation of finalizeObject using for loop will

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Troy Settle > Sent: Wednesday, September 14, 2005 7:03 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] MAX PRI for single server (was:Not > enoughlinesavailable for Asterisk implemetation)

https://bugzilla.mindrot.org/show_bug.cgi?id=2101 Bug ID: 2101 Summary: Unaligned memory access on sparc in UMAC implemetation Classification: Unclassified Product: Portable OpenSSH Version: 6.0p1 Hardware: Sparc OS: Solaris Status: NEW Severity: normal Priority: P5 Component: ssh

If you are looking for real high density VOIP termination I would look at > something like a Lucent APX 8000, configure correctly it can pass 2500+ > g.729 calls to the PSTN course we paid lots of $ for ours. > > Chris > Chris, My experience has been that the APX and TNT products require a single SIP proxy, how are you load balancing 2500 calls? If all of the traffic is

I just implemented a system using a TE411P hardware echo cancellation card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as I always have. To my surprise calls out to the PSTN had a terrible echo. 1 - 2 second delay, and quite clear. The echo was so bad that I had to remove the hardware echo cancellation module from the card. We are only using the 1st span of this card right

Hi Radek, Sorry for the delayed response. I haven't had time to check your analysis yet, but you're probably right: MCJIT's support for multiple modules in a single instance is patchy at best. Do you have a test case (e.g. an lli invocation) that triggers this bug, or is this something you discovered just by reading the code? Cheers, Lang. On Thu, Nov 13, 2014 at 8:46 AM, David

Hi, I found E400P quad PRI card quite cheap (749USD): http://www.govarion.com/product_info.php?cPath=1&products_id=2&osCsid=68cdd6e3d08754 in comparison to te410p (approx 1500 USD ) http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE410P Now newer generation with HW echo canceling emerged (te411p). I'm not sure in what things those two cards

I just wanted to see if any one else has seen this or could help point me in the right direction on this problem. I have a TE411P card in my * box. I am running FC4 x86_64. I used to have two TE110 cards in the same box that worked without any problems. Since changing to the TE411P cards, I am getting random DTMF tones being produced on a bridged connection through the same Channel Bank that I

My TE411p does not seem to like to have two PRIs from different telcos (span 1 and span 2). I can get one working, but not the other. However, if I use vpmsupport=0 when loading the wct4xxp module, they both work. But here is the problem, vpmsupport=0 disables the on board echo cancellation. Any ideas? BTW, here is zaptel.conf span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs bchan=1-23 dchan=24

Hi Guys, I need your recommendation which card to buy?The TE411P or TE406P do they have any difference?I check their brochures they are differ only when it comes to PCI slot voltages.Which card best to use?Btw,Il be using supermicro board with 3.3V and 5.5V?Will il be experiencing problem during the implementation later on if i choose the wrong PCI voltage options?Will it depends on

Hi all, 1. Just bought a new TE411P and about to install it replacing the existing TE410P. I am assuming I need to set echocancel=no and echocancelwhenbridged=no now that it will be done in hardware, correct? 2. Is there any way to monitor hardware echo cancellation to ensure it is working (apart from being on a call)? 3. The Digium webite says "By supporting 16ms with 128 channels or

I am getting ready to purchase my first Digium card to start experimenting with Asterisk. Before I make my purchase, I wanted to make sure I'm not going to have issues with these cards (need to see what the specs are on my box, 5V or 3.3V PCI ). I will be using Asterisk @ Home, so will be Asterisk v1.0.9. I took a quick poke at the lists, and it appears several people have been having

Every once in a while calls get interrupted with a brief silence followed by a DTMF tone on our T1 lines. We are using a TE411P card. It does not happen on SIP connections, just the T1. Has anyone experienced this or know a possible cause? Thanks, -Bill

Hello, I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is talking to an Asterisk server via SIP for call origination and termination. Seems simple enough, and it works for the most part, but: 1) Caller ID name data comes in on the PRI, but doesn't appear to get handed off to the Asterisk server via SIP, at least not in any format that Asterisk

hi! anyone please recommend/guide me of purchasing a resonably high performance server system regarding processor(s) & motherboard (+ other compulsary peripherals i.e. VGA, Soundcard). Mentioning up-to-date vendor+model will be more helping I've to use Digium TE411p Quad E1 card signalling on the E1 is SS7 no. of simultaneouse calls is 100+ regards -------------- next part

Hi all, I replaced a TE410P Rev C 1st Generation Firmware with a TE411P without any cfg changes (zaptel/zapata). As a result Asterisk crashes on outbound from PRI4 going to PRI1 calls: Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1