similar to: Re: Auto Answering PSTN --> Asterisk using X 100PCard

PC: HP Vetra VL400 Mainbood: Intel815 BIOS: Phonix 4.0 release 6.0 OS: REDHAT 9.0 I installed the X100P in PCI slot 2 and disable the USB port, serial-port and parallel-port in BIOS. I can't found the X100P card in " cat interrupts" But I can found the card in the "cat ioports" use "lspci" I found the X100Pcard use the interrupts 11 too. Who can help me to solve

Hi all. We are getting our first PRI line to use with Asterisk and one of the technical specifications is about framing, structured or unstructured. The main difference about them is almost clear for me: http://ckp.made-it.com/g704.html says: "G.704 is the framing specification for G.703. A carrier can 'steal' a 64kbps time slot (TS0) from a 2.048 Mbps line and use this to

Hello List Just want to check if anybody else is having this problem. Every time the PRI connections are disconnected, the card freezes, and I have to reload the driver, to make it work again. We are very seriously considering switching to Sangoma at this moment, due to this and other problems, but I want to know if there is a solution, and to make sure it isn't asterisk that's freezing

Greetings, Our telco called last week, saying that a lot of channels on our PRIs are blocked. And with blocked they have the following description in the Siemens exchanges: BBAC BLOCKED BACKWARD This status is set when the partner exchange has a blocking set and the signaling of the trunk (non-CCS7) is able to report this blocking in the backward direction. This status can

Hi, Is it possible to use 4 TE405 boards in one server ? It mean, to have 16 E1s on just one server. Can somebody tell me how many boards is it possible to have on one server ? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060605/29e9d42d/attachment.htm

I have a TE405/TE410P card that was working on 1.0.X I upgraded the OS to Centos 4.5, Updated asterisk to 1.4 and zaptel to 1.4.5 and libpri. I copied all the zaptel and zapata and extensions.conf files from 1.0 I did update extensions.conf from 1.0 to 1.4 commands. I cannot get the card to work in 1.4.10. AHHH! I see with zttool that the T1 is in Green, I see calls coming in as the bits go

Hello ! I 've some questions how bridging of ISDN calls is done. Assume an asterisk system with a TE405 card equipped. (PRI1 - PRI4) An incoming ISDN call on PRI1 is transfered back to PRI3. Unless there is DTMF detection or other things involved, the bridging is done without Asterisk. Does this card have a some sort of cross connection ? Does the PCM leave the card ? Or is there some DMA

Hello asterisk-users, I have solved interconnection between Digium TE405 and Matra PBX. I plug this card to another computer and with the same configuration parameters card now works without problem. First server has 2xPIII/1GHz and 256MB of RAM, Adaptec SCSI adapter and SCSI system disk. I don't know now, what chipset it was, but mainboard was from Supermicro, then I pretend Intel chipset.

Hello asterisk-users, I have problem with E1 line between Asterisk computer and our PBX Matra: asta*CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 My

Nothing from me is posting to the list either. Julian

Hi, I would like to have some more informations on asterisk, I want to setup a linux based pbx and asterisk seems to be the best solution, I have some questions for configuration: 1) I have a PRI, so I must buy a digium card to interface with PRI, right? 2) If I connect an ethernet card from the pc (equipped with a digium card connectd to the PRI) to a switch I can connect users to this

Hello Folks, I'm trying to decide here between a few cards for connecting an Asterisk box to a single E1 channel (either PRI or R2 signaling): - Digium E100P: has been replaced by the TE110P below, but can still be had at places like digitnetworks.com for $475, and I guess there's always a place for good-olde-obsolete cards in the world as long as they work :-) - Digium TE110P:

We are doing this with the latest spandsp, iaxmodem and hylafax. Seems to work very well for us so far. -Jonathan > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michael Gaudette > Sent: Tuesday, March 21, 2006 3:34 PM > To: 'Asterisk Users Mailing List - Non-Commercial

What platform should you suggest to use asterisk ? I tried with SUSE now all the time but there are too many problems with the updates. On is the development platform on which * is developed ? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050611/1672c2ad/attachment.htm

Dear All, I am currently have this problem in which I am sending call out from the Zaptel TE405 to a VoIP gateway. But the problem that the call over to the VoIP Gateway will always have a fake ring tone. Can you please give some pointer how to fix this problem? This problem is existing in my Asterisk 1.2.1 box. Also when compiling Asterisk 1.2.5 and tried to run it with the Asterisk-addon,

Did anybody know, Is it possible to establish a ISDN DIAL up Connection and Analog Dial up Connection (V90) trough asterisk with Digium TE405? Thanks a lot for help. Nico Giefing -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060328/3b657058/attachment.htm

----- Original Message ----- > Ooooh, I've got it working a treat, thanks to some great help from people on > this list. I *am* currently writing up a how to, and will post it as soon as > I'm done. > > Basically, take the line from the meridian, and chop the rj45 off. (!) put > another RJ45 on, but with the wires crossed (IOW a cross over cable). Plug > the meridian

I was playing around with the Flash Operator Panel, and came smack into a brick wall. We have a * box linked to a legacy Meridian System using a EuroIDSN link (TE405p) with 10 channels enabled. I also have several SIP extensions. What I wanted to do was to have a button for each of our (say) 32 users, 5 of which are on SIP. That leaves the other 27 on Zap. A potential of 27 users on 10

I'm sure that this question gets asked frequently, but a quick perusal of the list archives shows that it hasn't been asked in a least a month or so, so pardon any repetition. What is the current state of asterisk on linux 2.6? I ask, because I spent yesterday giving it a whirl, and everything seems to go just fine till the very last minute. Zaptel, libpri and asterisk compile just

I have a channelized T1 coming in from our telco, terminated onto a TE405. There are three channelbanks serving internal analog extensions, and about 10 Cisco 7960s. I have no reports of echo on the analog extensions (as expected). The 7960 users complain of occasional echo (seems like 1 in 5 calls). Only the SIP user hears the echo, not the caller. I have echocancel=yes, echotraining=yes,