similar to: sirrix NT mode

I am unable to make quadbri board to work in NT mode, in TE mode it works OK. I have set 5 jumpers to NT mode. Dmesg shows everything OK: qozap: Junghanns.NET quadBRI card configured at mem 0xd084e000 IRQ 9 HZ 100 CardID 1 qozap: S/T ports: 4 [ TE TE TE NT ] qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. from Asterisk side seems to be ok. pri show span 4 Primary D-channel: 12 Status:

Hi, I have just installed a Sirrux PC140 card for the first time. Managed to build the drivers and get * to load them on FC4, but it does not work. It seems that layer1 in the ISDN is not even activated. The same ISDN lines connected to a Samsung DCS works so it is not the lines. I am including my sirrix.conf and the output of some of the * Srx commands below. Any pointers would be

Hi guys, How's things going ? Got a bit of a weird one here that I've been unable to solve. I have a Panasonic PBX linked to a Sirrix Quad BRI card that is running in TE (ptp) mode in a Asterisk box - this then links through Internet to another Asterisk box via IAX2. When a user on the Panasonic PBX system dials the extension of my Sirrix Asterisk box, Asterisk answers and says

Good day all How do i set a connection between 2 asterisk servers via PRI In Bri I would set one to NT and TE How shoud the zapata.conf and zaptel.conf look And how should the cable be? All I got on the web was to set one to "pri_net"...this cant be all? And the cable > pin1 <--> pin4> pin2 <--> pin5> pin3 <--> pin6> pin4 <--> pin1> pin5 <-->

Hi guys, I'm running a Sirrix PCI4S0 quad BRI card in a box with Asterisk CVS-HEAD (20050614). Something I've come across is that with 'echocancel = yes' in /etc/asterisk/sirrix.conf the echo is unbearably loud - so loud in fact that the echo distorts. To remedy this I've set 'echocancel = no' and disabled the echo cancellation. With the echo cancellation disabled

Hi all, How are things going ? Is there a way for me to individually identify each BRI channel on the Sirrix quad BRI board. The reason I ask is because our client uses the "Asterisk Flash Operator Panel" to monitor its external lines and transfer calls from the lines to the various SIP phones. The "Flash Operator Panel" requires that we set a static value for each line or

Good day all What is the amount of calls that asterisk can handle,SIP and from/to PSTN -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301

Hello! I am selling hardware I have been using in various Asterisk test installations. If you are interested in buying all or any of the following items please reply off list. I got a few * X100P clones * Sirrix PCI4S0 ISDN BRI cards (+ 1 Sirrix S0 voltage/power supply) * 1 Polycom Soundpoint IP 500 (+ Plantronics headset) * Conrad HFC-S cards to give away. Some of the pieces are unused,

Hi all, As i tested with Asterisk+dahdi+libpri with openvox BRI with NT mode, the ISDN phone does not work. There is the setting and error. ***********************Enviroment********************************** Asterisk-1.6.1.18 dahdi-linux-2.3.0.1 dahdi-tool-2.30. libpri-1.4.11.2 CentOS-5.5 OpenVox B400P ******************************************************************** In my case , I set prot 1

Hi all, I have a sirrix bri card, connected to 2 ISDN lines.I used to get a lot of slips on it, so i put the 'master' setting to 'yes'. But every couple of hours the machine completly hangs, and i have to reboot it. I only get 1 or 2 slips,I dont know whats wrong? yusuf

Hi, I'm trying to connect quadbri between powered ISDN phone and ISDN line: ISDN <---1---> TE - * - NT <--2--> Phone I use quadbri, suse 9.2 and latest 0.2.0-RC7k bristuff. I've used sample settings provided with package, but do get strange error (I think that I have wrong setting for P2P or P2MP setting and cables 1 and 2). If I connect phone to ISDN with straight cable

Good day all I'm just wondering for interest sake I have a box,hostname=myname.co.za,running sendmail If I send mail to someuser@myname.co.za it try to deliver to the box,witch does not have the mail box.How do I tell sendmail that it should send mail to myname.co.za's mailserver. I know how easy it is to change the name but there's a lot of reasons why we can.It is not in the

Good day all IS there a way to personalise the voicemail message when you leave a message? Thanks Altus

Good day all Is it possable to set asterisk up as a cdr server for other voip units We got a quintum dx here and its got a option to log to a cdr server on port 9002 Thanks Altus

Good day all We have a quad bri card,installed on fedora core1,downloaded the latest bri-stuff that download asterisk 1.0.3 and zaptel 1.0.3 and libpri 1.0.3 All installed and working.BUT after 5min+ of talking it just drops the calls? Any reason why? Please help Thanks Altus

I have successfully placed a call into and out of the card from Asterisk using vISDN. The current vISDN snapshot now contains the PCI id's for the card so no patching should be required. Most of the initial testing failure was due to misconfiguration of vISDN by me and a bad entry in my sip.conf resulting in one way audio. So I have established that at least 1 channel of 1 port works in TE

Good day all So I've installed asterisk with my openline4 card and I've setup sip and I can do sip on the local network,we are using soft clients,x-lite. But... When a call comes in from the outside(PSTN) and the you dial the extension it forwards the call the the client and you see incoming call on x-lite,you accept he call....BUT there is no sound.It shows there is a call and you are

Good day all I've installed a few systems with quad/octo bri cards On these systems incoming numbers are ether the full number,example 12345657 or ether the last 4 digits,example 7654 But for some reason the latest installation incoming numbers comes in as extension "s"?? Is this something to do with the telecoms provider or a asterisk config? Please Help ore advice Thanks Altus

Good day all Did anyone get the eicon 4 bri working with asterisk and fedora core 3 Please Thanks Altus

Good day all I got my voicemail message working,thanks but now,keep in mind I'm using SIP We have,for example 4 people in our admin department.Each user has its own voicemail so that when their extension is dialed directly and not answered it gos to voicemail. But there is also a option to dial 3 for admin with will dial all 4 number in sequence.This I got working 100% but now I want a