similar to: IDAP T1

Hello all, We have a TE405P set up with span 1 running to a channel bank, a PRI running into span 2, and a PPP internet T1 running into span 3. We have the first 2 spans up and running without a problem. We have hdlc compiled into the kernel and after making the appropriate changes to zaptel.conf and loading the zaptel, wct4xxp, and hldc modules we can bring up the third span with the internet

Hello list, For a customer I need to detect blue alarms on his T1 trunks. His server is in running asterisk and zaptel 1.4. I have a tool to generate all sorts of alarms, but on generating blue alarm, zaptel recognizes them as red alarms. This is not good for us as we need to get blue alarm as blue alarm. I checked using: cat /proc/zaptel/* zttool Are these versions of asterisk and zaptel

Hello, We have an asterisk setup at our client's site using a TE205P. The line to telco is a 23 channels T1 PRI, however the line has random echo problems (about 5-10% of the calls)! Can anybody tell me if echo cancellation is really needed even at a T1 PRI to the telco? Because people keep saying when they deploy voip solution in Hong Kong using T1 PRI, there is no need of echo cancellation.

Hi all I have a server that has two TE412P (T1/E1+DSP) cards installed. One of them configured as an E1 PRI connected to PSTN and another one configured as a T1 E&M connected to Avaya PBX. Each card only uses two ports, so there are 2 E1 lines and 2 T1 lines connecting to this server. The purpose of this server is as a TDM trunk gateway that gets call from E1/T1 and then forward to an IP-PBX

I'll keep asking weird questions here: I have two T1 WAN circuits originating at my asterisk box and terminating at Asterisk servers at remote sites. Currently I use zaptel ppp assigned to all 24 channels for data. Now I would like to split off a few of these channels for voice trunks (I should use VOIP, but at the moment I do not have fast enough servers for the codecs) At one site

Hi All, I'm trying to configure a Digium T100P to talk to a legacy voicemail system. I have the signaling specs verbatim from the original manufacturer documentation as follows: [T1 Signaling] Service Type: T1,D4 format, AMI(Super Fram) Signaling: Four wire, terminated, E&M (Robbed bit) Start Protocol: Wink start; 250msec duration Dial Tone: Enabled Digits: DTMF, 4-digits DTMF: 50msec

HI, I'm trying to config a span from port 2 of a DIgium T410p to a Brooktrout TR114-P8V-T1 card. I have a T1-PRI from the TELCO in port 1 (thru first port) working just fine with Asterisk 1.0.3 - been working fine for some time now. No problem with dialplan PSTN calls. Now I'd like to "route" specific DID numbers in thru the TELCO T1-PRI and out thru the second port (span) on

Heya everybody. I have been peering into the code for zaptel for a while now, I am keenly interested in the dacs support, being able to apparently redirect certain spans to other spans. Not sure if this has to be on the same T1 interface or can be used between T1 interfaces on the same board or possible two different cards. Any information on the functionality of this would be greatly

I am trying to do a wire level tap on T1 equipment using digum equipment. So far most call monitoring hardware for call centers try to stay on the analog side requiring a lot of rewiring. I have already posted to the list about T1 "bridging" using DAC's support in the zaptel drivers. I still don't know if I can spy on channel information since I don't have any digium

Does anyone know if I could use modprobe command to find out rather than set the jumper on a Digium PRI card? I want to find out the jumper settings on the card without opening the box which will cause down time. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Could anyone with any experience with * over a non-PRI T1 help this newbie? I have a fractional T1 that is working fine through a channel bank, but I can't get any response on * using a T400P. My analog line and the extensions work fine, but the T400P port 1 shows a red led (what exactly does that tell me?). I brought up zttool which shows a condition red also, but ztcfg shows nothing amiss.

I have about a dozen SIP phones up and working, now I want to connect the asterisk box to our Fujitsu 9600 PBX. I currently have two dial-up servers conencted to the Fujitsu PBX that I built with mgetty/pppd and have the lines provisioned the same way as those dial-up server, ESF, B8ZS, and E&M wink start, so I have confidence in the guys who set up the PBX. I've built a loop back plug

Hello All, I have a Digium Wildcard TE410P Quad-Span T1 Card, when I do connect T1 lines it goes in RED. When I do connect the same line on a different Server (Same Model T1 Card) it works fine. How do I examine/diagnose my T1 Card for any hardware failures. I heard about loopback test , how helpful it is? Here are my configuration /etc/zaptel.conf span=1,1,0,esf,b8zs

We are trying to debug a connection between Asterisk and a legacy PBX (Mitel SX200). We turned on the Zaptel debugging and we get the following message quite frequently: Oct 12 07:14:09 localhost kernel: T1: Lost our place, resyncing ( 28 ) Oct 12 07:14:09 localhost last message repeated 3 times -- Oct 12 07:14:11 localhost kernel: T1: Lost our place, resyncing ( 28 ) Oct 12 07:14:11 localhost

I've got a system where I'm integrating a Nortel Option 11c with a Trixbox 2.0.0 system using a Sangoma A101 T1 card. (Running on a Dell PowerEdge 350) We've got things mostly up and running and all seems well... except... If I call from a SIP extension (X-lite soft phone) dialing 9xxxx where xxxx is an extension on the Opt 11, the call goes through to the Opt 11 but I have terrible

I figured it out. The old system (Televantage 3 and 4 I think) has limited specifications on the T1. After setting up the system, I was able to send and recieve calls. I still have some work to do like figuring out faxing and a floating receptionist, but this is a nice start. ----------------------------------------------------Televantage T1 Requirements: Framing: D4 Superframe or Extended

I'm having (I think) timing issues in relation to bridged T1-T1 calls via dynamic spans. Fax calls are intermittently working, but voice is fine. My box has a Sangoma A400 inside it as the primary Zaptel timing source. My T1 PRIs that are hooked to the box come in via a foneBRIDGE2 (dynamic TDMoE spans). PRI #1 is the telco and PRI #2 is an existing Comdial FX-II. For some reason, bridged TDM

I know this topic has been covered before - I just want to make sure I am understanding this correctly. | ------------------- | | | | T1 | Channel Bank | | ----------| | | | T1 Cross | | | | | | | | ------------------- -------------- | Asterisk | | | |

I've been having some problems getting a channelized T1 working with a T100P card. Perhaps someone can help: I have an Eschelon T1 coming into a Vina Integrator box. This box splits out the T1 into an ethernet plug for bandwidth and a secondary T1 which I plug into the T100P card. I've connected the two with a T1 crossover cable. I get a green light on the t100p card and can make

Hello. New to the list. We're in the process of deploying Asterisk. Actually, we're going live tomorrow, and I just found out that my Zaptel cards have been mis-configured. I'll preface this by saying that I have looked in the wiki, read through the samples, and attempted to call Digium (they're closed.) So I'm praying that someone on the list can help me out! I have two