similar to: Generic question about PBX PRI installs

We've been trying unsuccessfully to connect our Meridian Option 81C to a TE110P via PRI. We've followed the directions in asterisk-meridian-a1.pdf (link on http://www.voip-info.org/wiki/view/Asterisk+legacy+integration), but it doesn't seem to work on our 81C (even though many, many users report it works very well on Option 11's). Has anyone had any success in getting the above

We are currently working to trunk from a Nortel 81C to an Asterisk Server 1.4 running on Red Hat Linux. We have two PRI trunks which work with the exception of the clock slips, which is causing the Nortel to reset the PRIs once a hour. Thanks for any suggestions. 81C MSDL Asterisk Digium TE110P REQ prt TYPE adan dch 10

Anyone interfaced * to the Nortel option 61c pbx via T1's, pri, etc? Need to begin planning the implementation, purchase cards, etc. Any recommended approaches, configuration problems, etc? Off-list is fine if you'd like. Rich radamson @ routers . com

Hello all, I have Asterisk 1.2.1 and a TE110P connected to a Nortel Meridian Option 81C system. The PRI line is currently setup as DMS100. Here are the relevant lines from zaptel.conf and zapata.conf: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone = us zapata.conf: [channels] language=en context=from-internal musiconhold=default switchtype=dms100

Hi, I have asterisk running a single T1 card with a connection to a nortel CS 1000. All calls to extensions, local and long distance are working just fine. My issue is this: The nortel CS 1000 supports connections to and intercom system that is just line level audio to speakers. When my PRI tries to call this "extension" it is never answered so my AGI never runs to play a message

I wonder if anyone has come across this odd behavour with a T1 PRI using NI2 signalling from a Nortel switch. Sometimes, when bringing up a PRI trunk, a channel gets into a state where asterisk can't request a channel, and gets reason 0, but the channel is not busy. The only thing so far that clears this state is to make an incoming call to the channel, which succeeds. After that, outgoing

We are in the process of installing a PRI line and we are going to connect it to an Asterisk Box. Verizon called us today to find out some information. I am surprised that they have never heard of Asterisk or Digium. But anyways, they needed some information in order to set up the circuit. Does the TE110P support NI1 or NI2? (I think the answer is both) What is the number of digits

I am terminating a PRI, NI2 signaling into a PBX (My company's PBX not the PSTN) from an Asterisk server. Caller ID number appears to be transmitted caller id name is not being transmitted is their a compile time flag on libpri that I need to un-comment to enable this feature or is this is unsupported for now? Thanks Chris. -------------- next part -------------- An HTML attachment

Hi all, I have asterisk 1.4.26 (and I tried 1.4.29) connected PRI all 23 lines and for the most part everything works. Dialing out on 23 lines to phones works fine. I have to use the Local channel to call the intercom system (from call files). If I only call 1 intercom system at a time so it uses DAHDI/1 everything seems to work as I can call all 8 intercom systems and play a message. The

Greetings, This may be a bit arcane but does anyone know what the contents of a facility message should be for initiating a TBCT on an NI2 ISDN. I've been trying to get it to work on a DMS100 for the last four months to no avail. The message I am currently sending makes it to the switch but is returned with unknown message. Perhaps someone here has done it before and can help me out.

I've been going back and forth with my telco for several days, trying different configurations to get a new PRI to come up. The bchannels are all up and the T1 is not in alarm status. The dchannel refuses to come up however. We've tried ni2, qsig, and now dms100 for the switchtype. The telco tech I've been working with says that he's been sending "reset all channels"

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

My legacy PBX accepts CID number, but not name. My old PRI vendor never sent the name, so there was never an issue. I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI - asterisk - PRI - Legacy. Any calls from asterisk (sip and iax extensions) which have callerID set, will not connect. The legacy PBX hangs up, but asterisk thinks that it is still ringing. I have added

I am having my colo set up 2 PRI's for my new asterisk implementation. They asked the following... ##SNIP## What type (NI2, NTI, 4ESS, or 5ESS) and whether they want to be USER or NETWORK. If the equipment is flexible, NI2, with us as NETWORK is preferred. ##SNIP## We are using a digium quad span T1 card. What is the recommended setup? Thanks, Chris

Hello All, I am getting my first PRI installed in a couple of weeks and I wanted to ask for a little advice. I have a single span Digium card I will be using for the install. Id there a benefit to which protocol I use? When asked, I told them to set it up as NI2. The PRI is through MCI and will be used for local and long distance with DIDs and features like CallerID, etc. Any advice would be

In the next week or so, we'll be turning up a new T1 Pri from Cox Cable in Omaha using a T100P (installed but not yet configured). While discussing the interface parameters with a very knowledgable Cox engineer, we decided on some of the basics including b8zs, esf, dchan=24, callerid, 5 digits of called number forwarded to *, and switchtype=national (NI2). They indicated their CO switch is a

Does anyone know of a way to simulate the signaling of a PRI trunk for testing/setup purposes? I realize this may be a rather naive question, but I was wondering if you could take a TE110, for example, and using a crossover cable (or not?) and some means of emulating the NI2 signaling protocol connect it to another TE110 on another machine to test and verify an installation before the telco

Following is my scenario to connect back to back PRI of two asterisk server. PRI cards are Sangoma A102D [Asterisk1]------------[PRI]-Cross Cable---------[Asterisk2] Asterisk1 ; Span 1 (MASTER) switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_net channel => 1-23 Asterisk2 ; Span 1 switchtype = national ; commonly

folk, I see from the archives that the issue of nortel handsets not sending dtmf tones to asterisk has been discussed a couple of times, but there is something I quite havent seen answered yet. Is this dtmf issue a problem with the nortel handsets or the PBX itself? If the handset were changed to another one, would this issue be solved? Or must the PBX be changed completely? rgds eb

Maintainer: Express Line Date opened: January 17, 2006 Status: Open Value of bounty: $5000.00 Licensing for code: We retain intellectual rights to the underlying source code. We need Asterisk (stable version) to be able to perform a 2B channel transfer for a NI2 B8ZS PRI line. We can't use a channelized T1 at the time for our work. This feature is commonly called a call transfer on analog