similar to: Is there a product to simulate a PRI trunk?

Looking for someone with a successful experience similar to this; I have a need to cross connect a 3COM NBX PBX PRI interface to asterisk, but over a long distance. We do not need any IP connectivity and the solution requires G.711u audio so there is no benefit to using IP. Has anyone here successfully cross connected any PBX PRI interface expecting NI2 PRI signaling B8ZS/ESF with an

Robert, > Is there a product to simulate a PRI trunk? (Robert > Goodyear) TDMoE emulates a T1. ;) Once the TDMoE link is up, Asterisk just sees 24-lines that appear to be a T1 instead of having to deal with all of the complexities of VoIP. This is useful, since probably 75% of the utility of VoIP is really just the fact that it can run over a network. It's also handy because it

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 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

"Pacelli, Louis M, ALABS" wrote: > > Hi, > I apologize for sending in this problem via email, but I had trouble using bugzilla. Please use openssh-unix-dev at mindrot.org for problems with OpenSSH Portable (ie anything that's not OpenBSD). > I'm trying to install openssh-3.7.1p2 > When I run the configure step, I get the following message: > >

Hey, I'm up and running fine with 30 Polycom 500s connected to Asterisk 1.2Beta on Cent OS 4.1 with a Digium TE110 connected to a PRI line. Nearly every hour, almost on the hour I get this: Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/1 successfully restarted on span 1 Sep 29 23:01:38 VERBOSE[3567] logger.c: -- B-channel 0/2 successfully restarted on span 1 Sep 29

Darren, I went to install zlib/openssl and openssh on one of my Sun Servers(Solaris 2.7) and they would not install. Is there a website where I can get Sun versions of these products? Thanks, Lou -----Original Message----- From: Darren Tucker [mailto:dtucker at zip.com.au] Sent: Saturday, November 22, 2003 9:35 PM To: Pacelli, Louis M, ALABS Cc: OpenSSH Devel List Subject: Re: zlib missing when

Hi, Is there a simply way to allow dialout from ISDN modem to outside number through Asterisk? I've got an server with an Asterisk and the following cards: 1. TE110 -- to telco 2. TE400P with one FXS to analog phone 3. Two HFC-S based cards in NT mode I'd like to connect ISDN modem to one HFC-S card and allow dialout through TE100 to some external number (being more precisely, to some

I'm little bit confused which Digium hardware is certificated for use in Europe. It looks like new cards are certificated, like TE4XX series. What about TE110 or TDM400P? Can someone confirm that? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060309/1d1c65a1/attachment.htm

Hi I've bought the Wildcard TE110 some days ago but I'm unable to get it to work with Siemens HiCom 300. I've tried this so far: 1. I've used standard cat5 cable cut off on one edge and twisted wires 1 to 4 and 2 to 3 which is according to cisco a short circuit for E1 lines. Wildcard takes a few seconds and sets the link to green (OK). 2. I've tried to connect our running

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

OK I have been battling with echo problems with asterisk on ISDN for a few weeks now, and still can't solve it (although I think I have tried everything I can find.) I will try a post everything I think is possibly relevant that I can remember with the hope someone can point me in the right direction. current spec I am using (Trixbox 1.0 was still in beta at the time an didn't detect

Dear all, May I do multiple integration using R? I was looking adapt but it is saying it integrates a scalar function over a multidimensional rectangle. I have integrand of several variable and upper, lower limit too variable. I wanted to see the result using adapt (though it is not for this purpose, I suppose) Func<-function(x){(x[1]*x[2])} adapt(2, lo=c(0,1), up=c(1,x[1]), functn=Func) it

Hi Guys, I have several * servers connected to T1 PRI's from various service providers in multiple locations the US. All the * servers use the same hardware with the same OS and * version. When connected to 5ESS Switches, using the NI2 (national) PRI protocol, the CID name and number come across fine and populate into the * CDR fine. I connected to a DMS-500, NI2 (national) protocol and

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

what happens when you drop your gains? use /etc/asterisk/zaptel.conf and fiddle with tx and rx values. Works, most of the time. -----Original Message----- From: John McEntee [mailto:john@mcentee87.freeserve.co.uk] Sent: Wednesday, October 11, 2006 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Echo problems on ISDN. (mainly incoming calls) OK I

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

I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've the following kind of data Time Country Values 2010Q1 India 5 2010Q2 India 7 2010Q3 India 5 2010Q4 India 9 2010Q1 China 10 2010Q2

I have an NI2 PRI connection to a 5ESS switch. I am explicitly told to use NI2 as the protocol, not 5ESS. I am running Asterisk 1.2.4. When making calls to the PSTN, Asterisk sends the calling name as part of the display information element in the SETUP message. I need to be able to send the calling name as part of the facility information element in the SETUP message. I've seen that work