similar to: 5ess

The ISP giving net access at our office has installed a 24 port CISCO 2950 switch in our server room. I can buy 24 connections from them and get 12Mbps of Upload but each individual connection is restricted to 512Kbps. Currently we have requirement of 20 simulataneous calls so we purchased 4 connections from the ISP. Giving us a total of 2 Mbps of upload b/w but spread over 4 different connections

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

I'm having difficulty getting a PRI working between asterisk and a Merlin Legend. I have been attempting to get it going as esf, b8zs & 5ess (although I have tried other switchtypes...). I found the following info in the Legend docs: If the switch type is 5ESS, the protocol MUST be set for ATT Custom. If switch type is DMS-100, the protocol MUST be set for NTNAPRI The README in

Hi, I am looking to connect 66 analog phones to an asterisk box. I was thinking of a Xorcom astribank 32port (2 of them and another 8 port). this is because the phones have no near connection to an ip network, so replacing the phones in favor of voip phones+network cabling is kinda out of the question. In your experience, will these units support all the phones talking at the same time with

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

Hi, I am getting this message on the * console on my first pri span. Pri show span show it is down, and I can't make any calls from the span. Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27

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

I am connected with 5ess to a nortel switch. Running dahdi 2.1.04 and libpri 1.4.7 and asterisk 1.4.22 Everything is working for a short time. The other side reports back to me that he is getting slip-r errors from the TE122P card. After to many errors (like 11) the d channel shuts down on their side. Any idea what to do about this? I have not ran into it before. Jerry

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

Hey All, I have an Asterisk box connected to a Nortel Option 11C via a T1. In the Asterisk box we have a Sangoma A101C and in the Nortel we have a TMDI card. The Nortel is also hooked to the PSTN via a T1 on a different NTAK09 PRI card. I've included the Zapata.conf and zaptel.conf files below. Our issue is that when a call is sent over the tie line between the two systems, the audio on the

If you are looking for real high density VOIP termination I would look at > something like a Lucent APX 8000, configure correctly it can pass 2500+ > g.729 calls to the PSTN course we paid lots of $ for ours. > > Chris > Chris, My experience has been that the APX and TNT products require a single SIP proxy, how are you load balancing 2500 calls? If all of the traffic is

Hi, i have one Asterisk with a Digium E1 card, and a Meridian Nortel Release 11. I need to connect both of them. We are using MFC/R2 for this.. The Diagram: [ NORTEL ] ( AMI ) ------------------------------------------------ (DIGIUM) [ ASTERISK] we have green light at the digium card, and at asterisk we see all 31 channels as idle. But when i want to recive a call from the Nortel to the

Hi, Is libpri rpm version 1.4.12 for CentOS 5.6 made available ? [root@ ~]# rpm -qa | grep libpri libpri-1.4.11.5-1_centos5 [root@ ~]# cat /etc/redhat-release CentOS release 5.6 (Final) [root@ ~]# [root@ ~]# yum list updates | grep libpri [root@ ~]# Please suggest/guide further. Regards, Kaushal

I've tried to do it myself, but my head is now bleeding from hitting it on the wall so much. We need someone who knows asterisk and Meridian PRI cards to help! If required, we will pay for a day's consultancy in order to get this thing working. Or, do I need to scrap my plans to keep the meridian system (60 phones ...) ... Please say no .. :) Please contact me offline

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.

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Hello! In the book Modern Applied Statistics with S (4th ed), section 5.6 the concept of the "average shifted histogram" or ASH is mentionend. Also it is mentioned in the same section "The code used is in the scripts for this chapter" (from figure caption 5.8, analysis of the geyser duration data). *However*, I have trouble finding the code for that function! Admittedly, I am a

Hi- A customer of mine running asterisk has inquired if asterisk currently supports TBCT, that is the ability of asterisk to transfer a call back up to the carrier's switch to complete a connection, freeing up the B-channels on the asterisk box. I saw a reference to this feature in the asterisk "bounties" section on the Wiki, and someone added a note saying that this might be

Anyone have a practical experience/knowledge relative to why a 5ESS central office switch would require a "w" in the Dial statement to handle analog pstn-fxo calls? I fully understand what "w" is doing, just trying to better understand why a 5ESS doesn't accept dtmf a little quicker then it does. Does that switch make use of dtmf receiver cards (or something) that

Hello, I have a Sangoma A200 analog card with 2 FXO ports. It's working well with asterisk 1.4.22 and Zaptel. I decided to upgrade to asterisk 1.6/dahdi. I compiled and installed, dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.10 wanpipe-3.4.1 asterisk 1.6.1.1 My analog card is recognized in dahdi_hardware. However, asterisk cannot compile chan_dahdi.so. I've tried passing