similar to: Zhone channel Banks

This may be a long shot, but I suppose it can't hurt to ask! I have seen chatter regarding Zhone channel banks amongst the asterisk users primarily. In all of my searching I am guessing that the majority of people using the Zhone's are probably reading this, and with the support that I have seen from Zhone so far this is probably my best bet, I recently picked one up for a project

Hey guys, I've had a working Asterisk setup going for a while now, but am having problems with the Zhone Zplex 10b thinking that during ringing, an extension has answered the call when infact it hasn't. This only seems to happen on some of the ports and doesn't appear to be specific to the handset. Does anyone have any suggestions as to how I could go about fixing this (aside from

I've just received a brand new zhone z-plex 10b. At first, I was able to login using the default (admin/zhone), and started exploring with the configuration, trying to set it up to work with the TE410P. Somehow the admin password was changed, but I don't know what it is anymore. So now I'm stuck at the login prompt, and can't get back to the CLI.... Is there a way to reset the

Mine has been working well, but the only problem is that it doesn't support callerid (from the POTS side). > -----Original Message----- > From: John Schmerold [mailto:john@katy.com] > Sent: Tuesday, 5 August 2003 12:37 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Zhone Zplex 10 units > > Thanks for the Zplex heads up. > > Steven

Anybody ever put a Sangoma and a Digium card in the same server? Specifically a four port card from each company? -bill wlloyd@slap.net

We were just testing forwarding one of our numbers to a VoIP CLEC and ran into an issue that we've seen before but never figured out the cause of. It seems if you call us and immediately hang up - or if the call is forwarded by Verizon (which stills rings about ? of one time) the Zhone detects the ring and answers the call.. The Zhone however DOESN"T detect that it's no longer

Hi. I was wondering if the zplex in the dev kit could be configured to have all fxs ports, instead of the standard 8 fxo + 16 fxs. If so, anyone managed to do that? Matteo. -- Matteo Brancaleoni Powered by RedHat Linux 8.0 Linux User #153521 -----BEGIN GEEK CODE BLOCK----- Version: 3.12 GS d? s:- a- C+++ UL++++ P+ L+++ E- W+++ N++ o K- w-- O- M-- V-- PS PE- Y PGP++ t 5 X+ R tv- b++ DI D+ G e

I have just make zhone set2default to factory default setting. What do i have to do to make T1 interface zhone match with T100P.

We have a client with a Nortel PBX with digital phone sets. Due to T1 problems (old firmware), we are interested in trying a FXO channel bank. Is there a channel bank (or equivalent) which emulates Meridian digital phone sets? In order words, an FXO channel bank that's Meridian digital? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I have the last firmware for zplex, if you like i send it to you, about the second question 24s means 24 extensions so you can configurate as you wish as fxo or fxs. Att Yelson Vivas

Hi, I have a mint, like-new (used only once) Zhone Z-Plex channel bank for sale. It has 16 FXS and 8 FXO (part number: Z-PLEX-10-24S/O). It functions as a channel bank, router, and CSU/DSU, combining T1, analog voice, and data. D4, SF, and ESF line formats are supported with AMI or B8ZS coding. It has dual T1 jacks, one ethernet jack, one serial port, one V.35 port, and 24 analog pairs

Hi all, I know this issue has been discussed over and over again, but I'd like to voice my opinion while 3.0 is still fairly early-ish in the pipeline. So the issue is... API breakage. I understand and agree with the rationale why, namely faster development. But this principle should mean that for each breakage, the dude who makes the breakage should accompany the final commit (or something

On 13 July 2011 15:47, fly language <flylanguage at gmail.com> wrote: > I don't really buy the manpower argument. Updating the release doc when > breaking the frigging API is the Right Thing To Do and shouldn't take that > long, when done when the change is fresh in memory. I regularly make small API-breaking changes in the name of cleaning things up. Sorry! I'd be

I'm having a major problem at one of my branch offices with "Phantom Rings" on their asterisk-based phone system. The system was originally built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC card. The upgrade severely increased the frequency of the phantom rings. I've read everything I can find on-line about automatic testing and noise on the line and

Hello,

FYI http://www.itbusiness.ca/index.asp?theaction=61&sid=55298 did not like this by line in the story t"he CRTC has said it will likely regulate voice over IP the same as other phone services."

Hi List, I have a client that has an interesting request. He wants to have people call in to a conference room and all be able to talk however they should not hear each other. There should be admin access to for one user to call in and be able to listen in to everyone that is talking (they may want this admin to be able to talk to). Any ideas ? Thanks. Dovid -------------- next part

> I have a customer that has two buildings that are connected with a > fiber link. We have a single Asterisk server to cover both buildings. > Now the customer went and bought an overhead paging system for the > remote building and they want to integrate it with Asterisk. Is there a > device that can connect over IP or an ATA that has an audio output port? > The buildings

Can someone advise on how to go about finding someone QUALIFIED to make changes to libpri? We have a pilot stuck on hold, due to old buggy PRI software on a meridian PBX. Upgrading the meridian software is not an option, so....we would like to have libpri changed to compensate for the bug. Is this a digium only type fix? I've called digium support but they only offer support for their

Hey, I've checked all over and can't find what I need to know, so I'm posting here. I want to use Asterisk with my Primus VoIP service but it seems I need a username and password to authenticate with at Primus. Has anyone had any experience with this? How did you get it? Is it stored somewhere in the D-Link gateway they gave me? Thanks in advance and sorry if this makes no sense.