similar to: foneBRIDGE2 vs. foneBRIDGE2-EC

I'm scoping out HA for a relatively simple Office/Call Center PBX. Current setup uses a TE412P with 4 PRI our telco with SIP hard/soft phones for users. Some outbound also goes to a SIP provider. Active/Active looks to be too much hassle for an installation this size, so we're looking at adding an extra * in an active/passive configuration with Linux-HA in between them. Does anyone

Hi I've published my Asterisk/foneBRIDGE2/heartbeat setup: config files, scripts... along with a brief description of the architecture and working of the cluster. It's available here: http://www.bisente.com/blog/2007/08/26/asterisk-cluster-fonebridge2/? lan=english Hope somebody finds it useful. :) Regards -- Vicente Aguilar <bisente at bisente.com> |

Hello List Does anybody have any experience with the FoneBridge line of products from RedFone? I think their HA implementation sounds interesting, and like the prospect of having dedicated hardware for our PRI connections. Kind Regards Jon Leren Sch?pzinsky

Hey guys, I am trying to diagnose a hum in my FXO lines I am using an Adtran 750 with 8 FXO ports. I am getting a pretty bad hum on the line during a call. I have checked the Telco side of the 66 block and there is no hum there so it's my problem to fix. I have tried to lower the gain but that reduces the call volume to much. Where else should I be looking? Setup as follows Dell

I've been asked to look at deploying Asterisk in a high availability environment and I've been looking so I've been searching for methods to decouple the voice PRI circuits from the Asterisk server so failover to another server could take place. I've been looking at the RedFone foneBRIDGE2 2e1 product here:

We tried them out early last year when we were looking at a large deployment and they gave us a lot of the redundancy that we wanted. However we did run into issues where calls seemed to get caught up in the system. It was as far as we could tell rather random. No consistency to it at all. Asterisk hung up the call but the telco side of the line didn't actually hang up. The channel was left

I'm looking for reports of recent experience with redfone fonebridge2 (with echo can) TDMoE gizmos. Anybody? Good? Bad? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080609/23aeddd6/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s

We have been using Red-fone foneBridge2 Quad T1/E1 for last few years. As these devices are not available anymore, we are looking for alternatives. Are there any similar devices available ? -- Regards, Tirveni Yadav www.udyansh.org What is this Universe ? From what it arises ? Into what does it go? In freedom it arises, In freedom it rests and into freedom it melts away. Upanishads.

This is concerning an Asterisk 1.4.18 server. We have approximately 70 DID numbers. Incoming calls are placed into the "incoming" context (by zapata.conf) and are routed based on the dialed number. I want to do some manipulation (CallerID name override) to all incoming calls before they are routed. I would prefer to avoid duplicating the necessary code in each DID extension stanza,

Ok Guys, I've done a tonne of hunting around on this problem, but can't find much help. I'm running: asterisk 1.4.19.1 libpri 1.4.3 and zaptel 1.4.9.2 which I believe has been modified by RedFone to add the ztd-ethmf module. My interface is a RedFone foneBridge2 4 Span; and I'm connecting to a BT E1 PRI / ISDN30e with 15 lines on span 1, and a legacy Panasonic PBX on span 4. Upon

I need to monitor the states of my T1/PRI Zap channels. Specifically, I need to be able to programmatically determine whether a channel is unused, carrying an inbound call, or carrying an outbound call. Using the manager interface, I can easily tell whether a Zap channel is used or not by looking at the results of: Action: Command Command: zap show channel <x> Or: Action:

I am having two problems, possibly related, while performing pre-deployment testing of a Samba/OpenLDAP PDC with data that was vampired from an NT4 PDC. The Samba server fails to become a local master browser, and password change attempts (from a Windows client) fail. I followed Samba-Guide/ntmigration.html (taking some liberties with various items of configuration), ending with step #19.

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

For high-availability Asterisk when using PRI Lines, you might also want to check out our product (FSV-4PFS). It's available at www.failsafevoip.com Bill Also to add to the last post does this device have hardware echo cancelation? if it does it could be a great replacement, if not may not be what I'd really want to use. Thanks, tom I've been looking at the RedFone foneBRIDGE2

Need to add some fxs and fxo ports behind a fonebridge2 box any recommendations a channel bank Thanks Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070829/1131cad4/attachment.htm

Does Icecast support bandwidth negotiation like Real's server? If so, how would one configure this (can't find it in the docs or list archives). If not, is there any interest in adding this capability? -- Kevin DeGraaf --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to

My apologies if this has been covered already. My search of the archives was unfruitful. OpenSSH seems to be lacking a certain capability present in ssh.com's client; namely, the ability to store remote hostkeys on a per-port basis. I have various machines that, due to iptables port-forwarding, appear to be running copies of (open)sshd on multiple ports. "Commercial" ssh stores

Hello, If anyone is using a TE121B card and it works reliably (i.e. no "HDLC Bad FCS" or similar errors), could you pass along the make, model, and basic configuration of your Asterisk server? We tried upgrading our old Dell PowerEdge server to a SuperMicro system, but that didn't help. I would like a solid recommendation before I suggest another purchase. Thanks. -- Kevin

Greetings, I am writing an AGI script that needs to check on the idle/busy status of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and Snoms thrown in for fun). Is it possible to call Asterisk functions (e.g. SIPPEER) from AGI scripts? Based on my Googling, I would guess in the negative. I have tried various permutations of Set() and Eval() without success. I have also

OpenSSH gurus: Apologies if this has been covered already (or is a genuine FAQ). I've searched both Google and MARC extensively on this issue, and have come up empty. I use OpenSSH 3.0.2p1 (openssl-0.9.6c) on a group of Linux (Slackware 8.0, kernel 2.4.13, glibc 2.2.3) machines that have this in /etc/hosts: 10.1.1.2 s1 s1.[domain].com ... 10.1.1.6 s5 s5.[domain].com This is in