similar to: asterisk+heartbeat

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

So here is my setup. Hardware: Intel P3 1.2 Ghz 1 GB RAM 36 GB Drives Mirrored Software: CentOS 5 2.6.18 Kernel Asterisk 1.4.14 Zaptel 1.4.7 (redfone) LIbpri 1.4.2 I'm using TDMoE with my PRI using a product called fonebridge from a company called redfone. They require that I use their own build of zaptel and I am trying to figure out if the problem is with them or something else. The

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

Has anyone on the list used a Soekris engineering PC as a TDM - Ethernet bridge? For example something like a net4801 with a TE110p in it and then using TDMoE to get it into a bigger server where the call processing proper will occur. Anyone know if it might handle a quadspan card ok? (no transcoding, just pure PRI to TDMoE bridging). Craig

Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. Thank you.

Hi all I am looking at implementing asterisk at a company with two ISDN bricks (60 lines). I know that the VoIP will absorb at least on brick worth of lines but that still leaves me with a need for 30 ISDN lines. As far as I can tell most of the Digicom cards have 4 FXS ports and I've read on this list that at most two could coincide in a box simultaneously without causing an interupt

Looking for a mentor... Having some issues with Asterisk 1.4.22 install. I am new to both Linux and Asterisk, however have 20+ years programming experience. First off I hate asking questions I could answer myself. I have and am reading The Asterisk manual, 2nd edition. I have successfully installed CentOS 5.2 and used yum to get a C compiler, current speed bump is with ./configure bash:

It is amazing to me at this point that there is not an official Digium list of supported servers (including 1u models!). Clearly the number 1 issue with the Digium PRI cards is the server that they are used in. The new cards even go as far as listing server that DO NOT work on the Digium site! The wiki references are old and do not have any testing parameters. C'mon guys! Certify a

First, Something seems to be wrong with the list. I'm not the only person who has expressed seeing their messages either arrive late, or not at all. With that out of the way.. Is anyone aware of any type of failover device for PRI on asterisk? I've found the ISDNGuard, however it is currently not made in the U.S., nor does it run on U.S. power. Is anyone aware of a device that will

Hi, Is it possible to use realtime for Music On Hold? Is it also possible to store the music/audio files on the database, same way a voicemail can be stored on the database? Thank You Regards, Nhadie

Hi, Anyone knows a DID provider that can do both outbound and inbound? Regards Nhadie

Does any have experience with E1 telephony support plus asterisk in costa rica ? Regards, Luis Morales -- --------------------------------------------------------------------------------- Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 --------------------------------------------------------------------------------- "Empieza por hacer lo necesario, luego lo que es posible... y

Hi, Is it possible to detect where the call came from and route it out to different sip trunks. e.g. i have user 100300 when that user calls outbound i will make him use of [sip-trunk-100] another user, 101300 when that users calls outbound i will make him use of [sip-trunk-101] actually the 100 and 101 at the beginning of the username is the accountcode i used for cdr. hope my question

Hi all, I am new to Hearbeat so please be kind :) I also posted this on Linux-HA lists with no responses so I posted it here. I have successfully configure two machines to use heartbeat to cluster httpd. The two nodes are called etk-1 and etk-2. I am trying to configure another two machines to act as a separate cluster (on the same IP subnet). These two nodes are called radu-1 and radu-2.

Hi all. This is probably more of a HA list, or possibly even linux-practices, question but all hosts concerned are running CentOS and I reckon some of you guys might have some good suggestions. Feel free to tell me to piss off. :) I'm building a new CentOS, DRBD, Heartbeat and NFS HA cluster. We already have boxes running similar setups on FC2/3 running NFS v2/3 with the Ultra Monkey

Asterisk is an excellent PBX system, and makes a very good endpoint in the SIP chain for all sorts of things -- IVR systems, voicemail applications, automated messages, etc. It has an extremely well-written CDR engine, so many people mesh it with billing applications to produce accurate accounting information. It also is fully aware of the media stream, which means it's capable of cutting

Hi, Is it possible to assign a plus sign on the callerid(num) ? currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing "bs523450017" instead of +6523450017. i tried putting it inside double quotes CALLERID(num)="+6523450017" telco says the same thing. is this possible? thank you Regards, nhadie

hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start

I hope someone can help me out with this issue. It has been dogging me for months and I can't seem to get it to go away. I have a Rhino Ceros box running Asterisk 1.4.21.2 connected via eth0 (nVidia MCP61 Ethernet) to a RedFone FoneBRIDGE2 dual-port with EC. The FB is the latest hardware rev and the latest firmware. I'm running the latest fonulator version and I'm running Zap-1.4.11