similar to: Backup Proxy & Automatic Failover

> Tilghman Lesher wrote: >> On Sunday 18 January 2004 12:01, Adthrawn wrote: >> >>> Spell out RAID - Redundant Array of Inexpensive Disks. Bingo. >> >> >> That's "Independent Disks". It's the independence of each spindle >> that >> is valued, not the cost proposition. If one spindle goes, it's not >> all >>

Hello all, I post this issue thinking too that could help other people on an asterisk deployment over distributed offices considering both quality, prices, devices and so. Well, i am working on a deployment of a telephony system based in asterisk. My company have a central office with seven remote offices connected all through a VPN. To reduce and evaluate costs i consider solutions like:

Thanks Alex, some great ideas. I think, however, I'm leaning towards Asterisk at this point- since I have quite a bit of experience there, and very little with SER. At this point, I'm wondering from a dimensioning standpoint, what kind of capacity my machine will have (Dual Core Xeon 2.4GHz 4GB RAM). As I said, I don't plan to do any transcoding. I read the voip-info page on

Hello, all. In section 4.3.10 of the Asterisk Handbook, there is an example of an LCR/Failover Trunking scenario. I've tried it, and it works, as long as I fail over from something else to ZAP, but I can't get it to "hunt" to the other context if the zapata channel (or group) is used first. Can anyone help? Here is my extensions.conf, and the error message I get.

Hi, I know it's not really the place, but if anybody in the UK (or US) is interested, I'm clearing out lots of new Cisco stock... 7970G's (colour LCD), 7960G's, 7940G's, 7920G's (wireless IP phone), 7935's (conference phone) and 3550-24-PWR switches. I also have boxes of 7914's, the single-7914 foot stand and double-7914 foot stand (these are required to

I am facing one way audio problem in sip trunking between asterisk and avaya. +-------------+ +----+ | avaya sip |-------| P1 | +-------------+ +----+ | | | +-------------+ | Asterisk | WAN

Hi List, I'm finding a solution to provide failover and load balancing features to my IVR system. Anyone suggest me what is the best solution please?. what the hardware I should use ?. I heard about RedFone, but someone on the mail list said that it is not good because TDMoE module in asterisk is not so stable and TDMoE is stale. And It seems that RedFone doesn't not support load

Hi, Does anybody know of a VoIP compatible doorbell or door intercom unit? I've contemplated buying a cheap SIP phone, ripping it apart, and putting it inside an IP66 sealed unit... It would need: - At least one speed-dial key, or some way to make every button dial the same extension number - PoE (power over ethernet), so I can power it off the central switch - cheap enough to rip apart

Hi, I'm after two very specific ringtones for the 79xx's... A dog barking, and a horse either galloping or neighing. I've tried making the sounds, but for some bizarre reason they're not working. I used to make quite a few ringtones for the 79xx's, but I seem to have forgotten how to do it! And to top things off, I can't even find the documentation on Cisco's site

Hi, I'm just considering buying two Telecoms grade Sun Netra's to run a lab-based VoIP solution. Not my immediate thoughts as a VoIP platform, but from what I've heard, they can run Linux, and run it well. Only thing is: The Wiki and the Whitepaper just state that Asterisk is for the x86 architecture, but has been compiled to run on PPC architectures. No mention of UltraSparc.

Hi. I'm proposing a wireless distribution system in infraestructure mode for a valley with no cables and distant houses, respect to Asterisk, I can get a company I work with to provide me with E1 and use LCR to get 4 different telcos based on prefix. I've been reading a very interesting paper about voip on wireless, i recommend it really

Hi, I have a growing voip business Im i looking a solution that can handle at least 3000-4000 concurrent calls with great performance. Also with a billing platform, reports, reseller platform, LCR, call routing,real time reports, SQL dababase access real time Load Reports. Any recommendation? Thanks Ignacio -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I've got myself all confused about the capabilities of *. I somehow convinced myself (because I see a lot emails flying around about IP phones) that Asterisk works as a PBX and trunking gateway, but does not do voice coding (i.e. TDM in, VoIP out). Does Asterisk work as a VoIP gateway that regular (non-IP, non-SIP) phones can connect to and establish voice connections to other

Hi, Does anyone have any advice in how to implement PSTN failover should an internet connection for IAX trunking go down? to route outbound to analog lines Can this be written into the dialplan using a GotoIf statement by testing the whether the internet connection is up, or from a IAX/SIP response? Thanks Robert McNaught -------------- next part -------------- An HTML attachment was

How to set-up LCR ? a. which companies can be used with LCR? b. how to set-up & maintain LCR? c. multiple connection to one gateway? Example: +886223456789 could be reachable via a. ENUM free b. Dundi free c. Voipstunt free d. Voipbuster free e. Nufone $ f. Voipstunt $ g. others with 4 concurrent connections $$ h. others with 3 concurrent connections $$ I am looking

Hello. I would like to know if somebody did a wireles voip with Asterisk PBX. I think to deploy a wireless for about 500 potential customers, it's a 3 km radius maximum coverage with houses without phone lines, I work for public places telephony small enterprises ( a common bussines in Spain) so I can get good rates from 4 telcos and do LCR at my asterisk PBX. Is anybody did this before

Hi I have come across a problem, with my tdp410 and soekris board (basically pc on a chip amd geode cpu). I am using the box as a firewall/asterisk box. The problem occurs when I drop ppp and I get dead loop dectiotn going, I seem to lose interrupts and get lots of messages in syslog from wctdm24xx saying missed interrupt increasing latency its out lined here

Pardon my ignorance, but I was hoping someone could clear up something for me. - For a few POTS lines, digium has a single port card for that, or a T1 card to a channel bank. - For 10 or more lines, digium has a T1 or E1 card for that too based on PRI channels - For 100's to 1000's of lines, I suspect a soft-switch is in order??? A traditional phone company will sell: - POTS lines for

Is there any possibility of DAHDI supporting Automatic gain control on TDM ports? I'm having issues at a couple of offices where calls made to local numbers are fine but a when a calls from or goes to a large percentage of long-distance or 1-800 numbers the person at the remote end cannot hear the person in my office. Boosting the gains in zapata.conf (I'm still using 1.4.21) to 8

hello there, I am facing an issue with misd/misdnuser/lcr in the system I am running debian 7 and I managed to install from git misdn/misdnuser but in lcr I am getting: chan_lcr.c: In function 'load_module': chan_lcr.c:3520:24: warning: assignment makes pointer from integer without a cast [enabled by default] make[2]: *** [chan_lcr.po] Error 1 make[2]: Leaving directory