Re: [asterisk-users] WAS: 64 analog phones NOW: Selection criteri a and recipie for a good Asterisk install [long]
Colin Anderson wrote: I concur with your approach, but Tier 1 means as little here as it does when evaluating Internet backbone carriers. could you expand on what evaluation criteria you use? I'm going to be pre-speccing some stuff myself this month... Sorry I should have been more clear. A good Asterisk install needs a holistic approach to use a hippy dippy phrase. A Tier 1 server, which is a midrange to high end name brand server from the Big 3 (Dell, HP/Compaq, IBM, am I missing someone?) is usually highly optimized for bus bandwidth although that design was intended for a different use - usually massive disk I/O. As well, a Tier 1 server will have two seperate, independent PCI buses and this to me is a critical feature - it allows you to completely separate your TDM traffic from network, disk I/O etc. On my big production Netfinity, nothing a good opteron motherboard from tyan can't do (something like http://tyan.com/products/html/thunderk8we.html ) 5. Tuning of Asterisk box itself - this cannot be under emphasized. This is a very important step and tuning methodologies vary according to distro, skill of the admin, and particular circumstances. I've learned *way* more than I ever wanted to about processor affinity sinc I started using Asterisk. I'll be interested in more pointers on that one 6. Termination of PSTN. Basically I would never do an Asterisk install where I was forced to do something stupid like aggregate a dozen Centrex lines or some mickey mouse deal with FXO ATA's or whatever except for a hobby or prototype install. PRI, BRI, IAX or SIP, don't mess around with anything else. sometimes you don't really have a choice. some providers don't know what PRI is ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: max number of devices in hint
Lacy Moore - Aspendora wrote: I have one extension that rings in many places. It has just come to my attention that I can only monitor 4 devices within a hint. Pretty sure I know what the problem is now. It's not limited by devices, its limited by the length. This is when it would be nice to know C. I'm assuming the variables are declared and also declared as a certain length. I need to increase this length for whatever variable holds the hint information. Any hints? :-) I don't even know where to start looking. I noticed most strings in asterisk were of fixed length... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] señalizacion te110p, signaling te110p
Melcon Moraes wrote: What a confused message, isn't it? As far as I could understand, if you're getting a RJ45 for conection, you won't need any kind of adaptor. For coaxial cable, you'll need a balun. That's all layer 1 talk - physic layer Yes, you need to know a lot more about your pbx to proceed with the connection to your * box(TE110P). hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan, bueno como habia dicho quiero conectar una pbx a una te110p, la pbx me ofrece señalizacion r2 europea en cable rj45 o coaxial. ese tipo de señalización me sirve para la tarjeta te110p, ademas, alguno de esos dos tipos de conexiones me sirven o tengo que comprar algun adaptador. vi algo que tenia que usar un balum, es necesario para cualquiera de las dos conexiones?. cual tipo de conexioon me recomiendan mas? necesito saber algo mas sobre la pbx para configurar en la te110p? then he'll have issues with that thing using R2 ;D ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I stop lost DNS from killing Asterisk?
Rich Adamson wrote: Eric ManxPower Wieling wrote: Use IP addresses instead of hostnames in your Asterisk config. It sucks, but that is the only way I know of. Eric Bishop wrote: When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions? Using IP addresses only does not fix the problem as the asterisk system does not know who he is. Need to define him in /etc/hosts as well, then it works just fine. you can also install a non-crashing DNS server ;D ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] University dumps CISCO VoIP for Asterisk
Andrew Kohlsmith wrote: On Wednesday 20 September 2006 21:40, Douglas Garstang wrote: We stuck OpenSER in between the phones and Asterisk, and pointed our phones towards the OpenSER boxes for SIP registrations and subscriptions. When OpenSER received a REGISTER or SUBSCRIBE message, it would use the send() command to forward the messages onto each Asterisk server. By doing that, ALL of our Asterisk servers had a copy of all sip registrations and subscriptions. It seemed to work pretty well, but for unrelated reasons, we dropped that approach. Which approach do you use now? what were the unrelated reasons ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Dual core
Matt Florell wrote: For the Asterisk installation, no. For Linux, yes. I built a custom SMP kernel, which depending on your Linux distribution may or may not be necessary for you. what specific things have you done, that isn't in the base kernel ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Format_MP3, Streaming, File Formats, MOH
Asterisk [Submusic] wrote: musiconhold.conf [shoutcast] mode=custom application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 http://stream128.submusic.ch:8004/ ; The '/' in the stream URL is important ! I tried this. however it doesn't work. apparently, asterisk doesn't read from the mpg123 when no one is listening to MOH, and stuff appear to be stacking inside a pipe of some sort. when the next caller gets the MOH, he gets the music from 5 minutes ago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is there anyone working with 5ESS?
Luiz Miguel wrote: I am a new member and I got this error message: rmv aiu stoped data base error this sounds like a rather big problem tks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is there anyone working with 5ESS?
Luiz Miguel wrote: I am a new member and I got this error message: rmv aiu stoped data base error google for it... it found a few interesting references, among which: http://www.textfiles.com/magazines/TOT/tot-o6.txt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I need to record 30 conversations and have other 30 with music on hold, all at the same time, can a PIV handle it?
MF wrote: Hi, I have a 2 E1 system with 32 zap FXS extensions (all Zaptel, with TDM2400), on a PIV, 3GHz, 1GB, Well my question is wether I'll be able to use it for peak demand moment, that is having all 60 channels busy 30 talking to agents on the FXS, while recording their conversation at the same time, and the other 30 with music on hold while wait. This is all based on the Queue application of *. Does any one thinks this system WONT be able to handle it? Am I crazy of even trying it? the intel processors at great at number crunching, but suck ass at I/O I'd get an amd64 box instead ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
