similar to: What is the best way to configure this?

Hi, I have a strange problem, when I try to connect to les.net from our local asterisk server through Procurve router I seems to be connecting on any port above 1024 and when I reload sip the port is changing too ... So I never get 5060? Any ideas on what is going on and how to resolve it? ? Sincerely, Robert Augustyn 519-997-3106 ext:802 www.linqone.com ? ? -------------- next part

Hi all, Occasionally, DIDs from different providers stop working for some reason. I would like to be able to monitor situations like that and react before any of my clients start going ballistic on me. Any ideas? Scripts you know of or wrote and willing to share? Any info?would be greatly appreciated. ? Robert ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, We have zabbix running and would love to be able to monitor our asterisk box with it. I believe that some sort of SNMP is build in 1.4+ correct? Where do I find more info or a how to on what is supported and how to use it? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Interesting read in Alec Saunders blog today. http://saunderslog.com/2007/03/01/mashable-telcos/ Thought it may interest some people on this list. As food for thought, why it is that ITSP's haven't come up with more 'interesting' voice applications? When asterisk first became available I thought it was the beginning of seeing really neat applications, think Verzion's

Hi all, We have put together a firmware for a range of inexpensive routers. It has been configured to provide optimum VoIP performance. We have internally tested it for number of months and it looks very good. You should be able to run it easily with 20+ phones on local network ( we still did not hit the upper limit ) assuming that you have bandwidth. Your VoIP will get prioritized over other

Hi, One of my clients, wants to use * box to run weekly meetings between remote locations over the internet. What would be the best configuration for this? We are talking about two conference rooms. I am referring to the actual hardware/software and bandwidth requirements for this to work well. I have run two software video phones and I had marginal results with it when displayed on large LCDs,

Hi, I am working on developing a professional Marketing Materials for my systems. I plan on using a very good(expensive) company to do that so splitting the costs with several people would be nice. Let me know if you are interested on taking part in it. robert -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am planning couple small business installations and wader what should I use for 2 to 6 lines a gateway or pci card? Any comments greatly appreciated on pros and cons and brands. Thanks, robert

hi, i have 2 tables (counties and towns). counties has_many towns and towns belong_to counties. now my question: i thought i would need to do is say @counties = Counties.find(:all). should that not return to me all counties in the counties table WITH all towns associated with each county? in my view i was getting error when doing this if(counties.has_towns?) saying undefined has_towns

Hi all, I'm quite new to R and have a very basic question regarding how one gets the standard error of the mean for factor levels under aov. I was able to get the factor level means using: summary(print(model.tables(rawfixtimedata.aov,"means"),digits=3)), where rawfixtimedata.aov is my aov model. It doesn't appear that there is an equivalent function to get the standard

Hi all, Is using a Citel gateway with Asterisk a good solution for reusing of the old Nortel digital phones? Would love to get some input from actual users. Any/all opinions welcome. robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070510/bc5fc18f/attachment.htm

Hello, Is it possible to set a variable in dialpan when the someone calls a particular DID number so that i can use that variable for calls coming to that number only. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/25402ade/attachment.htm

Hi, Some background: I would like to call my girlfriend over the internet. We are both behind a nat router and I want to avoid portmapping. I've heard that you can call someone behind a firewall (nat router) with the IAX protocol, but I'm not sure. The questions: Do I have to set up my own PBX asterisk server or are there any other (free) servers where I can register on and connect

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I have this [TRONCAL-SIP] exten=>225/91,1,Answer exten=>225/91,2,Echo exten=>225/91,3,Hangup exten=>225/92,1,Answer exten=>225/92,2,Playback(conf-invalid) exten=>225/92,3,Hangup When I make a call CLI> -- Recv IAM CIC=8 ANI=91 DNI=225 RNI= redirect=no/0 complete=1 Dont work If I add this rule exten=>225,1,Answer Works ok -------------- next part --------------

Hello, There is a problem that i can not figure out how to solve. I got users with 5 digit usernames for sip. Some users has a callerid for outside calls. I have such problems When a user activates (for ex) call forwarding, System creates that entry on database as CFIM/callerid not the username, So this rule works only if a call is made from outside to the callerid. Not the local calls made

Hi, I need to be able to unable and disable iax2 trunks from the cli? Is there a way to do it if so how? Sincerely, Robert Augustyn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081128/7ee24fa1/attachment.htm

Can anybody provide DIDs for Chatham, ON? Usage based preferred, but flat-rate is not an issue. ? ? Contact off list. ? Thanks for your time, ? ? Sincerely, Robert Augustyn ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100529/e8ac734e/attachment.htm

As seen in the following URL: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I also tested some time ago with an old release of Asterisk, RealTime Extensions didn't support the Ex-Girlfriend syntax. Is it already working in recent 1.4 or 1.2.15 releases? Is there any other way that I can use to do the same thing but only using contexts, for example? If yes, please

Ex-girlfriend logic not working in latest CVS? Incoming sip calls don't work. Anyone else seen this problem? Extension logic looks good: exten => 6153248305/_931NXXXXXXX,1,Queue(queue1); exten => 6153248305/_615NXXXXXXX,1,Queue(queue2); ;exten => 6153248305,1,Queue(queue3); show dialplan looks good: -- Added extension '6153248305' priority 1 (CID match