search for: lawetz

Hi, I wish to use Asterisk as a SIP server. How do I use Asterisk in a redundent network? So, if one Asterisk server fails, how does failover work? James

I'm looking for a way to transmit a user specific variable to my dialplan If we use the example of the hair color, I was thinking of having something like: [bob] context=users host=dynamic secret=password type=friend username=bob hair=brown [lary] context=users host=dynamic secret=password type=friend username=lary hair=black And be able to access from the dialplan: [users] Exten =>

I've got a friend who's spending 6 months on the other side of the world. So before he left I configured him a softphone on his laptop to connect to my asterisk so he can call home free of charge. Unfortunately, he just found out he has horrible internet connection. Bandwith and latency is ok, the problem is the stop almost all connections. He has to connect to a proxy server for his web

...the right combination of words :-( Thanks -- \\\|/// \\ - - // ( @ @ ) ---oOOo-(_)-oOOo------------------------------- A compliment is something like a kiss through a veil. -- Victor Hugo --------------Oooo----------------------------- oooO ( ) Benjamin Benthos Lawetz ( ) ) / mailto:benthos@step.polymtl.ca \ ( (_/ ICQ# 4269530 \_)

...Thanks for any help Benjamin -- \\\|/// \\ - - // ( @ @ ) ---oOOo-(_)-oOOo------------------------------- There are times when truth is stranger than fiction and lunch time is one of them. --------------Oooo----------------------------- oooO ( ) Benjamin Benthos Lawetz ( ) ) / mailto:benthos@step.polymtl.ca \ ( (_/ ICQ# 4269530 \_)

...' -- \\\|/// \\ - - // ( @ @ ) ---oOOo-(_)-oOOo------------------------------- Most people can't understand how others can blow their noses differently than they do. -- Turgenev --------------Oooo----------------------------- oooO ( ) Benjamin Benthos Lawetz ( ) ) / mailto:benthos@step.polymtl.ca \ ( (_/ ICQ# 4269530 \_)

I'm having a little problem. I have a dial-plan with a lot of SetVar's and loops, and under certain circumstances (reproducible) it makes asterisk crash. Wanting to debug this, I compiled using "make valgrind". But doing so, I eliminated the crashes and the dial-plan works perfectly. Now from what I understand, valgrind removes compiler optimisation to ease debugging. What kind

Hello everyone, I have a couple of bugs I'm trying to debug compiling asterisk with valgrind. But of course when compiled like that the bugs don't occur. What are the exact effects of Valgrind? Would there be a hit on performance running asterisk compiled with valgrind ? Thanks for you insight. -- Benjamin

Hello all, I've been playing around with early audio, and I'm able to get some things working We have PSTN calls coming in to asterisk in SIP from a Cisco AS5300. If I do the following: Exten => i,1,Playback(ss-noservice,noanswer) Exten => i,2,Congestion(15) Exten => i,3,Hangup() The PSTN caller does not get an answered call (doesn't get billed) but hears the ss-noservice

Is there anyway to prevent an extension from dialing certain codes. ie I want to prevent extension 203 from dialing number which start with 00 087 086 etc Sean -- +----------------------------------------------------+ |VOIP: FreeWorldDial 689482 VOIPBuster thecivvie | |GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc | +----------------------------------------------------+ --------------

We're getting our feet more and more wet with VOIP at work. We want to experiment with a good wireless (as in WiFi) phone. What would be a good phone to impress my boss with? I'm personally drooling over the UTStarcom F3000, but compatibility and shipping ETA info is a bit sketchy. Phil

All, I made some progress, but it seems the further I go with clustering the harder things get. Hmmm, I guess if it were easy, it would be documented...... Anyhow, I have 1 * server as the DUNDi peering master with a ttl=1. The only function of this server is to lookup where other sip peers are registered and forward that info on to the requesting * server. I have 4 * servers accepting

Is asterisk able to forward it's ALERT_INFO data to another asterisk server ? My situation should look like the following: Call comes into asterisk1 in SIP. Asterisk1 sets the ALERT_INFO=Bellcore-r2, Asterisk1 dials Asterisk2 (SIP), Asterisk2 dials our SIP device which should ring with the Bellcore-r2 Any way to pass the ALERT_INFO through to the SIP device? Thanks -- Benjamin

Was wondering if anyone else was getting these kinds of messages with Sipuras (SPA-1001 and SPA-2100): WARNING[26867]: Forbidden - wrong password on authentication for INVITE to '<sip:5555@192.168.0.100>;tag=as25fb4c1a' It's a warning, so I'm guessing it's not more critical than that. But I'm curious about the error, what can cause a wrong password, but the previous

I'm running multiple asterisk servers and need to use the CheckGroup function (and other group functions) across multiple servers Ex: - there are 5 channels in group test on server 1 - there are 8 channels in group test on server 2 I would need a checkgroup to return me 13. Any way to currently do this ? What would be the best way to implement this if not ? Store group setting in shared

Hi, I've been having a little problem with my asterisk servers, I have 4 identical asterisk servers setup (same hardware, same OS, same config). Once in a while (once or twice a day) one of the server crashes on the cron job reload. But I realized this only happens on 3 of the 4 servers. Tried to spot the difference between that one server that wasn't crashing. The difference I found was