Re: [asterisk-users] Cisco remote reboot
I've fired a script from an AGI-BIN to accomplish that.
Try this one:
#!/usr/bin/perl
# mk 2004 feel free to distribute
# [EMAIL PROTECTED], _Vile
# perl script to reboot phones
# try telnetting to your phone, first.
#
use Net::Telnet ();
$phone_ip = shift;
# Your Cisco 79xx prompt
$prompt = "Enter Your Prompt Here";
# Your Password
$password = "xx";
# Reset Command
$command = "reset";
if ($phone_ip eq "all")
{
reboot("xxx.xx.x.xx",$password,$command,$prompt);
reboot("xxx.xx.x.xx",$password,$command,$prompt);
reboot("xxx.xx.x.xx",$password,$command,$prompt);
reboot("xxx.xx.x.xx",$password,$command,$prompt);
} elsif ($phone_ip eq "") {
print "Enter an IP or 'all' for All.";
} else {
reboot($phone_ip,$password,$command,$prompt);
}
exit;
sub reboot{
my ($ip,$password,$command,$prompt) = @_;
$t = new Net::Telnet;
$t->open($ip);
$t->waitfor('/Password :.*$/');
$t->print($password);
$t->waitfor('/'.$prompt.'>.*$/');
$t->print($command);
}
___
--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] Asterisk High-Capacity Stability
On May 14, 2007, at 11:27 PM, Atlanticnynex wrote: I'm curious what kind of configuration/features/modules you could recommend for my setup. Can you explain further what you mean by OpenSER to Asterisk? If you want to go Open Source, I think OpenSER is a good choice. You won't need to do any hacking to make it work..I'd suggest making 1 or 2 openser boxes to act as registrars for your user agents, and use the openser dispatcher module to point at one or more openser boxes that do LCR for calls that go directly out, and at one or more asterisk boxes for feature servers if you need them. Using Asterisk realtime and the database extensions for OpenSER you can share the user database between them and things should just work. Write your CDRs to a separate database (as to separate business data and call flow datajust in case someone does a complex CDR query you don't want your PDD to go through the roof) and come up with some kind of CDR remediataion for billing. ___ --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] Asterisk High-Capacity Stability
On May 14, 2007, at 1:29 PM, Zoa wrote: Several people do use it for handling > 50k minutes a day. (I'm one of them). Yes, you need to know what you are doing, and have a nice design, but it is possible.Our code is only slightly altered. (mainly for billing purposes). That's great if you're good enough/have the time to make that happen. But when I have issues and call/pay Digium and don't get timely or meaningful answers, it's doesn't make for a good business decision to continue using it for that purpose when I can toss in a Nextone or Sansay and have it "just work". All the time. No babysitting. Full professional and timely problem resolution from the vendor, etc, etc, etc. Don't even get me started on Digium not being able to get TC400Bs to properly negotiate g.723.1 5.3k when a client requests 6.3k first (thank god for Cantata). I guess it all comes down to whether you want things to just work and be able to have tier 1/2 support capable of actually doing anything meaningful, or if you want to have the engineering level people forced to do all the work. From my standpoint, the smart business decision is quite clear. But, as I said, Asterisk is still driving the feature servers, and works well for it. As mentioned by someone else previously in the thread, it makes a great endpoint. If you're having good success with it, that's fantastic. I would hope that you contribute back to the list how you set things up to make this a possibility. ___ --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] Asterisk High-Capacity Stability
On May 12, 2007, at 4:11 PM, Atlanticnynex wrote: 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 dimensioning and it seems theres some mixed feelings about Asterisk in high-capacity environments. I guess I'm looking for input as to whether Asterisk could handle roughly one DS3's worth of calls (672 calls) just doing the LCR (I've seen some pre-built LCR apps, looks like they all do on-the-fly MySQL queries- I think I'd write my own AGI that would use a cache). With my hardware, could Asterisk run stable for this amount of traffic? What stability issues does Asterisk have at this scale? Simply put, NO. I am on a project now where a client had an OpenSER box acting as an SBC and registrar passing traffic to several asterisk boxes which are doing LCR lookups on the fly as well as writing custom CDRs all through PHP AGI scripts to a Postgres DB. The Asterisk boxes do not scale, and randomly start swallowing calls or, more often, restart the process (safe_asterisk is handling this). There is some light IVR type usage for reporting account balances and the like. With anything more than 80 or 90 calls on the box, the IVR prompts start to break up. Ben through replacing hardware, more memory, different Asterisk builds, etc. I've had an open issue with Digium support on this for at least a couple of weeks, and the best advice so far was "try using the SVN build". That makes things better, but it's still not anywhere close to fixed.. It's absolutely incredible that Asterisk works at all for some of the situations its been put in - major kudos to the developers. But I don't think using it for what you're talking about is a long-term business strategy. When the highlight of the 1.6 release is bridging channels, you know high volume sip to sip usage in a carrier class call routing environment is NOT what development is focused on. And that's fine. If you use a wrench to do the job of a screwdriver, you shouldn't complain when you bust your knuckles That being said, I don't meant to trash Asterisk at all. It's a fantastic feature server, and a great PBX, both of which things I use it for very successfully. I just don't think it's ready to handle 50k plus minutes a day SIP to SIP with LCR and billing data, no matter what you do with it. I'm 100% positive there are people out there doing it successfully, but those are the exception, not the rule. And I doubt they are running unmodified code. ___ --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] Packet2Packet Bridging Questions
OK...that makes much more sense. So here's my follow-up question: what's the easiest way to check if I'm native bridging a call. I'm trying to offload as much RTP traffic as possible, and want to have a way to check quickly (there are well over 50 calls on each of these boxes at any given time). I've been going the ethereal route, which is great for debugging, but not so good for a quick look. Thanks again, Daryl ___ --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
[asterisk-users] Packet2Packet Bridging Questions
I'm just starting to upgrade some boxes from 1.2.x to 1.4.1 as well as trying to get some of the RTP traffic offloaded from the network. I think I'm misunderstanding what the console messages mean when it says "Packet2Packet Bridding SIP/blah to SIP/blah". I though that meant that it had successfully (re)INVITED and the media was no longer going through my Asterisk box, but ethereal says different. I'm not having much luck finding any information on this on the wiki or google. Can someone point me in the right direction? Thanks, Daryl ___ --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
