similar to: Turning a profit (WAS: More words for Allis on)

Greetings, Back in September someone asked about documentation for the new SLA feature in 1.4, however they received no replies. I thought I might ask the same question now in December. Apart from sla.conf.sample and a few comments in app_meetme.c I have been unable to find useful documentation. Is anyone using this feature right now? Is there a helpful source for information this highly

Thanks to everyone for your quick responses to this question. I'm very excited about the Asterisk project, and the growing community seems to be very active these days. Hopefully when the time comes for our county's transition to VoIP we may be able to go for an Asterisk-based solution. -- Tony Kava Network Administrator Pottawattamie County, Iowa -----Original Message----- From:

Greetings: I did some quick searching of my history of this list, and I tried a quick Google search as well, but perhaps someone on the list can quickly answer this question. I have a very nicely working Asterisk system at home with two Digium X100P FXO cards. When my SIP phones want to dial-out I have them setup to grab the first analog card (Zap/1) with the following extensions.conf segment:

Dear Asterisk Gurus: Our county is finally ready to begin implementing IP telephony. We intend to use a Cisco router as our PSTN gateway and Asterisk as our soft switch. The plan is to use SIP between the Cisco router and Asterisk. We will have a single PRI T1 connected to the Cisco router for PSTN access. My question is this: Are Cisco routers able to pass caller ID information (from PRI

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Sent: Tuesday, 25 November, 2003 08:56 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for outbound calls > > > > Yep, we use it for international calling. Works great: > > exten =>

I realize this is a shot in the dark, but I'm trying to find a VoIP provider that offers 402 or 712 area code DID numbers. I'm almost completely convinced that no one offers these area codes (eastern Nebraska, western Iowa), however considering the wide audience of this mailing list I thought this would be a good place to ask. I would prefer a provider that allows for Asterisk use, but I

> From: Philipp von Klitzing > Sent: Tuesday, 02 December, 2003 10:50 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] How to restart * thru phone "when convenient" > > You could use "at" to issue the command at a deferred time. > Yes, sure, but this ain't that nice "asterisk only". :-> You should be able to place

> the default should not be to tell people to run CVS code, > that should only be for people interested in hacking on > the code and trying out bleeding-edge features. I second this motion. While I am not a developer I do notice that most projects tend to take this approach. The CVS is generally for those who want to experiment with the 'bleeding edge', and regular releases of

> I have placed "ignorepat => 9" in just about every context I > can think of in my extensions.conf, but yet when I dial 9 > from my sip devices, the dialtone is broken. I even tried a > nearly untouched version of samples, and it stil doesn't > work. Is there something somewhere else that needs to be set > to make this work properly, like may in the sip

Hi, I have * set up as a PSTN->VoIP gateway (with an E1 with multiple numbers pointing to it). I'd really like to be able to dial out to a SIP server like so: exten => _X.,1,Dial(SIP/${DNID}@hostname) I.e. the remote SIP server receives a SIP INVITE with a "To:" header containing the dialed number (e.g. 02085555555@computer.company.com). This is equivalent to having a

Hello, I think I speak for many people here when I say we'd love to see the specifics of how you have your Asterisk network set up (phones/server hardware/Asterisk setup/...). MATT--- -----Original Message----- From: Jared Smith [mailto:jsmith@drgutah.com] Sent: Thursday, January 08, 2004 11:50 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Asterisk success stories

Hi. I'm trying to do some custom call logging, and I want to call an AGI script from a hangup handler to log call durations and things. Although the script executes, it isn't retrieving variables from the AGI interface. Looking closer, I realised the variables are actually getting unset before the h extension is reached. [foo] s,1,SetVar,foo=bar s,2,Play(audio/a-long-prompt)

Thanks we just figure it out a bit ago. It's amazing how simple some things are when you just ask - and then realized that you were making it too hard to begin with!! :-) Lane Hoskins, MCP Network Engineer 540.767.7626 -----Original Message----- From: Jared Smith [mailto:jsmith@drgutah.com] Sent: Tuesday, January 13, 2004 10:59 AM To: asterisk-users@lists.digium.com Subject: Re:

We've changed E1 providers and I'm trying to reconfigure an E400P to make it work with the new lines. They're supposedly "standard" EuroISDN lines (in the UK). I'm initially just trying to get a single line up. I have the following in /etc/zaptel.conf: span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone=uk defaultzone=uk The LED on the back

For anybody who didn't know there is an article on asterisk in February's Linux Journal. AJ

Hi people. The latest version of my Ethereal plugin for IAX2 is now available here: - http://almaw.com/ethereal-iax2-plugin-0.3.zip A screenshot showing what you're missing is here: - http://almaw.com/ethereal.png The new version adds the following features/bugfixes: - Decomposes the CODEC fields for supported CODECs, complete with nice English descriptions. This gives you a

As mentioned on the devel list earlier today, I'm interested in writing an IAX2 plugin for Ethereal to make debugging IAX protocol implementation and simultaneous calls on normal networks easier. Anyway, I started work on it this evening, so it's not complete yet, but it's starting to look quite sensible: - http://raq626.uk2net.com/~al/ethereal.png A couple of people have

I can't seem to hide and/or set my caller ID from *. I'm using a quite recent (three weeks or so) CVS with an E400P card. I have pridialplan=unknown in zapata.conf and I'm based in the UK. The relevant bit of pri debug looks like this (reformatted to fit 80 char width): > Calling Number (len= 4) [ Ext: 0 > TON: Unknown Number Type (0) >

Interesting problem: An organization has departments. Each department has a single phone. Each department has multiple people. Each person within the organization has a direct dial incoming number. It's easy to set * up so that multiple DDIs get mapped to the same extension. What I'm wondering is if there's any way, with reasonably priced hardware, to notify the person who's

On 03/11/03 00:25, Mark Spencer wrote: > As a side note, I strongly would like to see someone implement a > client using libiax2 which implements IAX2 instead of the (now > obsolescent) IAX version 1. I'm implementing a Java-based IVR server (and yes, I know Asterisk does IVR, and no, it's not flexible enough to do what I want and no, it doesn't integrate well with the Java