similar to: Picking a channel (FXO port or SIP) for outb ound calls

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:

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:

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

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

> -----Original Message----- > From: Jared Smith [mailto:jsmith@drgutah.com] > Sent: Monday, 12 January, 2004 10:41 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Turning a profit (WAS: More words > for Allison) > > > On Mon, 2004-01-12 at 04:49, Alastair Maw wrote: > > Hmmm... I think John's turning a profit... :) > > That was my

> 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

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

These are the three cheap SIP phones that I've used. Grandstream BT10x $65/street Number only LCD Zultys ZIP 2 $100/retail No LCD Uniden UIP 200 $120/retail PoE, built-in switch -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."

Hi all, Let me be the first to wish everyone, especially the Digium team, an awesome year in 2004.. Later..

Just curious if any of the Asterisk installers are doing anything special to protect themselves from a possible lawsuit caused by 911 failure during a Asterisk/computer crash? I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. Anybody requiring customers to acknowledge and sign any kind of

It's the day before Christmas here in Sweden, actually the night before at this time... We celebrate Xmas on the 24th, so I'm about to log off and switch my Asterisk into "merry-christmas-mode" with the yet undocumented CLI command "frosty-mode on", a mode where the PBX will connect all incoming extensions to the "ho-ho-ho" sound file and then randomly pick a

Is there a was to reset asterisk by dialing? Eg *99 and have it execute the commands to stop now and restart? Also eg *56 to reload? The reason for the restart command is over the weekend it started acting weird, the voice got really choppy and aver i restarted asterisk it was fine. We are just in testing right now but it would be nice to know how to do this if it were to happen again and im

Help! I've been using mysql for cdr storage, I need to switch to MS SQL. I must be stupid or something but I cannot figure out how to setup the cdr_tds. I have FreeTDS configured properly, but my unixodbc is not working properly either... I'd be happy with either solution, but I'm in need of assistance. Luke Catranis

Hi, I'm a newbie to the list, but have been screwing around with Asterisk for the last 6 months or so (on a purely experimental basis so far). I'm not a linux expert by any stretch, (I'm a Mac OS X user), so I'm unsure where the line is drawn in terms of Linux issues or Asterisk issues. At present, I have to manually start Asterisk from the command line, but I'd like to

The answer is yes, I think, but I don't recall precisely how off the top of my head, and I'm walking out the door in a moment. The phone will hold more than a dozen distinct ring tones which you can create for yourself, and you can have asterisk direct it to use a ring tone independently of line appearance. The most direct way would be with the SIP Alert-Info field, but the phone itself

Hey, Thanks for the input Andrew. I did all you suggested but noticed that when I did the loopback test, the output *was not* there as you mentioned ("I'm set to pri_net, but the other side thinks it is pri_net!"). In fact, the same message as before kept repeating every second or so: >> Unnumbered frame: >> SAPI: 00 C/R: 0 EA: 0 >> TEI: 000 EA: 1

Hi all, Anyone has experience using * through 128 kbs (or bigger) satelite link? In particular I am interested to hear how many calls could be put through 128Kbs satelite link simultaneously? Ta SJ