[asterisk-users] SIP vs Analog lines
Never having actually rolled an Asterisk (Trixbox in my case) system into production. I was wondering if in most peoples opinion if given the choice would rather have a straight VOIP/SIP system or would rather have a system with normal POTS/analog types lines and something like a digium card? As far as reliability etc. Thoughts? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo and static on PRI with errors
What do you do if you find things sharing interrupts (IRQ 11) in my case with my X100P card. I believe there is some sort of internal audio card in my cheap slow PC. Alejandro Kauffmann wrote: Tom O'Connor wrote: On Tue, Jun 30, 2009 at 3:12 PM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com mailto:tilgh...@mail.jeffandtilghman.com wrote: On Tuesday 30 June 2009 08:24:29 Tom O'Connor wrote: I'm currently pointing fingers at either the hardware (someone on #asterisk said it could be a cruddy chipset, but it's an HP Server.. so should be kosher.. ), I Is it an HP server from the HP server line, or is it an HP server from the old Compaq line? Don't assume that because of the HP name, it's actually reliable with 3rd party hardware. It's a HP DL145 G2. more than that, i can't say. -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org mailto:t...@twinhelix.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The card is TE110P compatible and as such probably suffers from the same interrupt sharing problem. The ...HDLC Bad FCS.. messages tend to be related to interrupt sharing. What does lspci -vb show? Anything sharing interrupts with the card? Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- John F. Ervin *Central Florida TeleSource, LLC.** *4270 Aloma Ave #124-69C Winter Park, FL 32792 (W) 407-679-6238 (F) 866-566-1282 (F) 321-445-0781 jer...@jervin.com mailto:jer...@jervin.com http://jervin.com/cft ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Building a System.
So, people have recommended building a system from scratch, start with a CentOS base and installing asterisk and all of the other utilities. I've only used Trixbox for my business system. I'm wondering what surprises I'd run into. Right now, I know I'd need the OS, Asterisk, something like FreePBX, I have a x100p card so I'd need Zaptel, does that come with asterisk? Fax support, seems to work with Trixbox, but I've heard that it needs to be loaded. Voicemail etc.? I mean, I don't know exactly what you'd need because almost everything I need comes with the Trixbox build. Are there (??) instructions for people who are experienced at the Trixbox level but wish to move on? -- John F. Ervin *Central Florida TeleSource, LLC.** *4270 Aloma Ave #124-69C Winter Park, FL 32792 (W) 407-679-6238 (F) 866-566-1282 (F) 321-445-0781 jer...@jervin.com mailto:jer...@jervin.com http://jervin.com/cft ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cheap CHEAP ata
Have you checked ebay? David fire wrote: hi i need many cheaps atas or some very cheap way to connect analogs phones to asterisk what do you recomend? i searches and only find solutions like 40 U$D (in the states, here in argentina is like 80 U$D) per phone any links or something? thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Origination and Termination
I like LES.NET. Tom wrote: I'm looking for sip origination and termination companies. Anyone know of reliable ones? I am interested in wholesalers also. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialling multiple extensions in an internal context
Can't you handle that by defining an outbound route? set it to hit a
trunk or set of trunks when the correct dial pattern is detected?
Matthew Pounsett wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi there. I've done some googling around to try and find an example
of what I'm trying to do, but it's one of those things that just seems
hard to find the right terms to search for. If there's some
documentation out there on this, I'd appreciate being pointed in the
right direction. If not, then if someone has some ideas I'd
appreciate that too. :)
What I'm trying to do is set up some 'special' extensions in my
internal context to change variables, or change something else in the
session before dialling. To be clearer, here's an example. Say I've
got this rather simple dial plan:
[globals]
TRUNK1=Zap/1
TRUNK2=Zap/2
TRUNK=${TRUNK1}
[internal]
_NXX,1,Dial(${TRUNK}/${EXTEN})
I'd like to add an extension which I can dial before placing the
actual call to change which trunk I'm using, like so:
*55,1,SetVar(TRUNK=${TRUNK2})
The problem is that once that's done, asterisk stops looking for me to
dial an extension, and I'm trying to figure out how to get back to the
top of the context and have asterisk wait for a new extension to
dial. I've dug through the Asterisk O'Reilly book, and googled around
some, but haven't come up with the answer.
Thoughts anyone?
Thanks,
Matt Pounsett
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