similar to: Caller ID Info from Cisco router to Asterisk

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:

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

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

> -----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

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."

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

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

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 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

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

I am planning to buy Dell 2650 server with dual Xeon processors. And I would like to buy two TE410P cards for PCI with 3,3v. This is on Dell site about PCI slots for Dell 2650 server: 3 PCI-X (1x64-bit/133MHz, and 2x64-bit/100MHz) Does that mean I will be able to buy two TE410P cards ? Or I need to buy two TE405P cards ? Thanks for help. bartek

Because of space limitations and because of the location of the punch-down blocks, my * server is located on the shelf in a coat closet. Sadly, there is not enough space (or ventilation) for the monitor and keyboard. This will all change when we move to new quarters, but... Does anyone have experience running Linux/Asterisk without a monitor? What, if any, are the issues? TIA

Digium X100P / new cards are is available on ebay for $43. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309 <http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309 > Hope this helps to who want to play with X100P! Are these being sold by Digium ? I don't know ?? - SamW -------------- next part -------------- An HTML

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:

All, I have access to a Cisco AS5300 w/ 4 T-1's and a Cisco 3600 with no boards. I was wondering if it would be possible some how to Have one of these Ciscos in-between our sip phones and the asterisk server so that we could use G729 Codec. Sip Phones (7960's & ATA's) via G729 -->Cisco Gateway-->Asterisk via G711. Any ideas? Has anybody done such an implementation or know

Hello all, We are looking for some hardware requirements/recommendations to be able to handle a full DS3's worth of TDM -> VoIP traffic. The DS3 would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to convert those calls into G729 SIP VoIP calls to send to our asterisk box over ethernet. Since everything is going in/out of asterisk is 729, and no features

I have been compiling information on this configuration onto the Wiki: http://voip-info.org/wiki-Asterisk+cisco+FXO I can call out to the PSTN just fine, but inbound calls all appear in my [bogon-calls] context. Can anyone help me locate why? (Config files are on the Wiki) I have done a packet sniff & decoded using Ethereal-0.10.0, but this doesn't tell me a great deal - I just see