--On Monday, January 17, 2005 22:20 -0800 [EMAIL PROTECTED] wrote:
<...>
The basic arrangement would be: Telco <-> T1 <-> Asterisk <-> T1 <-> Channel Bank <-> POTS <-> regular phone
Here are my questions thus far:
- Firstly, which channel bank would best suit me? I only need FXS, but I'll need Caller ID, Visual Message waiting, and a decent ring generator. I haven't found much info on which FXS channel banks output FSK, or what loop length they can tolerate. ASDI would be nice too, but not yet a requirement.
- Can someone point me in the direction of sample configs for DID applications?
Check the asterisk wiki http://www.voip-info.org/ but basically incoming lines are tied to a context, with a T1 the phone company usually delivers a 4 digit DID so you'll have an incoming context with a bunch of 4 digit DIDs *or* you can do something more complex and do a lookup to some sort of database, which for you might be ideal since you want things more dynamic. extension changes on the fly are definitely doable, reload reconfigures the extensions list without disturbing in-flight calls.
- What size server will I need? Assume for now a pair of quad-T1 cards, 2 T1s incoming, and 5 channel banks. Shouldn't require much horsepower since it's T1 <-> T1 switching
Without transcoding you'll be able to get by with almost any mid to high end P4 Xeon (not celeron) system. Go for the 1MB cache as the Pentiums really suffer badly without the extra cache in my experience, and the price jump is not that much. go to a 2.8ghz and 1mb or a 3.0 ghz and 1mb instead of a higher clock rate.
You might want to consider one of the other options rather than the Digium hardware...someone had a link to some hardware that had onboard DSPs and was very similar to the Digium hardware otherwise. This would certainly make a solution much more scaleable. That said I think you'll be fine with 2xQuad PRIs in the same box, and if you're worried though it'd be simple to split the Quad PRIs to two different boxes, then have them handoff to eachother only when necessary.
I'm posting probably later tonight a few bits of framework to asterisk-users here that I used to make a dialplan based 'agent' system since chan_agent was horribly and uselessly broken for me. I'm still running 1.0/1.0.1 so I dont' know what's been fixed in 1.0.3 or 1.0.4 -- nor will i find out anytime quite soon, everything's in production now and i can't afford to experiment for lack of time at the moment.
- Can I get my NMS AG-T1/E1 card working with Asterisk for tinkering purposes?
Uhm, probably not, but someone else can better answer this.
- What kind of uptime am I going to expect on Asterisk? Am I going to have to reboot the server every 2 weeks? Can I hope for carrier class service?
I'm having problems with 1.0 and 1.0.1 on the analog TDM ports after about a week to two weeks, they stop ringing. However I'm pretty sure there are patches in that fix this as of 1.0.2. I haven't had time to investigate it nor upgrade but scheduling a late night reboot after checking for live channels has me fixed for the time being. The T1 card has been flawless since we put it in.
- Is it possible to change extensions on the fly? For example, this week 555- 1111 rings in unit #10. Next week theres a different tenant, so i want to make 555-2222 ring in unit #10, and send 555-1111 to voicemail. Some kind of GUI to accomplish this would also be nice.
See my above note....I am not aware of anything exists out of the box, but there are certainly those whose services could be contracted to produce this.
Thanks in advance for your responses
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