similar to: Need help understanding SIP phones

>From: Jeff Cohen <jeffc at jolt-lang.org> >Date: Wed, 15 Sep 2004 10:35:36 -0700 > >What's a "compiling mesh?" What I meant, was that there are some implicit defines in mingw (like __GCC) and vcX (like _MVC) but possibly also other unsupported? internal structures. As I stated earlier mingw should be win32 api compliant, but not for complicating matters. But

FYI, Sorry for crossposting to the users and commercial list but this is really a HUGE development and deserves a large Asterisk audience. Digium = Shrewd business moves as I predicted when they purchased Sokol, it's like a game of chess! This is your 1-day reminder for the event: Event: 3Com? VCX IP: The future of communications ? taking businesses where they need to go. Date: Friday,

when check java process lock statistics, plockstat failed, please see below: # prstat -mLp 21162 PID USERNAME USR SYS TRP TFL DFL LCK SLP LAT VCX ICX SCL SIG PROCESS/LWPID 21162 7677 0.9 0.1 0.0 0.0 0.0 99 0.0 0.3 83 89 215 0 java/81 21162 7677 0.3 0.1 0.0 0.0 0.0 0.0 99 0.2 106 33 305 0 java/35 21162 7677 0.1 0.0 0.0 0.0 0.0 100 0.0 0.1 79 6 85 0 java/59

Hi, How do you folks handle really large queues (350+ simultaneous callers) in your Asterisk PBXes? We're going to be bringing in around 16 PRIs' worth of inbound callers, doing skills-based routing, and queuing them up for approximately 200 agents. What's the best way to handle all of these callers? We want to record the calls and we'll probably use the ramdisk method that has

I'm coming from the scipy community and have been using R on and for the past week or so. I'm still feeling out the language structure, but so far so good. I apologize in advance if I pose any obvious questions, due to my current lack of diction when searching for my issue, or recognizing it if I did see it. Question 1, plots: I have a data frame with 4 type factor columns, also in the

Hi- I'm looking for an appropriate system board to power a system with two (2) Digium TE410P cards. Since these cards require the 3.3 volt PCI, I'm considering vendors like Tyan for the motherboard. Can anyone please tell me their experiences with the Tyan i7501 series (Xeon-basd), or recommend an alternate motherboard? Thanks Scott Scott M. Stingel Emerging Voice Technology Inc.

Hi, Has onyone heard of successful deployment of 3Com ToIP over IBM iSeries system (formely AS/400) ? A prospective customer seems to looking for this but, in my whole life, I've never of a such setup. Does it work ? regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Does anyone know how to dial toll-free (800) numbers through FWD or Siphone? Using the configuration below, I can dial out to SIPphone.com users by simply dialing their number (1747XXXXXXX) and can dial out to FWD users by dialing 1383<FWD#> However, when I dial 18005551212 through SIPphone, or through FWD (depending upon which line is selected in "; 800 Toll Free Numbers"

Good day all, I switched my SIP FWD account to IAX and connect my * in IAX. Working great, but I face one problem: I have an iaxtel account and try to call from there (iaxcomm) my FWD Iax # by 17009xxxxxx. It's ringing but no termination on my *. Calling 17009999612 or 17009<SIP FWD#> is working (SIP FWD# being an normal SIP account on FWD). Has someone some idea on what's

I found this configuration file on Wiki for FWD behind firewall ; SIP Configuration for Asterisk ; [general] disallow=all allow=ulaw port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind SIP channel to externip=xxx.xxx.xxx.xxx localnet=172.16.1.0 localmask=255.255.255.0 context=inbound-sip ; Default context for incoming calls maxexpirey=180 defaultexpirey=160 tos=reliability

Greetings, We are looking at installing a VoIP system with Asterisk and are currently looking at the line of 3Com phones. Has anybody had success with using the following phones? We need to buy a lot and we don't want to end up with phones that don't work properly with asterisk. 3Com 3101 (model with speakerphone) 3Com 3102 Business Phone 3Com 3103 Manager Phone 3Com 3105

Anyone know what model and what support the new $100 Cisco has? http://biz.yahoo.com/djus/030428/1030001060_1.html -- Steven Critchfield <critch@basesys.com>

I've tried to register * at FWD but * segfaults, so i guess my register-line in sip.conf isn't correct. It looks like: register => fwd#:pwd@192.246.69.223 should it be different? Chris

[Just starting out with DTrace and was hoping to get some guidance.] I have a "benchmark" program that I monitored with both prstat (prstat -mL -P <PID>) and pfilestat (from the DTrace toolkit). Prstat reports LAT values in the 0.1-0.2% range, but pfilestat reports "waitcpu" values in the 6-10%. Since those two numbers supposedly represent time waiting for the CPU,

Hello: Many thanks to Professor Ripley for responding to my earlier post, included below, about trying to use mingw32 gcc.I did put back the original Rinternals.h. I am still crashing R every time I try to use C code. Here is what I did, basd on readme.packages: 1.) Ensured that C:\Rtools was first in my path, and obtained a new tools.zip from BDR's R tools site. 2.) Ensured that I

On Wed, 15 Sep 2004 19:59:01 +0200 "Henrik Bach" <henrik_bach_llvm at hotmail.com> wrote: > >From: Jeff Cohen <jeffc at jolt-lang.org> > >Date: Wed, 15 Sep 2004 10:35:36 -0700 > > > > >What's a "compiling mesh?" > > What I meant, was that there are some implicit defines in mingw (like __GCC) > and vcX (like _MVC) but

Has there been any luck getting the 3Com NBX series phones to work with Asterisk? Thanks! -Andrew

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Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the "very inexpensive" Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to

On Thu, 2004-01-15 at 19:18, dkwok@iware.com.au wrote: >> I do not have any zaptel hardware on the Asterisk box, I could not have >> meetme functioning. I did modify the Makefile in zaptel directory on >> line 168 by including ztdummy as one of the modules to compile in. try modprobe ztdummy This works. Should I include this in /etc/asterisk/modules.conf so that it will