I thought I'd post my experiences for the benefit of anyone else who may be at the point I was when I first started with asterisk.
I have 2 incoming analog lines (north eastern U.S., Verizon) where one is set to ring if the first is busy. I bought a bare-bones system from abs-pc with the following components: POWER SUPPLY 450W ALLIED ATX450P4 R(41) MB NFORCE2 A7N8X DELUXE ASUS RTL(Standard) CPU AMD|2500/333 ATHLON XP BARTON R(Standard) DDRAM 256M|DDR333 PC-2700 -K %(Standard) HD 40GB|WD 7200RPM 8MB WD400JB%(70) VGA ASUS|V8170MAGICII/T 64M MX440SE(58) CD ROM 56X|AOPEN CD-956 RTL(22) I also bought 2 X100P's and 1 TDM400P from Digium, and installed them in the above system. I installed RedHat 9 onto the PC. During the RH install, I selected the "server" install, and tried to weed out most of the packages that I didn't need. I'm no Linux expert, but I didn't want a lot of stuff running on my server. IMO simple is better (and more secure). Along these same lines, I ran the RH command 'setup' and turned off all of the services that I didn't need. I would do the same with the kernel, but I'm not that Linux savvy yet. Setting up Linux, installing Asterisk, and writing some basic conf files took about 2 weeks in my spare time. Most of that time was spent learning about asterisk, and what I needed to include in my conf files. My initial conf files were mostly adaptations of others that I found around on the net. I bought two radioshack single line phones (one was cordless), plugged them into the TDM400P. After getting the drivers loaded, and asterisk running, I ran into my first problem. I've covered this problem extensively in earlier posts (subject: "TDM400P??"), so I will just briefly mention it here. The Pro-SLIC modules were resetting on hook transitions. Its like they were not getting enough power. After much debugging, and work with Digium, the problem could not be solved. I sent the card back to Digium, and they sent me a new one. The new card behaved the same way. Mark edited the driver on my machine to prevent the module reset from crashing the wcfxo driver, but the problem was not solved. Eventually I came to accept that the card simply did not work with my motherboard, an ASUS A7n8X-Deluxe. Digium refunded my money for the card, and I returned to the drawing board. I bought a Grandstream 101, then I bought 2 more. I also got a Cisco ATA186. I had looked into using the ATA186 with asterisk, and it looked like I could get it to work. When I got it, I realized that It didn't have the same firmware as I thought it would. In fact, as it was, I couldn't get it to work with asterisk at all. I tried to get a firmware update from the Cisco website. Their website is ridiculously complex and annoying. In the end, though the web site didn't tell me this explicitly, I found that they would not let me download a firmware upgrade. Luckily I was able successfully navigate their huge and annoying phone system to reach an engineer who was nice enough to email me the SIP firmware upgrade "as a courtesy". After I loaded that firmware the Cisco ATA186 has worked good. The motherboard I am using has 2 Ethernet ports, but RH9 only recognizes one. I downloaded a Linux driver from NVIDIA, and had to manually edit the /etc/sysconfig files; redhat's config menus can't handle 2 Ethernet ports apparently. I set a DHCP server to run on the second Ethernet port, and also set up a NTP server for the grandstream phones' time display. I did not set up a route between the two ports. This gives me a separate isolated network for my BT-101's and the ATA186. I recorded audio using a regular PC mic, and Goldwave. Goldwave is nice as it lets you edit wav files, equalizing volumes, and applying filters. I converted the files from wav to gsm using Sox. After I got all of this set up I began testing after-hours in the office. The echo problem immediately became obvious. Everything else seemed to work good. I set the grandstream phones to use SIP-info for signaling, and spent some time massaging my conf files. After activating the Aggressive Suppressor option in the zaptel makefile, and recompiling the zaptel driver, the echo problem was greatly reduced on all but one grandstream phone. I noticed that one phone had older firmware. I set up a tftp server, and updated the BT-101's firmware. The firmware upgrade seemed to fix the remaining echo on that one phone. The echo is still audible as the occasional chirp or crackle, but it is now at a tolerable level. There is the additional problem of regular speech audio occasionally getting suppressed when both parties start talking at the same time. That is not a bad problem as it doesn't happen often, and quickly fixes itself. The current system is working good, except for the above mentioned problems with audio. The Grandstream phones' function buttons integrate nicely with asterisk. All of them seem to work. I loaded some nice (and hopefully legal) tunes for the musiconhold (had to install mpg123), and that works great. I haven't really had any experience with other PBX's, but after all of the work I put in, I am happy with the results. And I don't think anyone at my company suspects my often irrational commitment to open-source. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
