# Success! My first Asterisk PBX **URL:** \ **Category:** Asterisk General\ **Created:** [July 16, 2008, 10:20pm UTC](https://community.asterisk.org/t/success-my-first-asterisk-pbx/22000 "2008-07-16T22:20:07Z")\ **Posts on this page:** 4\ **Page:** 1
**Author:** ![BrianMiller](https://avatars.discourse-cdn.com/v4/letter/b/9de053/32.png) [@BrianMiller](https://community.asterisk.org/u/BrianMiller)\ **Post date:** [July 16, 2008, 10:20pm UTC](https://community.asterisk.org/t/success-my-first-asterisk-pbx/22000/1 "2008-07-16T22:20:07Z")
Just thought I’d share my experience in deploying my first Asterisk system. I used the 1.6.0-beta9 source, built it on CentOS5 x86\_64 running in ESX. (Virtual Machine.) I’ve got three Cisco 7961G-GE phones loaded with SIP firmware SIP41.8-3-5S, a free DID from [ipcomms.net](http://ipcomms.net) and a pay-as-you go outgoing SIP trunk from [talkinip.com](http://talkinip.com). I also have a few 7921G wifi voip phones I’m going to try out tonight running SCCP because Cisco doesn’t support SIP on the 7921 yet. From what I’ve read Asterisk supports SCCP. We’ll see 😄 The configuration took me a few hours to understand and make everything work. I have my DID ringing all phones, all phones have their own extension but share the same mailbox so the MWI is lit on all phones. I wanted the ability to just press the message button to check voicemail without entering an ID or password. As of right now I’ve been able to do everything I wanted to. Asterisk 1.6 is really nice. P.S. I’m using an ADSL 6000k/608k connection over a PIX 515 firewall/NAT and a 3750G-POE layer 3 switch with many different VLANs. I’ve read a few posts about people having problems transversing NATs and routes; although I’ve had no problems at all. I’ll post my configs if requested. Oh yeah, I only use this at home; well that and I have a few SIP programs I can connect with from the internet. ---
**Author:** ![lbeckm3](https://avatars.discourse-cdn.com/v4/letter/l/a3d4f5/32.png) [@lbeckm3](https://community.asterisk.org/u/lbeckm3)\ **Post date:** [July 17, 2008, 2:18pm UTC](https://community.asterisk.org/t/success-my-first-asterisk-pbx/22000/2 "2008-07-17T14:18:06Z")
Sweet! Congrats Brian! Sounds like a nice little system you have set up there. -Lyle ---
**Author:** ![BrianMiller](https://avatars.discourse-cdn.com/v4/letter/b/9de053/32.png) [@BrianMiller](https://community.asterisk.org/u/BrianMiller)\ **Post date:** [July 17, 2008, 4:11pm UTC](https://community.asterisk.org/t/success-my-first-asterisk-pbx/22000/3 "2008-07-17T16:11:23Z")
I’ve got a 7921G wifi phone working with the Skinny protocol. The configuration was slightly different but for the most part pretty simple. I had to upgrade the SCCP firmware from A.7 to 1.1(1). The original firmware the phone came with worked but had horrible echoing issues and had problems hanging up calls. 1.1(1) works great! I can post configs if requested. What’s the difference between chan\_skinny.so and chan\_sccp.so? I realize they’re different shared objects/interfaces but why created two different Skinny/SCCP controllers? 1.6.0-beta9 only has chan\_skinny. From what I’ve read chan\_sccp is something you have to do manually. ---
**Author:** ![BrianMiller](https://avatars.discourse-cdn.com/v4/letter/b/9de053/32.png) [@BrianMiller](https://community.asterisk.org/u/BrianMiller)\ **Post date:** [July 28, 2008, 6:14pm UTC](https://community.asterisk.org/t/success-my-first-asterisk-pbx/22000/4 "2008-07-28T18:14:01Z")
Internal calls works great for the 7921G over the Skinny protocol and G.711. But whenever I dialed out through my SIP trunk to the public phone network the calls would get real choppy. I enabled all of the jitter control options in skinny.conf with not luck. Finally I turned off “Call Power Save Mode”, it works great now.