FreePBX set up and get working

Hi. I have experiences with Asterisk and FreePBX. I can help you to fix your FreePBX problem on SPA phone registering issue, No audio issue and fix pstn trunk problem. Hope can work and discuss more details with you. Thanks

hi bid updated i have more then 14 years experience with asterisk, check my feedback note, there are no way grant passing of double nat. in some configs it not doable. recommended install something simple like openwrt router with asterisk or enother small linux box at that location.

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I have done a lot of asterisk projects. Please check my profile for feedback. Your project is clear. To install freepbx, please provide a server with linux. The installation include asterisk, database, web and freepbx. Then configuration of extensions.

Hi sir , i am an expert VOIP developer with 5 years of experience , this task is very easy , i'll describe you the process : 1- Preparing Server for the install (installing requirements) : 1 hours 2- Installing FreePBX with all requirements and modules : 2 hours 3- Create Extensions and Destinations and Inbound / Outbound routes and ctrunks : 5 hours 4- Test work remotly : rest of time my skype is mrdevelopper you can see my reviews here i can start immediatly Best regards

Hello, Thank you for your message. Let me summarize your situation for more clearly: 1. You have asterisk PBX ( with FreePBX ) on location 1. There is 1 IP phone ( with 3 SIP accounts). This phone can register to PBX and making call ( incoming calls & outgoing calls) properly. NO problem with this site 2. There are 5 IP phones : 3 Cisco + 1 Linksys + 1 Gigaset (with 10 SIP accounts) on location 2. Cisco & Linksys phone cannot register to PBX at Location 1. Gigaset can register to PBX at location 1 but no sound when making call. 3. There is 1 IP Phone: Gigaset ( with 1 SIP account ) at location 3,4,5. They can register to PBX at location 1 but there is no sound. In my opinion, your trouble is networking trouble. All you need to do is fixing network issue with NAT. As you may know, you must open these port on Modem (Router) as following: SIP trunking: 1.1 Open SIP port 5060 - UDP (for SIP signal ) 1.2 Open range port from 10000 to 20000 - UDP ( for audio ) On FreePBX, configure SIP extension using NAT In case you still get trouble, you could try with IAX. IAX only need open 4569 port for all. I think I can fix this trouble for you. If you still have any question, please let me know. I rebid and hope we will work together ASAP. Rgds, Vu Le

Hi I can do this for you ... Also looking for good reviews ...................................... THnx

Hi, I am an individual working for a UK VoIP company, we happen to use Asterisk with Elastix/Free PBX as our IP PBXs, I have over 2 years experience so I am pretty sure I can handle your project. Kindly give me a breackdown of you requirements. Respectfully, Emanuel

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Hi, I already using asterisk almost 7 years. Distro using freePBX, AsteriskNow, Elastix, and TrixBox CE and prefer to use elastix. Trunking using digium card and also SIP but now I am using sip trunking. Until now all going well. you can contact me by email at nad(at)j28.in. Brgds, Nad

I fully understand the purpose of your project. I have 10 years of experience in this field (Asterisk/Elastix IP PBX configuration and installation) and hope I will be able to finish your project in 2 days upon your request.

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Hello, I have already done similar situation, although you should consider implement a site-to-hq VPN logical topology to avoid further nat related issues. My first opinion: - sip signaling ports are not forwarded correctly to cisco ( not registering) - audio rtp ports are in the same situation since you have "no audio" on siemens.