You are new to this! A good victim... ( you really need to read a bit, start by downloading Asterisk, The Future of Telephony from a site like http://cdn.oreilly.com/books/9780596510480.pdf or read the online html versions. Then either print or buy a copy to mark up as you learn)
That is actually the way I set the context statement in my settings. Unless you have specifically created a working [from-voipo] context in the configuration files, I would not use that.
The settings to 'publish' your * box location to the external world will depend on whether your IP is static or dynamic. and your domain name.
For example, I use static IPs and have statements 'like' (meaning these are not real) this in the sip.conf files:
externip=72.85.94.206
localnet=192.168.45.0/255.255.255.0
On my PBXiaF box, I insert them in the sip_custom.conf file since the sip.conf is automatically maintained by FreePBX (in other words, don't change it, it will just get over-written). I have not played with TrixBox since they went a bit over the edge a while ago.
There have been some good recent discussions of proper setups for asterisk over at the http://www.dslreports.com/forum/voip forums.
I'll be truthful, this is all over my head, I will start reading and playing around and see if I can make things work. The one statement that stands out to me here is " I have not played with TrixBox since they went a bit over the edge a while ago." can you please explain what you mean by this? I thought trixbox was the defacto standard open source setup for getting asterisk up and running. Maybe I am wrong. I am just starting to dabble in these setups because the company I work for wants me to start looking into voip phone systems instead of straight digital "nortel" type systems. Yes I am fresh fish and am just starting to get wet, but dont want to head down the wrong direction....I think what you are referring to is the buyout by fonality if I'm not mistaken.
Thanks for all the help.
I started out on Trixbox several years ago and it was great. Over the years in my opinion they have become kind of a mess and have had some controversy. I switched over to PBX In a Flash and it has been great. I would recommend taking a look at that. It is similar to Trixbox as in it includes everything you need to get set up, but I think it is easier and less bloated. Some will also say your privacy is more protected..
Also, as someone already mentioned, you might be better off testing and learning with a VOIP provider that supports BYOD more openly than VOIPo does. I love VOIPo and their service and support is fantastic. I'm a big fan, but wouldn't be my first choice for learning Asterisk. There are a few pay as you go providers that support Asterisk and have good configuration settings and things like that. I don't know if it is appropriate to mention here but PM me if you like. The VOIP Tech Chat forum at broadbandreports.com is also a great resource.
Scott
I had no problems at all gettting * talking to the BYOD server -- the only time I plugged in the shipped adapter was when I initially got it to be able to get my SIP credentials, and then for some testing for a couple days.
Here's the FreePBX trunk config I used:
Dial rules:
Outgoing:Code:911 1NXXNXXXXXX 1+NXXNXXXXXX 1763+NXXXXXX
Incoming settings: Leave this entire section blankCode:username=763NPXXXXX type=peer session-timers=accept session-refresher=uac secret=as3cr37 rfc2833compensate=yes qualify=5000 nat=no insecure=port,invite host=sip.voipwelcome.com dtmfmode=auto disallow=all context=from-trunk canreinvite=no allow=ulaw
Register:
Then setup an inbound dial rule to match your DID.Code:763NPXXXXX:as3cr37@sip.voipwelcome.com:5060/763NPXXXXX
Note: If you're behind a NAT device, make sure to set NAT=yes, define a port range in rtp.conf, setup externip or externhost + localnet in sip_general_custom.conf to define your external and internal IP ranges, and of course forward UDP port 5060 + your UDP RTP port range on your public IP-facing NAT device to point in at your * box.
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