Trixbox Asterisk Help

**bigdessert** · 11-18-2009 12:02 AM

I am trying to play around with Trixbox and test it. I figured the easiest way to do that is with Voipo. I am pretty sure I have things setup somewhat correct as I can make outgoing calls fine, but no incoming.

I also have the PAPt2 registered on the network at the same time and dont know if this is causing the problems. Basically if I call my voipo number my pap2t rings, but not the trixbox.

here are my configs.

Code:

Outgoing:
disallow=all
username=218257xxxx
type=peer
secret=mysecret
qualify=yes
nat=yes
insecure=port,invite
host=sip.voipwelcome.com
fromuser=218257xxxx
fromdomain=sip.voipwelcome.com
context=from-sip
allow=ulaw

Incoming:
disallow=all
type=peer
secret=mysecret
qualify=yes
nat=yes
insecure=port,invite
host=sip.voipwelcome.com
context=from-sip
allow=ulaw

Registration:
218257xxxx:mysecret@sip.voipwelcome.com/218257xxxx

If I login to vpanel I can see the box registered. I also have 1 incoming route setup for Any DID/Any CID and have tried:ringing all, ringing 1 extension,ivr,extension voicemail.

I am not sure what I have wrong, but hopefully someone that has this working can see a problem.

Also this with my PAP2t are both behind a Cisco RV042 firewall and I forwarded port 5060 to the ip of the asterisk.

Thanks in advance for any help.

**chpalmer** · 11-18-2009 12:39 AM

Take the pap2 out of the equation and try it..

I have the same problems here trying to run two Voipo ata's behind the same public IP...

**bigdessert** · 11-18-2009 08:17 AM

interesting. When I disconnect the pap2 then I get "we're sorry, the number you are trying to call has been disconnected or is no longer in service. Please check the number and dial again."

any ideas

**gbh2o** · 11-18-2009 10:08 AM

Originally Posted by bigdessert

interesting. When I disconnect the pap2 then I get "we're sorry, the number you are trying to call has been disconnected or is no longer in service. Please check the number and dial again."

any ideas

I believe you will find that VOIPo requires that their ATA be online before any other credentials will work, unless you are one of the grandfathered original beta testers.

You are using the softphone/BYOD credentials, are you not? ;-)

context=from-trunk ; change for your proper context

How do you identify/advertise your * box IP to the world?

Actually, you might do better to use a free/cheap account with a VoSP that supports * and BYOD for your experimenting.

**bigdessert** · 11-18-2009 11:58 AM

Originally Posted by gbh2o

I believe you will find that VOIPo requires that their ATA be online before any other credentials will work, unless you are one of the grandfathered original beta testers.

You are using the softphone/BYOD credentials, are you not? ;-)

context=from-trunk ; change for your proper context

How do you identify/advertise your * box IP to the world?

Actually, you might do better to use a free/cheap account with a VoSP that supports * and BYOD for your experimenting.

Yes, I am using my BYOD settings.

context, so if my trunk name is voipo should my context be from-voipo?

my * is behind a firewall router that has an internal ip. my external ip is mapped to a domain name.

**gbh2o** · 11-18-2009 02:32 PM

Originally Posted by bigdessert

Yes, I am using my BYOD settings.

context, so if my trunk name is voipo should my context be from-voipo?

my * is behind a firewall router that has an internal ip. my external ip is mapped to a domain name.

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.

**fisamo** · 11-18-2009 03:13 PM

What voice is that recording speaking with? Check your Asterisk logs - does it indicate that the call is being received and rejected? I ask, because you may (at least temporarily) want to set "Allow anonymous SIP calls" to Yes under the General Settings tab within FreePBX.

Before you go changing any settings, do check your Asterisk log just after making a call to your number (with or without your PAP2T on). If the call is properly hitting your Asterisk system, the logs will help you troubleshoot.

**bigdessert** · 11-18-2009 09:47 PM

Originally Posted by gbh2o

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.

**scott2020** · 11-18-2009 10:05 PM

Originally Posted by bigdessert

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

**SpaethCo** · 11-18-2009 11:52 PM

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:

Code:

911
1NXXNXXXXXX
1+NXXNXXXXXX
1763+NXXXXXX

Outgoing:

Code:

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

Incoming settings: Leave this entire section blank

Register:

Code:

763NPXXXXX:as3cr37@sip.voipwelcome.com:5060/763NPXXXXX

Then setup an inbound dial rule to match your DID.

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.