Re: Trixbox Asterisk Help
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...
Re: Trixbox Asterisk Help
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
Re: Trixbox Asterisk Help
Quote:
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.
Re: Trixbox Asterisk Help
Quote:
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.
Re: Trixbox Asterisk Help
Quote:
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.
Re: Trixbox Asterisk Help
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.
Re: Trixbox Asterisk Help
Quote:
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.
Re: Trixbox Asterisk Help
Quote:
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
Re: Trixbox Asterisk Help
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.
Re: Trixbox Asterisk Help
Quote:
Originally Posted by
SpaethCo
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.
This worked perfectly. Thanks for the help!
