FreePBX incoming SIP settings HELP?

[butleredwin]

Hi

So I am trying to do a self install of FreePBX. I am making calls successfully but can't receive any , I have a SIP account with Centracom. Can anybody help me with the settings of the SIP trunk, and is there any router configs I have to do that I missed?

This is a PBX behind a NAT firewall.

I have done:

1. Tested the account on a softphone and works great (incoming and outgoing)
2. forwarded port 5060, 10001-20000 to my internal IP
3. Set my inbound routes (dont know if incorrect)
4. Successfully registered my SIP trunk (It shows on FreePBX that it is online)

Any help please.

 

[Clarotech]

Generally your VoIP provider should be advising you on how to configure your trunks.

What debugging have you done on the Asterisk CLI when you make a call in? What do your trunk configs look like?

Just a word of caution: if you're going to expose your Asterisk box to the general internet then you're going to want to make very sure you have secured it correctly. VoIP hacking is a very scary reality. There are some good tips here: http://mybroadband.co.za/vb/showthread.php/602150-Massive-hacking-attempts-to-hosted-PBX

 

[Vince0]

It's best if you describe the error or message you get exactly.

I often see an issue with FreePBX incoming calls where incoming calls result in a "invalid account code" type message being played back on an inbound call.

If I set the "USER context" on the trunk to the user name for the account, it routes inbound calls properly.

Vin.

 

[InfinityMVS]

If you are not running remote extension there is no reason for you to forward any ports to your pbx, you are just opening yourself up to a world of hurt.


Some trouble shooting steps

1. turn off the port forward.
2. check if you can make an outbound call.
3. remove previously created inbound routes and create new inbound route leaving the did and cli fields blank, this will create a any did/cli inbound route
4. make the destination for the above route terminate call -> put on hold forever, so when you call in and hear the sweet sound of elevator music you can fist pump and shout a satisfying "yes"
5. is the g729 codec installed and enabled in your settings -> asterisk sip settings page and at the top of the list
6. are you telling asterisk to nat on the same settings page, if you are using NAT you would choose yes or no and public ip, play around with these settings to work in your environment.
7. download putty, open and put in ip address of pabx, at the login prompt type root and your root password
8. at the cli in putty type asterisk -rvvv this will show you what is happening on the box and if calls are reaching your system or not, what number format centracom is sending 4 or 10 etc.
9. does your freepbx systems status page show green bars over ip trunks online and ip trunk registration, if registration is not green ask centracom what they require for a registration string, with most sip accounts on ippbx's you can make calls but not receive if the registration string is left blank.

some basic security tips
- under extension set type to peer
- under asterisk sip settings select sip guest->no, anonymous calls->no, other settings alwaysauthreject=yes
- make sure intrusion detection is on under system admin settings
- make sure password are as strong as possible...not your dogs name, best practice is 32 digit alphanumeric


trunk settings from a deployed system with SIP account behind dsl router - you only need to put in peer side
username=your username
type=peer
silencesuppression=no
secret=your password
reinvite=yes
qualify=yes
nat=yes
insecure=very
host=centracom sip server
dtmfmode=rfc2833
dtmf=rfc2833
context=from-trunk
canreinvite=yes
authname=your username

registration string example
username: password@sipserver/username

 

[Clarotech]

4. make the destination for the above route terminate call -> put on hold forever, so when you call in and hear the sweet sound of elevator music you can fist pump and shout a satisfying "yes"

 

[butleredwin]

InfinityMVS I thank you sir! I followed all your steps and seems like your configuration worked, and you were right at was very satisfying when I got the first call through "yes!!!"

 

[BiscuitSmuggler]

Nice going InfinityMVS... Eugene?

 

[e4et]

Very helpful, even after 2 years
Thanks

 

[oober]

Very helpful, even after 2 years
Thanks

If only I could figure out me one way audio problem on incoming calls! I'm going to try some of those trunk settings to see if it helps.

 

[oober]

Ok you guys are not gonna believe my I think I found what was going on and giving me gray hairs...

It seems like my VPS provider decided to start my VPS server which also used that SIP in it's asterisk configuration. So now it's turned off and things seem to be working now...

So I had two servers that were registering with the same SIP credentials.