Trunk setup ( sip settings )


(Australia) #1

Hi all i have Freepbx system at home for the last 11 years running over POTS by Spa 3102
last week they turn off the adsl / POTS for the fiber to the node
my provider sent me this

Provider name: Dodo Voip

Primary registrar: bwas01.voip.iprimus.net.au
port: 5060

Primary Proxy: bwas01.voip.iprimus.net.au
port: 5060

SIP domain:bwas01.voip.iprimus.net.au
Local Port: 15060

Outbound servers: sbc01a.mel.voip.iprimus.net.au
port: 5060

user:*****@dodo.com.au

pass:**********

but all i can find is the ( ADD TRUNK ) sip config is this below
or is there someware else or configs that are missing

host=provider ip address
username=userid
secret=password
type=peer

it will not register with the above config


(Ad Fun7911) #2

Are you using chan_pjsip or chan_sip? I’d recommend using chan_pjsip if there is no actual reason to use chan_sip.
There you’ll find a lot of options under ‘pjsip Settings’ and then ‘General’ and ‘Advanced’.


(Ruler2112) #3

I had trouble working out exactly how to get the trunk to register in my system. I’m in the US & don’t know if it’ll work on your system, but some of the lines came from an Aussie forum post I found. Here’s what I ended up with that works - hope it helps.

**Outgoing:**
username=<phone number>
type=peer
secret=<SIP password>
qualify=600
insecure=port,invite
host=<SIP server IP>

**Incoming:**
USER Context: <phone number>

type=user
secret=<SIP password>
context=from-trunk

Register String: <phone number>:<SIP password>@<SIP server IP>/<phone number>

(David55) #4

Given that, as is usually the case, you need insecure=invite, if this incoming section ever gets used, the call will fail, as you are asking the ITSP to authenticate, which they are going to refuse to do. Most systems only need a peer section, which will cover both incoming and outgoing.

insecure= only ever applies to incoming calls, so, if you need it in a section,that section is being used for incoming calls.


(Ruler2112) #5

Do you know of any explanation or reference material that goes over the possible lines in these sections @david55 ??? I was only able to make the above work after much online searching and trial/error experimentation. Was ecstatic when I finally stumbled upon something that worked, but now it seems like what works isn’t right… :cry:

I’m just a general IT guy trying to make a PBX work & generally don’t really know what I need until after the fact. What’s worse is that most of the time, I don’t even know where to look for help. :frowning_face:


(David55) #6

The primary reference for chan_sip configuration is sip.conf.sample, which comes with the Asterisk source code, but possibly not with FreePBX. You can find the latest version at:

I’m not sure where the idea of having separate incoming and outgoing sections comes from, as it is rarely needed to have separate sections.


(system) closed #7

This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.