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:
port: 5060

Primary Proxy:
port: 5060

Local Port: 15060

Outbound servers:
port: 5060



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

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.

username=<phone number>
secret=<SIP password>
host=<SIP server IP>

USER Context: <phone number>

secret=<SIP password>

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

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