Configuring SIP TRUNK

[avichai_a (Avichai)]

Hi,

I have FreePBX 2.6.0 and I am installing now the FreePBX 16.0

1. Is there an automatic way to migrate the settings, extension, routes to the new version or I need to do it manually?
2. I am trying to configure the SIP Trunk

On version 2.6.0.1 there is a section called “PEER Details” and his content:

username=myusername
type=peer
secret=mysecret
qualify=no
nat=no
insecure=very
host=myhost
dtmfmode=auto
context=from-pstn
canreinvite=no
disallow=all
allow=alaw

Not all the parameters are found on the new FreePBX (pjsip settings)

Could you please help me to match the parameters from the old FreePBX to the new

Thanks

[comtech]

Help converting SIP trunk to PJSIP

This is an example of a working PJSIP trunk. Depends on your provider for different settings. [Inked1] [2]

The codecs (alaw, etc.) are on the codecs tab.

[avichai_a (Avichai)]

Thank you

I hope my configuration is better now.

Could you please help also with the Dialed Number Manipulation Rules

Here is my configuration on the new FreePBX

And this is from the old FreePBX

04+ZXXXXXX
012+00|.
012+900|.
012+01[3-9]|.
012+901[3-9]|.
1222+*|.

Is this the same or there some mistakes (I need to recreate them according to what configured on the old FreePBX)

Thanks again

[comtech]

Reading material:
PBX GUI : Outbound Routes Module User Guide (sangoma.com)

PBX GUI : Outbound Routes Configuration Examples (sangoma.com)