Local IP sent between FreePBX systems on Same SIP trunk Provider

So for starts Site one Local IP is a 172...* and External is 216...* behind a pfSense Firewall

Destination Site is Local 192...* external is 209...* behind a Unifi USG-4-Pro

SIP Trunk Provider is 208...*

Trunk Settings at the destination are
Name: 1705******* (actual phone number)

PEER DETAILS:
type=peer
host=208...*
fromuser=1705*******
fromdomain=208...*
insecure=port,invite
disallow=all
allow=ulaw&alaw
qualify=no
nat=no
dtmfmode=rfc2833

Incoming
USER Context: 705*******

User Details:
type=peer
host=208...*
fromuser=1705*******
fromdomain=208...*
insecure=port,invite
disallow=all
allow=ulaw&alaw
qualify=no
nat=no
dtmfmode=rfc2833

Registration Sting : 1705*******:[REDACTED PASSWORD]:[REDACTED USERACCOUNT]@208.../1705******~120

Asterisk SIP Settings : External Address on each site is auto detected and is its respective IP
Local Networks: Internal Address on each site is auto detected and is its respective IP

Destination site NAT Settings:
NAT: No
IP Configuration: Public IP

Local Site NAT Settings:
NAT: No
IP Config : Static
Override External IP: Is not set but shows correct IP shadowed.

So the destination IVR Work’s So long as its not out local site calling We can call from anything else (cell phone, Land lines from another building, etc)

This error has presented itself in one other way as well. We have another site irrelevant to the one above where we can hear them but they can’t hear us.

A captured packet given to us but our SIP provider shows the following:

s=Asterisk PBX 14.7.5.
c=IN IP4 192...*
t=0 0.
m=audio 19796 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=maxptime:150.
a=sendrecv.

I have read other posts and enabled DTMF logging and nothing shows up on the destination IVR but its logged on out side as sent.

I have done PCAP’s from the FreePBX Server and Firewall as well as my PC and it shows a DTMF signal being sent on all steps.

I have also tested by using 2 different phones 1 Softphone (X-lite Free) and a Astra 6739i

Do you manage both FreePBX servers or just one of them?

This seems as a NAT misconfiguration. Are both FreePBX servers correctly configured for NAT, assuming they are both behind a NAT device?

The trunk seems to be configured with nat=no but it seems your FreePBX is behind a NAT device.

Yes we manage both. Both are behind a NAT, I will test a change with nat=yes in the SIP Settings and Trunk Settings

So NAT Was the issue and I thank you for taking the time to look over this.

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.