FreePBX 14.0.1.20: Incoming calls from outside network do not go through

Environment: FreePBX 14.0.1.20 running on a Dell PowerEdge server.

SIP Trunk Configs:

Outbound:
Trunk Name: ESVC-TRUNK
host=siphosthame
username=sipusername
secret=sippassword
type=peer

Inbound:
USER Context: ########## (our main phone number)
description=ESVC-inbound
host=siphostname
type=friend
dtmfmode=auto
allow=all
insecure=port,invite
canreinvite=no
context=from-trunk

We have two VoIP servers on our premises: A Barracuda Networks CudaTel that we’re transitioning away from and the new FreePBX server. Both are pointed to our ISP’s SIP host, with separate accounts. We have one Polycom VVX400 phone provisioned on the FreePBX server with extension 50. Provisioning works fine and we can make outbound calls on that phone no problem. We have an inbound call route defined that is supposed to pass any DID number that touches it. There is an inbound DID associated with extension 50, and it also shows up in the Inbound Routes.

If we try to call the DID line from an IP phone in the same office as the FreePBX server, it goes through (a phone connected to the CudaTel). If our ISP tries it from their office, it also goes through. No calls from outside our network or our ISP’s network go through. We get the following error messages when following the log:

[2017-12-14 12:24:01] WARNING[11388][C-0000009e]: chan_sip.c:17235 check_auth: username mismatch, have <ESVC-TRUNK>, digest has <s>

[2017-12-14 12:24:01] NOTICE[11388][C-0000009e]: chan_sip.c:26312 handle_request_invite: Failed to authenticate device "757XXXXXXX" <sip:[email protected]>;tag=as2c984c3f

Where 757XXXXXXX is my cell phone number and 162.etc is the IP address of our ISP’s SIP server. These errors occur whenever someone from outside our office tries to call the DID line. So it appears that the server is trying to authenticate the outside calls? I’ve looked around this forum and done some deep-diving on Google, but I’m coming up short. Any suggestions? I can provide config files and/or more detailed log entries, if need be, including what we see with a successful call.

I’ve also checked and re-checked the firewall settings. There’s nothing there that I can see that would be blocking the traffic. We are not running the firewall on the FreePBX server, since we have a corporate firewall at our perimeter.

Restarting the Asterisk service from the command line solved this problem. Incoming calls are now going through.

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