Nextiva (pjsip/chansip) Trunk Configuration

I am having an issue with inbound and outbound calling on a brand new FreePBX 15.0.16.53 (Current Asterisk Version:16.9.0) installation.

  1. I have configured the FreePBX server and registered a phone at extension 101.
  2. I have created both a pjsip and chansip connection to my Nextiva trunk and they both say that they are registered (I know both are not needed).
  3. I created an inbound route directly to extension 101 on a registered phone.
  4. I created an outbound route using the pjsip trunk connection because I think that is what I should be using, (ideally?).

See ->

PJSIP

Nextiva_1/sip:myhost:5060 Nextiva_1 Registered
Nextiva/sip:myhost:5060 Nextiva Registered
Nextiva2/sip:myhost:5060 Nextiva2 Registered
Nextiva3/sip:myhost:5060 Nextiva3 Registered
Nextiva4/sip:myhost:5060 Nextiva4 Registered

Objects found: 5

SIP

Host dnsmgr Username Refresh State Reg.Time
sip:myhost:5060:5060 Y 123456789 3585 Registered Thu, 04 Jun 2020 18:55:53
1 SIP registrations.

CHANSIP

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
Nextiva_ChanSIP/123456789 208.73.146.93 Yes Yes 5060 Unmonitored
Nextiva_ChanSIP2/123456789 208.73.146.93 Yes Yes 5060 OK (63 ms)
2 sip peers [Monitored: 1 online, 0 offline Unmonitored: 1 online, 0 offline]

I have a lot of trunk registered because I was trying to find the right configuration.

When I call my DID I get a message from Nextiva saying the wireless caller is not available. This is the same message I get when no SIP trunk is registered.

I have disabled the firewall on the FreePBX server. Also, we have a Sonicwall in front of the PBX that I increased the UDP timeouts on to 120. I installed Zoiper on a machine on the same LAN as FreePBX and can connect to the Nextiva SIP trunk and make and receive calls.

This is the format I am using for my registration string.
[Authentication Name]:[Authentication Key]@[Nextiva Host]/[Authentication Name]

Can anyone point me in the right direction to get these inbound and outbound calls functioning?

Allow Asterisk to do NAT handling. Specify your public IP address in Asterisk SIP Settings. Disable SIP ALG / helper / “fix-up” on the Sonicwall so that it does not also try to handle the NAT transition. Those are the places I would start. After that, I would capture SIP traffic (perhaps at different points in the network) to look for problems.

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