I am new to this forum, so if I am in the wrong place I apologize.
I am on FreePBX v14 Distro on an Asterisk SuperMicro server all handsets are Sangoma S505
I need some guidance on FreePBX registering on port 5060 with my provider NexVortex and subsequent call registering and port traffic. NexVortex requires all registration and signaling on port 5060.
I was able to register two trunks correctly with them back in March when I did the initial installation.
Here is my registrations from the Asterisk CLI-
dcmpbx*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
px15 . nexvortex . com:5060 Y 225 Registered Fri, 04 Jun 2021 15:04:55
px11 . nexvortex . com:5060 Y 225 Registered Fri, 04 Jun 2021 15:04:55
2 SIP registrations.
So our trunks are registered correctly.
I thought communications were ok, however they contacted me and said that I was using the incorrect port (5160) for outgoing call signaling and registration attempts. We also are experiencing intermittent connection and audio issues on outgoing calls. Incoming everything seems fine.
I originally configured all extensions as CHAN_SIP and now realize that these are using port 5160 not 5060. This week, I tried a test and converted a single extension to PJ_SIP (deleted the CHAN_SIP extension, reset the Sangoma S505 to factory settings, created a NEW PJ_SIP extension, re-provisioned the handset and did some test calling - same results as before.
Here is a log entry from NexVortex support. From my NEW PJ_SIP extension (x305) I called my cell phone (925-260-5790), you can see the invite from our public IP is on 5160 to NexVortex
2021-06-03 20:18:34 +0000 : 126.96.36.199:5160 -> 188.8.131.52:5060
INVITE sip:19252605790 @ 184.108.40.206 SIP/2.0
2021-06-03 20:34:31 +0000 : 220.127.116.11:5160 -> 18.104.22.168:5060
REGISTER sip:px11 . nexvortex . com SIP/2.0
Here is the AOR from the Sangoma-
Aor: 305 1
Contact: 305/sip:305 @ 10.10.99.17:5060 1254da2843 Avail 7.731
ParameterName : ParameterValue
authenticate_qualify : false
contact : sip:305 @ 10.10.99.17:5060
default_expiration : 3600
max_contacts : 1
maximum_expiration : 7200
minimum_expiration : 60
qualify_frequency : 60
qualify_timeout : 3.000000
remove_existing : true
support_path : false
Shows it is set up for communication on 5060.
So I thought that this test call from the PJ_SIP extension would show a call registration on port 5060.This is not the case. Can anyone point me to what might be causing this and how to correct it?
If you need more log/data please let me know.
Our firewall is set to no transformations and simple NAT policy (nowhere using port 5160). If it were transforming I am sure it would be a different port than 5160. I suspect it is in FreePBX and the extensions but at a loss as to where to start. I do have some log files from the calls but they are the FreePBX logs and not much use. I can do some firewall logs on the outgoing and see what is there if needed.
Thank you in advance for your assistance.