TIPS VoIP.ms trunk not working on iPBX 2027-U

[np1070]

I've been successfully running Incredible PBX 2027-D on a RPi 4 for a little over a year now, with around 30 endpoints, mostly located remotely. About a week ago, my ISP had a nearly 20 hour outage. When service was restored, all remote endpoints could dial and connect, but there was no audio. Local endpoints worked perfectly between each other. I ran every test I could think of, and concluded that it had to be ISP related. My ISP denied any kind of blocking or restrictions.

I used this as an "excuse" to migrate everything over to CrownCloud. The transition went amazingly smoothly, the lost audio problem completely resolved. Everything is great, except for one snag... I am using a VoIP.ms number as an inbound/outbound trunk. It has worked perfectly on the RPi, but I cannot make it work on the CrownCloud installation (2027-U). On the original setup, I would get the Authentication Type Registered in Green on the VoIP.ms Manage Subaccounts page. Now it's red. When attempting to dial out, I get the All Circuits are Busy recording, and when I dial the number from outside, I get a busy signal. No matter what I do on the CrownCloud version, I cannot get it to register. I tried changing it to Authentication Type: Static IP Authentication, no change. I've compared the Connectivity-Trunks-Edit Trunk-SIP setting in iPBX for outgoing and incoming with the working installation, no luck. I feel I must be overlooking something obvious, but I'm completely out of ideas on my own.

If anyone has any suggestions, I would me most grateful!

 

[dallas]

It looks like your trunks are not registering to CrownCloud. There may be multiple other issues with your installation. Make sure you logging is setup correctly and collect the logs from startup until you get all circuits are busy and look for errors. If you find that overwhelming, post the complete log here (don’t forget to redact sensitive data) and someone will help you understand what’s happening.

 

[islandtech]

under Settings --> Asterisk Sip Settings --> General Sip Settings --> Nat Settings
Did you click Detect Network Settings?

 

[np1070]

Thank you dallas and islandtech for your replies and help! I've made some great progress. I did the Settings --> Asterisk Sip Settings --> General Sip Settings --> Nat Settings --> Detect Network Settings and now when I type sip show registry from the CLI, it responds State Registered, and Outbound calls are successful!

Now it seems the only remaining problem is inbound. When I dial the trunk number, I get a "the number you have dialed is not in service" recording, and not the IVR as defined in the Inbound Routes Destination. If I disable the trunk under Connectivity --> Trunks ---> Edit Trunk --> Disable Trunk and call in, I get a busy, so it appears iPBX is recognizing the registration.

In the Inbound Routes settings, I have DID Number "Any", CallerID Number "Any", Set Destination "IVR/Demo IVR". If I change the Destination to an Extension/number, there is no change, still get the Not In Service recording. If I dial the IVR 3366 number from and extension, I reach the IVR as expected.

I looked at the log and saw from-sip-external: "Rejecting unknown SIP connection from xxx". I found a thread on here discussing this error, and it mentioned the port number not matching. When I look on VoIP.ms subaccounts registration is says it's on port 5060, and on my iPBX under Asterisk SIP settings --> SIP Settings [chan_pjsip] --> Port To Listen On --> 5061. I see no place to change this on VoIP.ms, but on iPBX it looks like it will allow me to change it. So, my $64K question is, is this setting global and if I change this in iPBX will that blow up all my endpoints?

I decided to try it, and it gave me an error, "PJSIP transport port conflicts with SIP port". Not sure at this point at all what to try next...

 

[islandtech]

Inbound route should be defined with your did.
try 1xxxxxxxxxx or xxxxxxxxxx

set Settings --> Asterisk sip settings --> sip settings [chan_pjsip] --> Port To Listen On --> 5060

also 5060 in trunk settings --> sip settings [chan_pjsip] --> Port To Listen On

 

[np1070]

Thanks for the reply. "You say Inbound route should be defined with your did". I edited the Inbound Route and added my trunk number in the DID field. No change. As mentioned before, if I try to change the SIP settings [chan_pjsip] Port To Listen On to 5060, it says it conflicts with [chan_sip] port. I changed the [chan_sip] to 5062, then changed the [chan-pjsip] port to 5060, and it blew up every extension. I restored everything back to where I started and I at least have all the extensions back. There must be something I'm overlooking!

 

[islandtech]

What are you calling trunk number?

 

[np1070]

The 10 digit phone number.

 

[islandtech]

What type of trunk sip or pjsip?

 

[np1070]

SIP. At least that's what's listed on the VoIP.ms Sub Account Registration Status page, and on iPBX Connectivity Trunks page.

 

[dallas]

As mentioned before, if I try to change the SIP settings [chan_pjsip] Port To Listen On to 5060, it says it conflicts with [chan_sip] port.

Yes you can't set sip and pjsip to the same port. I run chan_pjsip (port 5061) exclusively on my pbx except for the VOIP.ms trunk which is running chan_sip on port 5060.

 

[np1070]

Here's the latest, When I call into the trunk number, I get a "The number you have reached is not in service" recording. In the log, I get an error: "Rejecting unknown SIP connection from xx.xx.xx.xx". I've added the IP address to the Whitelist, no change. I then go to Settings --> SIP Settings --> General SIP Settings --> Security Settings and set Allow Anonymous Inbound SIP Calls to "yes", and it works as expected, incoming calls going directly to the IVR. I obviously don't want to leave this setting the way it is, just enabled it for testing. I have checked and double checked the username/password from VoIP.ms, and everything else I can think of, no luck. I feel like I'm getting closer to a solution... ;-)

Any ideas would be greatly appreciated!

 

[dallas]

Here's the latest, When I call into the trunk number, I get a "The number you have reached is not in service" recording. In the log, I get an error: "Rejecting unknown SIP connection from xx.xx.xx.xx". I've added the IP address to the Whitelist, no change. I then go to Settings --> SIP Settings --> General SIP Settings --> Security Settings and set Allow Anonymous Inbound SIP Calls to "yes", and it works as expected, incoming calls going directly to the IVR. I obviously don't want to leave this setting the way it is, just enabled it for testing. I have checked and double checked the username/password from VoIP.ms, and everything else I can think of, no luck. I feel like I'm getting closer to a solution... ;-)

Any ideas would be greatly appreciated!

If anonymous SIP is working then there is something wrong with the inbound route CID or the DID doesn’t match. Try setting the route CID to Any and check the log for the DID being sent.

 

[ostridge]

Regarding the pjsip incoming port, generally not a good idea to use 5060 as that is the port that gets bombarded by bad actors looking for a voip connection to abuse.

 

[np1070]

dallas, the Route CID has been set to "any" right along.

I did a "sip show peers" and shows the status is OK, so I'm guessing the authentication is correct? The Host IP it returned is different from the one shown in the log during an failed login. The forth octet it .37 instead of .35. I added that to the Whitelist, same results.

Next, I deliberately put in a bad password under Connectivity --> Trunks --> Edit Trunks --> Sip Settings --> Incoming, with the with the Allow Anonymous Inbound Sip Calls set to NO, I get a busy signal, with the correct password, I get the Cannot complete call recording, so I am assuming the password I'm using is correct and not the problem. With Anonymous Calls set to YES, either way works properly and I get the IVR.