BulkVS Trunk(s) using PJSip using Registration

[jaytaylormn]

So first off. Thanks to everyone for supplying such nice information on how to get bulkVS trunks connected on pjsip. I'm a chan_sip guy and I really need to spend some quality time with pjsip to figure it out.

Thanks also to WardM for all his work getting this published on nerdvittles.com as well.

Anyhow. I've gone through the messages on what to do to get a bulkvs trunk connected. I can make incoming and outgoing calls through bulkvs. I login to my bulkvs web portal and go to the dashboard and it still shows 0 SP Trunks with BulkVS. What's even weirder is that when I try to show calls under the "Your Active Calls" on the bulkvs dashboard for today, they don't acknowledge any of the calls I've made (in our out). I try to make the call at least 30 seconds so that there is plenty of chance for bulkvs to add it to the call list. It's a bonafied call completion.

I do a "pjsip show registrations" and it says Registerd next to the bulkvs trunk.

Under "Asterisk SIP Settings/ Sip Settings (chan_pjsip)" I have the Keep alive Interval set to 25. Bulkvs likes frequent re-connections.

I had "accept anonymous sip calls" turned on and then I can receive calls through bulkvs. I then turned accept anonymous off and the calls still completed. It's like it's a one time pass for bulkvs or "priming the pump, if you will" and then the system knows that it's OK to receive calls from bulkvs on this sip trunk. That was a good tip. I wouldn't have thought of turning it on to complete a call/connection and then turn it off afterwards. I'll remember that little tip.

So that's my ask. Do I really have a trunk connected to BulkVS if the dashboard doesn't acknowledge it?

I'll also take any other suggestions on how to make a better sip connection to bulkvs. I like the registered trunks vs IP registration. I need to have several bulkvs trunks from different accounts connected.

Thanks for your help!

 

[kenn10]

You have not set the "match" IP's in the advanced tab of the trunk on PJSIP. Look at the documentation on the Bulkvs site for all the IP's that can send a call to your system and add those IP's in the match field. Then the inbound calls are no longer "anonymous."



If you still have issues with inbound calls, make sure the IP's in the Match field have been added to your firewall. I don't recall if they are there by default in a new system so if you have issues execute /root/add-ip from the Linux prompt after logging in as user root.

 

[jaytaylormn]

Yes, I've done that. Interesting thing is that some of the addresses were already populated. I seem to recal reading someplace that sip.bulkvs.com has "proper SRV records" setup in DNS and that pjsip queries the SRV records and populates those field. Again I'm NOT a pjsip person so I have no real idea if this is true or not.
Jay

 

[jaytaylormn]

I should have given more information on that last post. Here's a quick update.

If you do a lookup on the SRV records for sip.bulkvs.com you get this info:

dig SRV _sip._udp.sip.bulkvs.com

(snip)
;; ANSWER SECTION:
_sip._udp.sip.bulkvs.com. 149 IN SRV 1 5 5060 la-198.bulkvs.com.
_sip._udp.sip.bulkvs.com. 149 IN SRV 1 5 5060 nyc-198.bulkvs.com.
_sip._udp.sip.bulkvs.com. 149 IN SRV 2 7 5060 dal-198.bulkvs.com.
(snip)

If you read the notes for the "Match (Permit)" setting under Advanced, the last sentence says "This setting is automatically generated by the PBX if left blank"

My guess is that pjsip does a SRV look up if this field is blank and then populates it with the IP addresses of the servers in the dns lookup..

My question would then be, What if blukvs updates the SRV records, do I have to recognize this and delete all the entries in this field so that pjsip can repopulate it with the new correct entries?

Does anyone know what pjsip is doing with these SRV records?

Thanks,
Jay

 

[kenn10]

Double check the port number on your PBX and insure that on the Bulkvs site that you send calls to your IP/DNS at that port. Bulkvs will default to port 5060 unless you implicitly set it at the end of the IP or DNS in the interconnect trunk groups manage section. Sending inbound calls to chan_sip when the trunk is on chan_pjsip (or vice versa) can result in an "anonymous" call.



Secondly, insure you have set your external address in the PBX on the Settings >Asterisk SIP Settings.

I've never seen the Match (Permit) set by default so I wouldn't count on the documentation being accurate.

 

[jaytaylormn]

I should have mentioned my version of Asterisk/FreePBX V16.0.40.7 It only has pjsip. The external and internal are the same. It's at a hosting location so no NAT to deal with.

Based on what the settings were for bulkvs.com I also was able to get connected to Vitelity. That was a bonus development.

Jay