Good Morning,
I’ve stepped in it and need some help.
FreePBX 14.0.13.12
Asterisk 15.7.3
Flowroute DID
Linux version 3.10.0-957.21.3.el7.x86_64
Our system was working fine until about 2 months ago it stopped registering with Flowroute. Looked like the issue was we were still using a chan_Sip trunk. So, I created a new PJSIP Trunk (using 5160) and followed the guide from Flowroute. (Our phones listen on 5060) https://support.flowroute.com/customer/en/portal/articles/2960886-freepbx-pjsip-trunk-setup
Registration fixed and all things worked fine. (Sip Registration with Flowroute)
Yesterday, we started getting calls dropped. Registration was failing/reconnecting over and over. I followed another guide to switch over to IP registration with flowroute.
(Nice, well done guide btw) I made all the changes in FreePBX and Flowroute.
However, this did not solve the registration problem. I felt this was b/c I was using 5060 for my sip phones and 5160 for the pjsip Trunk. So, I tried to swap those. I didn’t find a ‘How To’ for that so I winged it. Fail. Mistake. Lost connection to 50 phones. Time to punt. Did a restore from 2 days ago. Ran some important system and module updates. Network power-cycled the phones and now I am back to where I was. Internal communication is fine but no Flowroute registration so no calls going in/out. Understandable b/c I have not (and don’t want to) go back to Flowroute Sip Registration. Thanks for following me this far.
So, my question is: Is my problem that my pjsip trunk is using 5160 and my phones are on chan sip with 5060? How Should This be setup for the long term? System is broken now so I might as well fix it right. Is there a guide for changing my sip phones on 5060 to pjsip phones and should I do that? I’d bet there are guides but I’m not finding them. Let me know if I did not describe necessary details about my predicament.
I would appreciate any guidance, suggestions, useful links you can provide.
Thanks in advance.
Understood. That’s incorrect terminology on my part. I switched to IP Authentication. That’s why I was not seeing Registration attempt messages in Asterisk.
Update. I’m using IP registration now. No more dropped calls. Outgoing calls always successful.
However, I would guess that 1/2 of the incoming calls get “unable to complete call” message from ILEC. Any suggestions on what might be causing those failures or how to troubleshoot it?
Second, my pjsip trunk is using 5160 and my phones are on chan sip with 5060? Is that okay? Is there a guide for changing my sip phones on 5060 to pjsip phones and should I do that? Thanks.
Check your logs. I’m going to guess that half your calls are coming in as “anonymous” calls since you didn’t add the alternate POPs for your provider in the PJ-SIP Settings.
It’s not bad, but it is suboptimal. Chan-SIP is going away. There are VERY few cases where Chan-SIP is preferred over PJ-SIP now.
In your “extensions” app, there is an option to change your Chan-SIP extensions to PJ-SIP.
The gotcha is that I don’t think you can run PJ-SIP on two different channels at the same time. The “least painful” way to do this would be to move PJ-SIP to 5060 and update your ITSP settings to use 5060 instead of 5160 for your incoming connections.
Note, this is not a painless or easy way to do this - it’s possible that you’ll have to make configuration changes on the phones for some settings. Maybe not, but maybe. Also, the change from 5160 to 5060 for your provider is similarly not painless. All the work and testing you did to get that set up needs to be reaccomplished.
The advantage for you is that your system will then be set up for the removal of Chan-SIP as an active component coming soon. Doing it now, while you have the box open, would (to my way of thinking) be preferable to having to remember all of this stuff in a couple of years when you HAVE to do it.
For others coming to this thread or searching for it later, chan_sip still works fine for Flowroute trunks with IP authentication, you just need to define a chan_sip trunk for each of the now-many IPs that Flowroute uses for incoming calls. I put this in sip_custom.conf and used some asterisk syntax for setting this up to reduce the length.