Just fired up a few new servers with Asterisk 13 and I never used pjsip before so I’m wondering if if its suggested practice to start using pjsip or is adoption still too early? Obviously not looking to setup 3 servers only to find issues with all three due to pjsip… Thanks
It’s relatively easy to switch an extension between sip and pjsip. I just put 2 systems into production without any issues at all. The advantage is that you can have multiple endpoints per extension with pjsip.
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Thanks very much for the response… I’m setting up the new boxes today and will do it with pjsip. I am also converting 2 small clients from FreePBX 2 and were just building new boxes from scratch with no backup/restore process so I will cut them over also.
We ran late on these conversions but tried them both this weekend and they didn’t work. Not sure what we did but we couldn’t see any of the phones and needles to say could make any calls. If we setup PJSIP extensions do we need to set up PJSIP in the SIP Settings or any other required changes to make this all work? Any help would be appreciated.
When both pjsip and sip are active, you can use either. Not sure what you mean when you say you couldn’t “see” any of the phones. The phone system doesn’t really see phones. The phones make a request to register with the phone system. The system defaults of a new install should be more than enough to get any phone registered, considering that you are not traversing subnets or doing any sort of NAT.
Thanks for the response and here is a bit of further clarification:
We were actually converting a few running servers over to a new data-center and decided to do clean installs and Asterisk upgrades at the same time and just manually rebuild everything from scratch. Since we were going that route we figured we would also try PJSIP.
We carefully mirrored every setting from the working servers to the new servers and when we thought everything was ready we did an after hrs cut.
We had three issues hit us immediately:
- No outbound calls
- Couldn’t “see” the any phones in the Admin Restart Phones section (picture below)
- Our logs were overwhelmed with error messages and were almost useless due to the repeated errors and the volume
We cut back to the old servers and afterwards found the answer to problem 1. The trunks in our working servers had username=xxxxx in the peers details section but the new servers also needed fromuser=xxxxx once we did that the outbound calls were working
Item 2 we never figured out:
Item 3 we never figured out but were are currently working on it:
http://issues.freepbx.org/browse/FREEPBX-8986
For the time being we cut all the servers back to Channel SIP
I’m not familiar with the restart phones area. Though I notice mine is completely empty. Not sure what the deal is, but I’ll see if I can figure anything out.
I would double check two things:
Settings >Asterisk SIP Settings: Make sure local networks are all set correctly. In my case, we have 2 VPN conenctions to branch offices, so I have 3 local networks. Make sure the netmask is in long form (255.255.255.0) and not short form (24).
Settings > Asterisk SIP Settings > Chan PJSIP (top-right menu): Scroll to bottom and make sure Local Network is the network which is local to the phone system. In my case, the value is “10.0.1.0/255.255.255.0”.
If you have calls traveling over the internet, make sure the external IP is set correctly in both of the above locations.
Is your restart phones area empty even with Channel SIP? With ChanSIP all our polycom phones show but with PJSIP its always empty.
Settings >Asterisk SIP Settings: thats interesting we only use hosted services so no local networks but the netmask is definitely /24 and not the full mask. I will try and change that… thanks