ALERT Asterisk 18 Missing ChanSIP

[dallas]

I have watched people update to v18 on Asterisk from older versions, not realize chan_sip no longer compiles

@Samot is this correct?
I have a home system, Incredible PBX 2021.01, pi4. It was built late 2020 or early 2021. I have upgraded both Asterisk & FreePBX and it's now Asterisk 18.5.0 and it has chan_sip. I checked the source directory and chan_sip is there.

 

[Samot]

@Samot is this correct?
I have a home system, Incredible PBX 2021.01, pi4. It was built late 2020 or early 2021. I have upgraded both Asterisk & FreePBX and it's now Asterisk 18.5.0 and it has chan_sip. I checked the source directory and chan_sip is there.

It depends on the upgrade and what is being done. In your case you are using a project that deals with this on your behalf. Those just doing straight Asterisk have run into this problem. Even worse, when you finish compiling 18 it will tell you "chan_sip.so found but not being used" or something along that line. I can't recall the exact wording but you are told, you compiled a version of Asterisk that isn't using X, Y, X modules it found .so for.

 

[kenn10]

@dallas, @wardmundy includes chan_sip in his builds of IncrediblePBX. It is optional for now but default Asterisk does not include anymore without changing includes during download and install. That is why it is a good idea to bite the bullet and start using pjsip. I forget when chan_sip will no longer have any security updates.


jcolpJoshua C. ColpAsterisk Technical Lead

Sep '19

Just so people don’t just read the title and freak out, an important take away:
It has reached the point where chan_pjsip sufficiently serves the vast majority of users, and that the time is right to transition chan_sip to the “deprecated” support status, in favor of chan_pjsip. As of Asterisk 17’s release, there will be at least a 4-year time frame before the potential removal of chan_sip from Asterisk may happen. At that time, it will be up for debate and discussion.
The module is still there, still enabled, still works. Any possibility of actual removal is 4 years away.

 

[dallas]

That is why it is a good idea to bite the bullet and start using pjsip.

Sadly I have a number of Nortel / Avaya desk phones that work with chan_sip but I haven't been able to get them to work with pjsip. They register and after the registration timeout they fail to re-register and can't be called.

 

[krzykat]

Sadly I have a number of Nortel / Avaya desk phones that work with chan_sip but I haven't been able to get them to work with pjsip. They register and after the registration timeout they fail to re-register and can't be called.

Yeah, unfortunately, I've had a similar experience with some Polycom's. I'm trying to move everyone off those types of phones and get everything on PJSIP. Another sidenote on this that I think @dicko recommended one time is that my PJSIP is on a port other than 5060 which helps tremendously with the idiot routers by Comcast and the like that think they understand SIP and screw things up with SIP ALG. This problem is so pervasive that even large companies like Nextiva recognize this and have changed their ports.

 

[Samot]

I've had a similar experience with some Polycom's

Which Polycom's?

Sadly I have a number of Nortel / Avaya desk phones

Got some configs and debugs to show?

A device not sending a re-register isn't an Asterisk problem. It's a device problem. There needs to be some clarification if the device is actually sending and Asterisk is ignoring it or handling it in a way it shouldn't or it's not sending re-registrations.

 

[kenn10]

Sadly I have a number of Nortel / Avaya desk phones that work with chan_sip but I haven't been able to get them to work with pjsip. They register and after the registration timeout they fail to re-register and can't be called.

We're way off topic for this thread. I have since ditched my Avaya 9621 phone but. as I recall, you can put the port number at the end of the server IP address or FDQN.

Example: [email protected]:5061

Unfortunately, these legacy proprietary phones have minimal sip support. Things like distinctive ring, BLF, etc. are not supported. The message waiting notification on the Avaya phone was intermittent at best. Anyway, try appending the port number and see if you can get it to work for you. The Grandstream phones work the same way. They expect 5060 unless you append a different port on the server address. If the phones don't reregister, try setting the registration timeout in the PBX to zero.

 

[Samot]

I have since ditched my Avaya 9621 phone but. as I recall, you can put the port number at the end of the server IP address or FDQN.

Example: [email protected]:5061

Unfortunately, these legacy proprietary phones have minimal sip support. Things like distinctive ring, BLF, etc. are not supported. The message waiting notification on the Avaya phone was intermittent at best. Anyway, try appending the port number and see if you can get it to work for you. The Grandstream phones work the same way. They expect 5060 unless you append a different port on the server address. If the phones don't reregister, try setting the registration timeout in the PBX to zero.

If the phone registers properly the first time, it hit the proper port. Nothing at the PBX side will make the phone send a new register request. Registration timeout settings in PJSIP are for registration sections which handle outbound registrations.

This is all about the AOR settings which only control the min and max expire times and what to do with existing contacts and how many to allow.

If the phone sends a register with an expire time of 60 seconds, Asterisk will keep that contact for 60 seconds. The device must send a new register.

 

[dallas]

Got some configs and debugs to show?

Nothing current and I'm recovering from surgery and can't take traces at the moment.
I'm interested in your comment about AOR. It was set to 1 for the extension involved. I have changed that to 10 and will test when I've recovered.

@kenn10 you are correct about being off topic. I will start a new thread when I've done some more testing.

 

[wardmundy]

FYI: These messages already have been moved to this new thread.