[[email protected] ~]# asterisk -rx "module show like chan_sip"
Module Description Use Count Status Support Level
0 modules loaded
Both app_macro and chan_sip had a similar natural timeline as app_fax of being removed in Asterisk v19. The only reason (I believe) that app_fax didn’t get extended was that no one in the Asterisk team realized FreePBX was still using it. No one on the FreePBX made them aware (or did anything about it) so it was removed. When people moved to v19 when the RPMs were released, faxing got broke.
So the natural timeline was v19 but their removal got stayed to v21 specially because of this project. Since things can only be removed in Standard releases of Asterisk they will exist in v20 which has the side effect of these two things existing for another 4 years as it is LTS. In other words, FreePBX missed deadlines and got extra time to do their homework. Let’s not wait till the night before the new due date to get to that homework.
Please explain the nature of the lack of support (which is more likely to be the other way round, i.e. chan_pjsip doesn’t support something unusual that they do). As noted above there are issues with TEL, but that tends to affect newer systems, and partial support is being added. I also noted that I have seen specific claims of cases where chan_sip can work round unusual uses of SIP, but chan_pjsip can’t but I’ve been unable to find those again.
If you have specific experience of systems that require features that are missing from chan_pjsip, please let us know exactly what those are.
I read the point as confusing the implementation with the protocol, and thinking PJSIP was a non-backwards compatible replacement for SIP, whereas chan_sip and chan_pjsip are actually both implementations of the, single, SIP protocol. If someone is using an old version of Asterisk, they are likely to be using an old version of FreePBX, as well, and won’t be concerned about later versions.
No. To clarify, we’re not talking about older systems here, we’re talking about current and future PBX versions. chan_sip is going away in Asterisk 21, that’s set in stone now. It’s time (past time really) for all Asterisk based PBX admins to take whatever steps are needed to eliminate chan_sip from their setup. This thread raises points about how it will be phased out in FreePBX and when.