I have a cisco 8841 on Asterisk 18 with Freepbx 16. I tried the patch usecallmanagernz and Chan SCCP with SCCP manager but I’m still stuck on the following features :
Conference - When I try do to a conference, the phone say “Unable to complete conference”
Do not disturb - When I active it, I have a lot of log like this :
res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘PUBLISH’ from ‘sip:a4b2392ed5ec@172.20.10.9’ failed for ‘172.20.10.5:49305’ (callid: a4b2392e-d5ec0006-1e8fb521-517befd5@172.20.10.5) - Failed to authenticate
idivert - When I refuse a call, idivert softkeys do not refuse the call
I didn’t understand supervised calls though. How to supervise calls ?
Supervision refers to whether a call has been answered or not. I know this feature tends to behave differently depending on whether calls have been answered (just a limitation at the moment).
I would recommend reaching out to the actual author of these patches since this is not native mainline Asterisk functionality, so you won’t get support from the Asterisk team on these. He might be able to tell you whether it’s supposed to work and how it’s supposed to work. I don’t have any of these phones handy at the moment to do any testing myself.
The “lot of work” bit I’ve heard from him before, but “whole new channel driver”?
To me, that seems kind of pointless, given this is the SIP protocol, not the Skinny protocol, so a whole another channel driver seems rather dumb to me, personally. It should be enhancements to PJSIP, perhaps as a separate module where possible, and integrated in otherwise.
The nice thing about PJSIP is that unlike SIP, where Sangoma is not accepting features or enhancements to SIP, this could all be mainstreamed into the codebase, making it easier for people to use.
A problem with usecallmanager at present is that since it relies on patches, if you combine it with other patches to chan_sip, which have grown in number due to Sangoma’s refusal to accept them into the codebase, conflicts have grown more and more prevalent thus making a number of patches incompatible with each other without going through some kind of rebase hell. It’s really all falling apart.
It’ll definitely be a lot of work, probably a several month project, but feasible and probably inevitable at some point. The days of usecallmanager in its current form are numbered.
Let’s clear this up. Chan_SIP received updates and fixes until the first half of 2022. Things like adding UserCallManager to the mainline make no sense because A) it was never added before chan_pjsip was created. B) Since v17 (2019) chan_sip is noload on building and was slated for removal.
Adding anything like this to chan_sip in the last three years made no logical sense. Let’s be honest too, it’s been almost 10 years. The maintainer of the UseCallManager project has had ample time to deal with this and work with the dev team on issues. It would seem this project is joining the list of other Asterisk 3rd party projects that gave up the ghost after v12/v13 with the major overhauls and introduction of chan_pjsip most projects just gave up because Asterisk became “harder to do”.
Chan_SIP received updates and fixes until the first half of 2022.
chan_sip continues to receive bug fixes, even now.
New features or enhancements have not been accepted since 2021. That is what I’m referring to.
Things like adding UserCallManager to the mainline make no sense because A) it was never added before chan_pjsip was created. B) Since v17 (2019) chan_sip is noload on building and was slated for removal.
Neither of these precluded the author from getting it added to chan_sip for the longest time, really. As of this year, it’s no longer possible due to Sangoma policy, but for the previous ten years, it could have been if the author was willing to get them merged. This is stated on the issue itself. The patch author never decided to persue getting them merged.
Adding anything like this to chan_sip in the last three years made no logical sense. Let’s be honest too, it’s been almost 10 years. The maintainer of the UseCallManager project has had ample time to deal with this and work with the dev team on issues. It would seem this project is joining the list of other Asterisk 3rd party projects that gave up the ghost after v12/v13 with the major overhauls and introduction of chan_pjsip most projects just gave up because Asterisk became “harder to do”.
Exactly my point. The project had the chance to get merged, and now even that option is not available anymore. So now it has no choice but to be rewritten for PJSIP if it wants to go mainstream. At this point, it’s either going to be relegated to obsolete, outdated systems or become irrelevant.