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Asterisk patches after v20 #9
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I had look at porting the patch to chan_pjsip, however because the features there are split into separate modules it wouldn't work as for example; the phones expect subscriptions to be working during register. The easiest option would be to port chan_sip to Asterisk 21 but remove any parts that the phones don't need. |
To strip back chan_sip it would probably be better to rename the transport to something like chan_ucm (or similar) I've only been using chan_sip for Cisco devices for quite some time now. I'm guessing that would be the same for most people with Cisco devices. |
Are you or anyone else stepping up to maintain chan_sip in Asterisk 22+? Can you expand more on why it would not be possible to port to pjsip? Is it a matter of being a lot of work and no ability/time/motivation to do it, or an actual limitation in the pjsip architecture that makes it impossible to support Cisco's phones? |
chan_pjsip is implements SIP as a set of separate modules that may or may not be loaded. It is possible to support registering endpoints but not subscriptions or mailboxes etc, also some of the request/response handling is done via generic callbacks that don't support endpoint specific behaviour. Cisco phones need almost all of SIP to be available and in the cases where Cisco-specific extensions are required those callbacks will need to be modified to include an extra parameter. That would take additional extra work over what it took for chan_sip and it would never be accepted upstream, at that point it would be better to create a new Cisco specific SIP channel driver that also uses libpjsip. |
Yes, a community Asterisk developer is maintaining chan_sip. And it DOES still apply to Asterisk 22. (which just went to RC1) The repository is located here: https://github.com/InterLinked1/chan_sip Note that the To-Do List on his repo's readme states: "The usecallmanager patches are not currently incorporated here because they present a huge merge conflict." This is not, in fact, true. |
Just wondering what the future holds for these patches after Asterisk 20?
It's clear from the codebase of Asterisk 21 that chan_sip is removed.
I guess the options would be to:
Have you put any thought into this?
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