modules.conf: update for Asterisk 22.x.x #468
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A number of Asterisk modules were deprecated (and removed) in 21.x.x and 22.x.x. These changes update the ASL3 provided "modules.conf" file to sync with those changes as well as some overdue cleanup. *** NOTE *** The file currently includes some added content and a bunch of questions. For those reviewing the changes, your comments and guidance is welcomed.
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I happened to notice this message in the log. Should or can res_adsi.so be changed to noload now?
David WD5M |
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@wd5m : you have raised a good question. A quick scan of the Asterisk 22.1.1 sources shows that the following modules would emit that message :
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Updated/added module descriptions
and expanded the module descriptions that were truncated
I think |
It hasn't actually been removed yet. One of Sangoma's customers or FreePBX (forget which) raised a big stink on it, which temporarily deferred that. Haven't heard any plans of it to be removed in 23 yet, but it'll probably be removed eventually so best to avoid it. The other 3 modules in the warning (getcpeid, adsiprog, and res_adsi) I saved from removal and and am the maintainer of now. They aren't going to go away, but Josh didn't want to change the status back from "deprecated" to "extended" which is why the warning is there. PhreakScript patches that warning to a notice since it's a bit confusing and can cause alarm for no reason. |
configs/rpt/modules.conf
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| load = format_gsm.so ; Raw GSM data | ||
| load = format_h263.so ; Raw H.263 data | ||
| load = format_h264.so ; Raw H.264 data | ||
| load = format_ilbc.so ; Raw iLBC data |
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Do you really need all these formats? Highly skeptical anyone is using ILBC for example, or G729, etc. And if they are, they shouldn't be
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The only format that I added was format_mp3.so
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I think most of the other formats could be safely removed then. If we are already disabling most of the modules in this config, many of which are useful to many people, there is little reason in enabling obsolete codecs that nobody uses or should use
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Q? when making or receiving a call do we negotiate based on the available / enabled formats? Where do we get that list of formats?
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It's as defined in iax.conf + depending on which ones of those are actually loaded/available
These days, I would expect most calls should just be using ulaw or alaw unless there's a good reason not to
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In iax.conf I see mentions of codecs, not formats
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In some cases I think they go hand in hand, but that's right, they're a bit different.
I believe codecs get used for the call, e.g. codec_ulaw, and formats are used for recording/playback, e.g. format_ulaw/format_pcm.
configs/rpt/modules.conf
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| noload = res_http_websocket.so ; HTTP WebSocket Support | ||
| noload = res_musiconhold.so ; Music On Hold Resource | ||
| load = res_rpt_http_registrations.so ; RPT HTTP Periodic Registrations | ||
| load = res_smdi.so ; Simplified Message Desk Interface (SMDI) Resource |
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I'm not objecting to it, but I'm pretty confident most ASL users aren't using SMDI
A number of Asterisk modules were deprecated (and removed) in 21.x.x and 22.x.x. These changes update the ASL3 provided "modules.conf" file to sync with those changes as well as some overdue cleanup.
*** NOTE ***
The file currently includes some added content and a bunch of questions. For those reviewing the changes, your comments and guidance are welcomed.