How can I disable chan_sip? unfortunately i am still very much a beginner in this ![]()
Can I do it on the freepbx web interface?
You can âChange to chan_pjsipâ your extensions in the extension â advanced gooey, Trunks need to be reconstituted as using chan_pjsip, when all is done you can disable chan_sip in advanced settings.
I would suggest you spend some time in the wiki, it explains lots of things better than I will, chan_sip is âdeprecatedâ so you best move trunks and extensions to chan_pjsip ASAP
My trunk is a custom trunk and I canât find or just donât know which menu item to disable chan_sip ![]()
SIP channel driver in advanced,
Get that all done, restart fwconsole and then explain you âcustom trunkâ
Unfortunately, it still doesnât work. I tried to set astdatadir to / usr / share / asterisk, but in a very interesting way it didnât work
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Why would you do that ?
OK, a very pertinent question "How did you install your system? "
I followed this links to install the system: chan-dongle-extended
How To Install FreePBX 15 On Ubuntu 18.04 » TechnologyRSS
Wow, not an easy thing to install a third-party and only âcommunity supportedâ driver onto a very ânon-standardâ and four year old OS installation, especially for one who is âvery much a beginnerâ
I would start over after searching the wiki for a better way to install FreePBX on a current debian like OS.
Chan_dongle will be a problem though as it is a little âlong in the toothâ and no longer well maintained. Maybe
https://groups.google.com/g/chan_dongle
But I will now âbow outâ from this thread.
Good luck though .
. . .as an afterthought
or
probably be a much easier 2022 way to go.
Can I connect freepbx to the pstn network only with chan_dongle or this pstn connector? ![]()
Chan_dongle only works with asterisk 13, but when I install version 18 instead of asterisk 13 everything works
Cheapest way is with a SIP service provider, GSM with a working gateway , Analog, TDM ISDN with gateways or cards.
Your dongle will always be a liability going forward but if your happy, then knock yourself out.
And interestingly, asterisk plays files uploaded in the music on hold menu when I call from pstn, But the files uploaded in the systemcrecordings menu will not be played ![]()
So voice menu system unfortunately I canât put in the beginning of the callđ
Hi, thank you for your time and help. Now I run into another problem. The freepbx does not detect DMTF tones from the PSTN. I have a custom trunk because chan_dongle only works with this trunk. Can you help me with that? ![]()
I donât believe mobile networks support accurate transmission of DTMF in the fixed to mobile direction. chan_dongle is third party code and I believe no longer maintained. The GSM specs say it is for further study in GSM networks.
(In the mobile to fixed direction, they are transmitted, out of band, as key codes, and the tones are generated in the base station or mobile switching centre. The codecs used for mobile phones are based on vocal tract models and cannot reproduce combinations of two pure tones accurately.)
SOLVED!! The problem was that chan_dongle was configured with asterisk 13 by default, which was deprecated, so I upgraded to asterisk 16, where astdatadir was already set to var / lib / asterisk by default. Thanks for all the supportđ