![]() Before doing so, make sure that your area and the areas you wish to call are covered by the service Create OnSIP Account You will need to open an account with OnSIP to use this service. ![]() OnSIP is a cloud-based, real-time business phone system. You can find all this information by logging in at, then select the user you want to configure and refer to the fields as pictured you see here > Odoo OnSIP Module Installation Go to webpage and create a new account.It offers InstaCall, a tool that lets visitors to your website make video calls to your sales or support team straight from their browsers. Go to Apps and install the module VoIP OnSIP. In the form view of each VoIP user, in the Preferences tab, fill in the section PBX Configuration: OnSIP Domain is the domain you chose when creating an account on If you don’t know it, log in to and you will see it in the top right corner of the screen.In the section Integrations/Asterisk (VoIP), fill in the 3 fields: ONSIP PHONE SYSTEM INSTALL PJSUA_CUSTOM_OPTIONS = '-outbound=sip: -no-tcp'SIP Login / Browser’s Extension: the OnSIP ‘Username’. Please, share the above information when looking for help System: Home Assistant OS 6.6 (amd64 / qemux86-64) You are running the latest version of this add-on. Here is the first part: making user provided files available at /var/run/s6/etc.exited 0. Here the configuration info from (same color => same word) | U send UPDATE | T Set online status | > 09:21:08.515 pjsua_app.c. | v re-inVite (release hold) | t ToGgle Online status | > Cycle next ac.| | H Hold call | u Unsubscribe presence | ru Unregister | | h Hangup call (ha=all) | s Subscribe presence | rr (Re-)register | | a Answer call | i Send IM | !a Modify accnt. | M Make multiple calls | -b Delete buddy | -a Delete accnt. | Call Commands: | Buddy, IM & Presence: | Account: | Temporary failure in sending Request msg INVITE/cseq=16193 (tdta0x7ff3b3524aa8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)Ġ9:21:08.289 pjsua_app.c. PJSUA state changed: NULL -> CREATEDĠ9:21:07.455 pjsua_core.c. Transport manager created.Ġ9:21:07.373 pjsua_core.c. Module "mod-msg-print" registeredĠ9:21:07.373 sip_transport.c. Creating endpoint instance.Ġ9:21:07.373 pjlib. This call will be terminated after '50' seconds.Ġ9:21:07.366 os_core_unix.c !pjlib 2.11.1 for POSIX initializedĠ9:21:07.370 sip_endpoint.c. Listening for messages via stdin service call. You'll find logs in /share/dss_voip/dss_autoanswer.log Or support in, e.g., GitHub, forums or the Discord chat. ![]() Please, share the above information when looking for hel System: Home Assistant OS 6.4 (armv7 / raspberrypi3) You are running the latest version of this add-on executing container initialization scripts. ensuring user provided files have correct perms.exited 0. This is the resulting log after running my script: making user provided files available at /var/run/s6/etc.exited 0. Note that I have to put username in quotes since it is numbers only (extension number). I tried various possibilities w/ and w/o port number, w/ and w/o custom options, the result is always the same: no sign of the SIP client starting. The add-on installs, registers, mp3 is created and available on link, but SIP client is not starting and calling the extension ends in voicemail.Ĭaller_id_uri: sip: (mypbx) I have a local pbx (mypbx, no external voip provider) and a raspi running home assistant. I have high hopes but I’m stuck with the config and need help.
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