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sdesalve / hassio-addons

MIT License
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fastweb sip not working #82

Closed Shiroe93 closed 6 months ago

Shiroe93 commented 9 months ago

Hi i have been try to configure the add-on with fastweb sip but with no luck (i have it working with microsip perfectly)

here's full log

`[s6-init] making user provided files available at /var/run/s6/etc...exited 0. [s6-init] ensuring user provided files have correct perms...exited 0. [fix-attrs.d] applying ownership & permissions fixes... [fix-attrs.d] done. [cont-init.d] executing container initialization scripts... [cont-init.d] 00-banner.sh: executing...

Add-on: DSS VoIP Notifier VoIP Notifier for Home Assistant

Add-on version: 4.0.0 You are running the latest version of this add-on. System: Home Assistant OS 10.5 (amd64 / generic-x86-64) Home Assistant Core: 2023.9.2 Home Assistant Supervisor: 2023.09.2

Please, share the above information when looking for help or support in, e.g., GitHub, forums or the Discord chat.

[cont-init.d] 00-banner.sh: exited 0. [cont-init.d] 01-log-level.sh: executing... [cont-init.d] 01-log-level.sh: exited 0. [cont-init.d] done. [services.d] starting services [services.d] done. [Info] Starting addon... PJSUA_CUSTOM_OPTIONS = '--stun-srv=stun.l.google.com:19302' [Info] Listening for messages via stdin service call... [Info] Received messages {"call_sip_uri":"sip:+39XXXXXXXX@voip.fastwebnet.it","message_tts":"ciao"} Converting audio file 'http://home assistant ip (one subnet network):8123/api/tts_proxy/1e4e888ac66f8dd41e00c5a7ac36a32a9950d271it-_google_translate.mp3'... Audio succesfully converted... Starting SIP Client and calling 'sip:+39XXXXXXX@voip.fastwebnet.it'... This call will be terminated after '50' seconds. 16:28:22.160 os_core_unix.c !pjlib 2.11.1 for POSIX initialized 16:28:22.160 sip_endpoint.c .Creating endpoint instance... 16:28:22.160 pjlib .select() I/O Queue created (0x7f1cc5429100) 16:28:22.160 sip_endpoint.c .Module "mod-msg-print" registered 16:28:22.160 sip_transport.c .Transport manager created. 16:28:22.160 pjsua_core.c .PJSUA state changed: NULL --> CREATED 16:28:22.191 pjsua_core.c .pjsua version 2.11.1 for Linux-6.1.45/x86_64 initialized 16:28:22.237 pjsua_app.c .Turning sound device -99 -99 ON 16:28:22.237 main.c Ready: Success 16:28:22.267 pjsua_app.c .......Call 0 state changed to CALLING

Account list: [ 0] : does not register Online status: Online [ 1] <sip:172.30.33.12:5060;transport=TCP>: does not register Online status: Online *[ 2] sip:XXXXXXX@voip.fastwebnet.it: does not register Online status: Online Buddy list: [ 1] <?> sip:+39XXXXXXX@voip.fastwebnet.it +=============================================================================+ Call Commands: Buddy, IM & Presence: Account:
m Make new call +b Add new buddy . +a Add new accnt
M Make multiple calls -b Delete buddy -a Delete accnt.
a Answer call i Send IM !a Modify accnt.
h Hangup call (ha=all) s Subscribe presence rr (Re-)register
H Hold call u Unsubscribe presence ru Unregister
v re-inVite (release hold) t ToGgle Online status > Cycle next ac.
U send UPDATE T Set online status < Cycle prev ac.
],[ Select next/prev call +--------------------------+-------------------+
x Xfer call Media Commands: Status & Config:
X Xfer with Replaces
# Send RFC 2833 DTMF cl List ports d Dump status
* Send DTMF with INFO cc Connect port dd Dump detailed
dq Dump curr. call quality cd Disconnect port dc Dump config
V Adjust audio Volume f Save config
S Send arbitrary REQUEST Cp Codec priorities

+-----------------------------------------------------------------------------+ | q QUIT L ReLoad sleep MS echo [0|1|txt] n: detect NAT type | +=============================================================================+ You have 1 active call Current call id=0 to sip:+39XXXXX@voip.fastwebnet.it [CALLING] 16:28:22.281 pjsua_app.c .....Call 0 is DISCONNECTED [reason=403 (Forbidden)] 16:28:22.331 pjsua_app.c .NAT detection failed: Server Error [status=370500] 16:28:23.238 pjsua_app.c .Turning sound device -99 -99 OFF No current call 16:29:13.705 timer.c .Dumping timer heap: 16:29:13.705 timer.c . Cur size: 0 entries, max: 3070 [Info] Call ended... [Info] Listening for messages via stdin service call...`

sdesalve commented 9 months ago

addon config?

Shiroe93 commented 9 months ago

addon config?

`sip_parameters: caller_id_uri: sip:username@voip.fastwebnet.it:5060 realm: "*" username: "**" password: *** pjsua_custom_options: "--stun-srv=stun.l.google.com:19302"

Tried with and without port `

sdesalve commented 9 months ago

Try to add --no-tcp to pjsua_custom_options

Shiroe93 commented 9 months ago

Try to add --no-tcp to pjsua_custom_options

Since I already have one how I add a second opinion? (formatwise I mean)

sdesalve commented 9 months ago

Leave a blank space between the options. You can also remove stun to test call. Stun will solve audio issues behind a NAT

Shiroe93 commented 9 months ago

Leave a blank space between the options. You can also remove stun to test call. Stun will solve audio issues behind a NAT

Microsip won't start any call without the stun option though Anyway I Wil try

Shiroe93 commented 9 months ago

Leave a blank space between the options. You can also remove stun to test call. Stun will solve audio issues behind a NAT

Tried with both options and only with tcp No dice

Shiroe93 commented 6 months ago

Why you marked it as completed? Am I missed something?