Buona sera a tutti
ho provato a configurare l'addon in oggetto con i parametri voip del mio numero telecom.
provato mille combinazioni ma ricevo sempre lo stesso errore.
`19:42:30.464 os_core_unix.c !pjlib 2.9 for POSIX initialized
19:42:30.466 sip_endpoint.c .Creating endpoint instance...
19:42:30.466 pjlib .select() I/O Queue created (0x7f9dc960f0)
19:42:30.466 sip_endpoint.c .Module "mod-msg-print" registered
19:42:30.466 sip_transport.c .Transport manager created.
19:42:30.466 pjsua_core.c .PJSUA state changed: NULL --> CREATED
19:42:30.503 pjsua_core.c .pjsua version 2.9 for Linux-5.4.83/aarch64 initialized
19:42:30.506 pjsua_app.c .Turning sound device -99 -99 ON
19:42:30.507 main.c Ready: Success
19:42:30.509 pjsua_app.c .......Call 0 state changed to CALLING
Account list:
[ 0] <sip:172.30.33.12:5060>: does not register
Online status: Online
*[ 1] sip:+39091123456@d91s2.co.imsw.telecomitalia.it: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:+391111111111@d91s2.co.imsw.telecomitalia.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:+391111111111@d91s2.co.imsw.telecomitalia.it [CALLING]
19:42:30.563 pjsua_app.c .....Call 0 is DISCONNECTED [reason=403 (Forbidden)]
19:42:30.563 pjsua_app_common.c .....
[DISCONNCTD] To: sip:+391111111111@d91s2.co.imsw.telecomitalia.it
Call time: 00h:00m:00s, 1st res in 56 ms, conn in 0ms
19:42:31.506 pjsua_app.c .Turning sound device -99 -99 OFF
No current call
inizialmente avevo dei problemi con i dns (telecom vuole i suoi a tutti i costi) sotto suggerimento dello sviluppatore ho provato a configurare il mio numero su MicroSIP e ci sono riuscito e sul pc funziona, adesso non riesco a riportare quei dati sulla configurazione dell'addon
giro la configurazione:
sip_parameters:
caller_id_uri: 'sip:+39091123456@d91s2.co.imsw.telecomitalia.it'
realm: '*'
username: '+39091123456'
password: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
pjsua_custom_options: >-
'--proxy=sip:d91s2.co.imsw.telecomitalia.it' '--nameserver=85.38.28.7'
'--nameserver=85.38.28.6' '--no-tcp'
qualcuno di voi c'è riuscito?
avete consigli?
