[New Addon] DSS VoIP Notifier: Hassio makes phone calls to you! ☎️

nicojmb · December 8, 2023, 8:19pm

Hi,

Thank @sdesalve , but we tried with all of possibilities and do not work.

Just configured microsip and work fine with this parameters:

I don’t know what else to look at, here is my log:

[Info] Call ended...
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:[email protected]","message_tts":"Write here your message"}
Converting audio file 'http://192.168.86.39:8123/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '30' seconds.
21:24:10.347         os_core_unix.c !pjlib 2.11.1 for POSIX initialized
21:24:10.348         sip_endpoint.c  .Creating endpoint instance...
21:24:10.349                  pjlib  .select() I/O Queue created (0x7f6b473ee100)
21:24:10.349         sip_endpoint.c  .Module "mod-msg-print" registered
21:24:10.349        sip_transport.c  .Transport manager created.
21:24:10.349           pjsua_core.c  .PJSUA state changed: NULL --> CREATED
21:24:10.366           pjsua_core.c  .pjsua version 2.11.1 for Linux-6.1.63/x86_64 initialized
21:24:10.370            pjsua_app.c  .Turning sound device -99 -99 ON
21:24:10.370                 main.c  Ready: Success
21:24:10.396            pjsua_app.c  .......Call 0 state changed to CALLING
>>>>
Account list:
  [ 0] <sip:172.30.33.3:5060>: does not register
       Online status: Online
 *[ 1] sip:[email protected]: does not register
       Online status: Online
Buddy list:
 [ 1] <?>  sip:[email protected]

+=============================================================================+
|       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:[email protected] [CALLING]
>>> 21:24:11.371            pjsua_app.c  .Turning sound device -99 -99 OFF
21:24:40.337            pjsua_app.c  .Call 0 is DISCONNECTED [reason=487 (Request Terminated)]
>>> 21:24:41.872                timer.c  .Dumping timer heap:
21:24:41.872                timer.c  .  Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...```

sdesalve · December 8, 2023, 8:34pm

Full addon logs…

If you make mistakes on your addon config (as I think) and you putted pjsua options in sip_setting field I can know this reading FULL addon logs, from start to call end

nicojmb · December 8, 2023, 8:43pm

Hi @sdesalve,

If i put the paramenter mentioned in github thread, i’ve get 487 error (Request Terminated).

CONFIG

sip_parameters:
  caller_id_uri: sip:[email protected]
  realm: "*"
  username: xxxxxxxxx
  password: xxxxxxxxx
max_call_time: 30
pjsua_custom_options: "--no-tcp --proxy=sip:myurlof3cx.my3cx.es:5060;lr"

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 11.2  (amd64 / qemux86-64)
 Home Assistant Core: 2023.12.0
 Home Assistant Supervisor: 2023.11.6
-----------------------------------------------------------
 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 = '--no-tcp --proxy=sip:myurlof3cx.my3cx.es:5060;lr'
MAX_CALL_TIME = '30'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:[email protected]","message_tts":"Write here your message"}
Converting audio file 'http://192.168.86.39:8123/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '30' seconds.
21:39:26.501         os_core_unix.c !pjlib 2.11.1 for POSIX initialized
21:39:26.502         sip_endpoint.c  .Creating endpoint instance...
21:39:26.502                  pjlib  .select() I/O Queue created (0x7f1f1aca2100)
21:39:26.502         sip_endpoint.c  .Module "mod-msg-print" registered
21:39:26.503        sip_transport.c  .Transport manager created.
21:39:26.503           pjsua_core.c  .PJSUA state changed: NULL --> CREATED
21:39:26.521           pjsua_core.c  .pjsua version 2.11.1 for Linux-6.1.63/x86_64 initialized
21:39:26.525            pjsua_app.c  .Turning sound device -99 -99 ON
21:39:26.525                 main.c  Ready: Success
21:39:26.574      tsx0x7f1f1ab10c28  ....Temporary failure in sending Request msg INVITE/cseq=28300 (tdta0x7f1f1ab0aab8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
21:39:26.574            pjsua_app.c  .......Call 0 state changed to CALLING
>>>>
Account list:
  [ 0] <sip:172.30.33.3:5060>: does not register
       Online status: Online
 *[ 1] sip:[email protected]: does not register
       Online status: Online
Buddy list:
 [ 1] <?>  sip:[email protected]

+=============================================================================+
|       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:[email protected] [CALLING]
>>> 21:39:27.525            pjsua_app.c  .Turning sound device -99 -99 OFF
21:39:56.490            pjsua_app.c  .Call 0 is DISCONNECTED [reason=487 (Request Terminated)]
>>> 21:39:58.028                timer.c  .Dumping timer heap:
21:39:58.028                timer.c  .  Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...```

sdesalve · December 8, 2023, 8:50pm

Ok, pjsua options was correctly recognised

sdesalve · December 8, 2023, 8:51pm

But are you trying to call the same extension of hassio? 41?

nicojmb · December 8, 2023, 8:56pm

It’s removed for security…

I realized that when I add the parameters, I have this error:

21:55:03.687      tsx0x7f8d3b58ec28  ....Temporary failure in sending Request msg INVITE/cseq=12394 (tdta0x7f8d3b588ab8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)

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 11.2  (amd64 / qemux86-64)
 Home Assistant Core: 2023.12.0
 Home Assistant Supervisor: 2023.11.6
-----------------------------------------------------------
 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 = '--no-tcp --proxy=sip:myurlof3cx.my3cx.es:5060;lr'
MAX_CALL_TIME = '30'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:[email protected]","message_tts":"Write here your message"}
Converting audio file 'http://192.168.86.39:8123/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '30' seconds.
21:55:03.659         os_core_unix.c !pjlib 2.11.1 for POSIX initialized
21:55:03.660         sip_endpoint.c  .Creating endpoint instance...
21:55:03.660                  pjlib  .select() I/O Queue created (0x7f8d3b720100)
21:55:03.660         sip_endpoint.c  .Module "mod-msg-print" registered
21:55:03.660        sip_transport.c  .Transport manager created.
21:55:03.660           pjsua_core.c  .PJSUA state changed: NULL --> CREATED
21:55:03.679           pjsua_core.c  .pjsua version 2.11.1 for Linux-6.1.63/x86_64 initialized
21:55:03.684            pjsua_app.c  .Turning sound device -99 -99 ON
21:55:03.684                 main.c  Ready: Success
21:55:03.687      tsx0x7f8d3b58ec28  ....Temporary failure in sending Request msg INVITE/cseq=12394 (tdta0x7f8d3b588ab8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
21:55:03.687            pjsua_app.c  .......Call 0 state changed to CALLING
>>>>
Account list:
  [ 0] <sip:172.30.33.3:5060>: does not register
       Online status: Online
 *[ 1] sip:[email protected]: does not register
       Online status: Online
Buddy list:
 [ 1] <?>  sip:[email protected]

+=============================================================================+
|       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:[email protected] [CALLING]
>>> 21:55:04.683            pjsua_app.c  .Turning sound device -99 -99 OFF
21:55:33.647            pjsua_app.c  .Call 0 is DISCONNECTED [reason=487 (Request Terminated)]
>>> 21:55:35.186                timer.c  .Dumping timer heap:
21:55:35.186                timer.c  .  Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...

sdesalve · December 8, 2023, 9:11pm

You cannot call your same extension. But if you have edited this log for your privacy and it’s not same extension it’s ok.

Try to remove this

nicojmb · December 8, 2023, 9:35pm

Yessss, removing “–no-tcp” works!:

Here is my config for 3CX

sip_parameters:
  caller_id_uri: sip:extension_number@host
  realm: "*"
  username: xxxxxxxxx
  password: xxxxxxxxxxxxxx
pjsua_custom_options: "--proxy=sip:host:port;lr"

Thanks so much @sdesalve !!

larsvanbeek1997 · January 30, 2024, 10:31pm

@sdesalve Could you please help me?

I spend 2 hours finding the problem but I can’t find it.
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 11.4  (amd64 / generic-x86-64)
 Home Assistant Core: 2024.1.5
 Home Assistant Supervisor: 2023.12.1
-----------------------------------------------------------
 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 = '--no-tcp'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:+316[myphonenumer]@sip.freevoipdeal.com","message_tts":"TEST"}
Converting audio file 'http://192.168.0.11:8123/api/tts_proxy/984816fd329622876e14907634264e6f332e9fb3_nl_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:+316[myphonenumer]@sip.freevoipdeal.com'...
This call will be terminated after '50' seconds.
23:21:42.689         os_core_unix.c !pjlib 2.11.1 for POSIX initialized
23:21:42.689         sip_endpoint.c  .Creating endpoint instance...
23:21:42.690                  pjlib  .select() I/O Queue created (0x7f8352368100)
23:21:42.690         sip_endpoint.c  .Module "mod-msg-print" registered
23:21:42.690        sip_transport.c  .Transport manager created.
23:21:42.690           pjsua_core.c  .PJSUA state changed: NULL --> CREATED
23:21:42.714           pjsua_core.c  .pjsua version 2.11.1 for Linux-6.1.71/x86_64 initialized
23:21:42.721            pjsua_app.c  .Turning sound device -99 -99 ON
23:21:42.721                 main.c  Ready: Success
23:21:42.738            pjsua_app.c  .......Call 0 state changed to CALLING
>>>>
Account list:
  [ 0] <sip:172.30.33.5:5060>: does not register
       Online status: Online
 *[ 1] sip:[email protected]: does not register
       Online status: Online
Buddy list:
 [ 1] <?>  sip:+316[myphonenumer]@sip.freevoipdeal.com

+=============================================================================+
|       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:+316[myphonenumer]@sip.freevoipdeal.com [CALLING]
>>> 23:21:42.759      tsx0x7f8351cf86d8  .......Temporary failure in sending Request msg INVITE/cseq=19388 (tdta0x7f8351b9baa8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
23:21:43.720            pjsua_app.c  .Turning sound device -99 -99 OFF
23:22:14.759            pjsua_app.c  ....Call 0 is DISCONNECTED [reason=408 (Request Timeout)]
No current call

Config:

sip_parameters:
  caller_id_uri: sip:[email protected]
  realm: "*"
  username: droeloe1818
  password: [mypassword]
pjsua_custom_options: "--no-tcp"

Service:

service: hassio.addon_stdin
data:
  addon: 89275b70_dss_voip
  input: {"call_sip_uri":"sip:+316[myphonenumber]@sip.freevoipdeal.com","message_tts":"TEST"}

Phone is not ringing.

Dissabled SIP Alg in Omada controller did not solve the problem.

I am desparate and I hope you can help me!!!

Thank you so much

Lars

Update:

I have installed a sip client on my android phone and added the sip and it worked perfectly (also on my wifi network (the same network as my Home assistant is in))

so my network is not the problem.

i’ve also tried adding the portnumber and also tried adding --no-tcp --ip-addr=192.168.0.11 (you mentioned that above)

I realy want this to work

sdesalve · January 31, 2024, 5:37am

Have you enabled sip on freevoipdeal control panel?

larsvanbeek1997 · January 31, 2024, 9:06am

Hi,

Thanks

yes is enabled

I solved by using DSS VoIP Notifier for ARM and it is now working flawlesly!

Thanks for your help so far.

Could I ask another question?:

I don’t want to use google tts but I wish to use the Nabu casa tts.cloud_say.

When I put tts.cloud_say I get the error parse error:

PJSUA_CUSTOM_OPTIONS = '--no-tcp'
PLATFORM_TTS = 'tts.cloud_say'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:[email protected]","message_tts":"TEST"}
parse error: Expected string key before ':' at line 1, column 4
[cont-finish.d] executing container finish scripts...
[cont-finish.d] 99-message.sh: executing... 
[cont-finish.d] 99-message.sh: exited 0.
[cont-finish.d] done.
[s6-finish] waiting for services.
[s6-finish] sending all processes the TERM signal.

Config:

sip_parameters:
  caller_id_uri: sip:[email protected]
  realm: "*"
  username: droeloe1818
  password: Lars1968!
pjsua_custom_options: "--no-tcp"
platform_tts: tts.cloud_say

sdesalve · January 31, 2024, 9:26am

TTS.cloudsay it’s not correct service name.

Try remove TTS.

I don’t remember how was named nabu casa TTS service. Search this topic

maxtesio · January 31, 2024, 4:03pm

Ciao @sdesalve
Have you ever tested it with Eolo’s VoIP?

I tried few times with the credentials Eolo gave to me, but with no luck.

With this Config:

sip_parameters:
  caller_id_uri: sip:EVA******@voce.eolo.it
  realm: "*"
  username: EVA******
  password: *******
pjsua_custom_options: "--no-tcp"

it raises this log:

Account list:
  [ 0] <sip:172.30.33.6:5060>: does not register
       Online status: Online
 *[ 1] sip:EVA*****@voce.eolo.it: does not register
       Online status: Online
....

>>> 16:55:19.997            pjsua_app.c  .....Call 0 is DISCONNECTED [reason=403 (not registered)]

I previously tried config parameters with MicroSIP on the same LAN and they work

I’m sorry but I’m not very expert

sdesalve · January 31, 2024, 5:04pm

Never tried.

403’s errors are related to authentication issues…
Check login password and if you need proxy or other special things.

Post full log. Otherwise I cannot see if you have pjsua_custom_options in the correct field

maxtesio · February 1, 2024, 5:28pm

Here it is:

[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 11.4  (amd64 / generic-x86-64)
 Home Assistant Core: 2024.1.6
 Home Assistant Supervisor: 2023.12.1
-----------------------------------------------------------
 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 = '--no-tcp'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:+39*********@voce.eolo.it","message_tts":"Prova messaggio"}
Converting audio file 'https://*******.duckdns.org:8123/api/tts_proxy/772c970523ed7389b60b2ed1b674b3705d965e37_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:+39*********@voce.eolo.it'...
This call will be terminated after '50' seconds.
18:18:46.747         os_core_unix.c !pjlib 2.11.1 for POSIX initialized
18:18:46.747         sip_endpoint.c  .Creating endpoint instance...
18:18:46.747                  pjlib  .select() I/O Queue created (0x7fcebf186100)
18:18:46.747         sip_endpoint.c  .Module "mod-msg-print" registered
18:18:46.747        sip_transport.c  .Transport manager created.
18:18:46.747           pjsua_core.c  .PJSUA state changed: NULL --> CREATED
18:18:46.757           pjsua_core.c  .pjsua version 2.11.1 for Linux-6.1.71/x86_64 initialized
18:18:46.761            pjsua_app.c  .Turning sound device -99 -99 ON
18:18:46.761                 main.c  Ready: Success
18:18:46.777            pjsua_app.c  .......Call 0 state changed to CALLING
>>>>
Account list:
  [ 0] <sip:172.30.33.6:5060>: does not register
       Online status: Online
 *[ 1] sip:EVA*******@voce.eolo.it: does not register
       Online status: Online
Buddy list:
 [ 1] <?>  sip:+39*********@voce.eolo.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:+39*********@voce.eolo.it [CALLING]
>>> 18:18:46.809            pjsua_app.c  .....Call 0 is DISCONNECTED [reason=403 (not registered)]
18:18:47.761            pjsua_app.c  .Turning sound device -99 -99 OFF
No current call
>>> 18:19:38.273                timer.c  .Dumping timer heap:
18:19:38.273                timer.c  .  Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...

sdesalve · February 1, 2024, 5:48pm

https://forum.fibra.click/d/12438-configurazione-voip-eolo/76#:~:text=Metti%20server%20SIP%2010.100.0.105%20e%20server%20Proxy%2010.100.0.101.

Do you use EOLO DNS on your entire LAN?
Try to ping voce.eolo.it and use that ip to register and also add it as proxy with

--outbound=sip:voce.eolo.it;lr

maxtesio · February 3, 2024, 3:13pm

Nothing to do
I tried everything, changed caller_id_uri with my phone number, changed server with its IP, changed proxies, ect. Everytime the same problem (403). I’m almost sure that is not a LAN problem, because I tried on MicroSIP and a self-hosted instance of 3CX and they worked fine. Here my MicroSip configuration

could there be a conflict with let’s encrypt? I use it through duckdns integration for remote access and Alexa

sdesalve · February 3, 2024, 5:46pm

no

Have you tried this?
Could you put here full logs?

Have you pinged voce.eolo.it?

Have you also tested this kind of config? Can I see logs?

sip_parameters:
  caller_id_uri: sip:EVA******@10.10.10.0:5060 [resolved IP of voce.eolo.it]
  realm: "*"
  username: EVA******
  password: *******
pjsua_custom_options: "--no-tcp --outbound=sip:10.10.10.0:5060 [resolved IP of voce.eolo.it];lr"

Have you double checked that -- is not replaced with – char?

Eisbaer2 · February 4, 2024, 11:53am

Hello, I’m quite a Newbie with HA. At my front door I have a Doorline DECT connected via DECT to my fritzbox. With that solution I am able to talk via Fritzfon to my door/doorline and to open the door. The Doorline acts as an additional telephone connected to the fritzbox.

What I want to have additionally:

Possibility to see in HA when somebody rings (to trigger further actions like a voice message on an echo)
If somebody rings you can open the door by accepting the call and send a code like “#9”. I want to do this via a switch in HA.

Is anything of that possible with HA? If yes, how?

sdesalve · February 4, 2024, 12:35pm

It’s possible, but not with my addon. This addon is meant to place calls.

There is some python library to invoke pjsip and programmatically manage incoming calls. But it needs some skill to use it