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sdesalve / hassio-addons

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Not really a bug, better a question #67

Closed kangaroo72 closed 1 year ago

kangaroo72 commented 1 year ago

Hi all,

it's an amazing addon, but sadly I'm unable to get it working. Does anyone has a config with Provider "Deutsche Telekom / Germany" to dial? Thanks a lot

sdesalve commented 1 year ago

Do you have parameters for your provider? Have you tried to use them successfully with other sip clients?

What kind of error do you get? Could you please post here

Your config Full addon logs

kangaroo72 commented 1 year ago

Hi and thanks for reply. Yeah - the data are working. I'm not an expert with these configs, so I think that's the problem...

Here my config

Here my automation (not sure if right at all...)

Here my log.

Thanks so much,

-Tom-

sdesalve commented 1 year ago

Screenshot_2023-01-31-09-53-00-139_io homeassistant companion android

   ...
    - service: hassio.addon_stdin
      data_template:
        addon: 89275b70_dss_voip
        input: {"call_sip_uri":"sip:+393334455667@sipserver.com","message_tts":"Write here your message"}
   ...

Disable Fritz box sip firewall

kangaroo72 commented 1 year ago

what's 192.168.178.136 in your setup? Home Assistant?

kangaroo72 commented 1 year ago

This is my actual log-file... it seems to be calling, but it's not ringing on my phone...

Link

sdesalve commented 1 year ago

what's 192.168.178.136 in your setup? Home Assistant?

Yes

sdesalve commented 1 year ago

Disable Fritz box sip firewall

eaf899c98666e945c885d18606e907d9935c5c32

kangaroo72 commented 1 year ago

It's already disabled... image

sdesalve commented 1 year ago

Full logs?

kangaroo72 commented 1 year ago 

[Info] Starting addon...
PJSUA_CUSTOM_OPTIONS = '--no-tcp --ip-addr=192.168.200.6'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:+xxxxxxxxxxx@fritz.box:5060","message_tts":"Write here your message"}
Converting audio file 'http://192.168.200.6:8123/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:+4916097627100@fritz.box:5060'...
This call will be terminated after '50' seconds.
06:06:38.262 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
12:34:54.263 sip_endpoint.c  .Creating endpoint instance...
19:03:10.264          pjlib  .select() I/O Queue created (0xb66f30c8)
19:03:10.264 sip_endpoint.c  .Module "mod-msg-print" registered
19:03:10.264 sip_transport.  .Transport manager created.
19:03:10.264   pjsua_core.c  .PJSUA state changed: NULL --> CREATED
20:18:06.290   pjsua_core.c  .pjsua version 2.11.1 for Linux-5.15.84/armv7l initialized
13:00:46.300    pjsua_app.c  .Turning sound device -99 -99 ON
19:29:02.301         main.c  Ready: Success
10:18:38.307    pjsua_app.c  .......Call 0 state changed to CALLING
>>>>
Account list:
  [ 0] <sip:192.168.200.6:5060>: does not register
       Online status: Online
 *[ 1] sip:hass-ipp@fritz.box:5060: does not register
       Online status: Online
Buddy list:
 [ 1] <?>  sip:+xxxxxxxxxxx@fritz.box:5060
+=============================================================================+
|       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:+xxxxxxxxxxx@fritz.box:5060 [CALLING]```
sdesalve commented 1 year ago

After that?

kangaroo72 commented 1 year ago

That's all logging

sdesalve commented 1 year ago

You have 1 active call Current call id=0 to sip:+xxxxxxxxxxx@fritz.box:5060 [CALLING]```

after this must be call logs and then

[Info] Listening for messages via stdin service call...

or addon crash logs

sdesalve commented 1 year ago

username: hass-ipp-user password: hass-ipp-pass

this are login data for a IP phone created in Telephony > Phone devices > add ip phone?

image

kangaroo72 commented 1 year ago

I will test other credentials....

kangaroo72 commented 1 year ago

tried credentials and name w/o special-characters - same problem - no further log.

sdesalve commented 1 year ago

post a screenshot of this page of your fritz

image

sdesalve commented 1 year ago

is this >

+49xxxxxxxxxx100 another phone number? or your fritz mobile number?

kangaroo72 commented 1 year ago

image

the number xxx100 is my smartphone

sdesalve commented 1 year ago

if you install microsip.org on your computer, could successfully place a call with same credential and same config?

kangaroo72 commented 1 year ago

Will try later - thanks for now - I'll update later

sdesalve commented 1 year ago

image

here I have my Voip phone landline number because it's my fritz that will register on my provider and act as a PBX. Is your fritz configured to allow call?

kangaroo72 commented 1 year ago

Yeah - I'm using the C6's below for outgoing Call's and the Yealink T54W

sdesalve commented 1 year ago

ok. please install microsip.org and use same log in data to place a call

bru-u-kno commented 1 year ago

Below my working config for FRITZ!Box 7590 with latest FRITZ!OS 7.50, HA on Debian Bullseye virtual machine, tts via google translate and Node-RED

FRITZ!Box

grafik

grafik

HA > /config/configuration.yaml

grafik

DSS VoIP Notifier Add-On

grafik

Node-RED

grafik

grafik

data > call service JSON-Editor

{ "addon": "89275b70_dss_voip", "input": { "call_sip_uri": "sip:+49358@fritz.box:5060", "message_tts": "Thanks for this great add-on" } }

kangaroo72 commented 1 year ago

Holy... it works!! Woohooooo. The usual thing... RTFM... Read the f*** manual....

I had caller_id_uri: 'sip:username@sipserver.com' But not with the username as mentioned, but the title of the virtual device... FACEPALM

Thanks so much for your patience.