I have cleaned the docker images but now I get this message during install:
The command ‘/bin/ash -o pipefail -c apk add --no-cache coreutils=8.32-r2 wget=1.21.1-r1 sox pjsua screen bc’ returned a non-zero code: 6
Raspberry OS is Raspbian GNU/Linux 9 (stretch)
Docker version 19.03.14
HA core-2021.1.5 / supervisor-2021.01.7
Last week another user have experienced same issue but, in his case, was a connection problem.
please connect with a SSH addon (https://github.com/home-assistant/hassio-addons/tree/master/ssh), run this commands and post results:
It doesn’t solve the issue. I think it’s an version problem?
amazing work sdesalve!!, im trying to configure with spanish main provider, Telefonica Spain, unsucessfully
my computer sip clients works using the following account data, and also a screenshot of my HA addon conf and service call:
i have tried several combinations, with port, no port, any suggest?
thanks!
On my post i haven’t writed version numbers… Do you have copied and pasted my commands?
Please post full terminal output…
yes, adding the proxy with this command it appears to call, but my mobile phone does not rings
added in conf:
pjsua_custom_options: ‘–outbound=sip:10.31.255.134:5070;lr’
Full logs. I can’t help you in this way
Ok which log do you need? entire one o can i capture a part?
dss_autoanswer.log ?
thanks
full addon log
also try to call your number to check if addon answer to calls and you can hear sounds
[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: 3.5.5
You are running the latest version of this add-on.
System: Debian GNU/Linux 10 (buster) (amd64 / qemux86-64)
Home Assistant Core: 2021.1.5
Home Assistant Supervisor: 2021.01.7
-----------------------------------------------------------
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:10.31.255.134:5070;lr'
[Info] Registering as SIP Client...
PJSUA_CUSTOM_OPTIONS = '--no-tcp --proxy=sip:10.31.255.134:5070;lr'
-----------------------------------------------------------
SIP Client registered.
Call <sip:[email protected]:5060>/VoIP phone number
to check system status.
You'll find logs in /share/dss_voip/dss_autoanswer.log
-----------------------------------------------------------
[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.X.XXX:XXXX/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_es_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '50' seconds.
17:19:30.468 os_core_unix.c !pjlib 2.9 for POSIX initialized
17:19:30.468 sip_endpoint.c .Creating endpoint instance...
17:19:30.469 pjlib .select() I/O Queue created (0x7fdb20a240f0)
17:19:30.469 sip_endpoint.c .Module "mod-msg-print" registered
17:19:30.469 sip_transport.c .Transport manager created.
17:19:30.469 pjsua_core.c .PJSUA state changed: NULL --> CREATED
17:19:30.488 pjsua_core.c .pjsua version 2.9 for Linux-4.19.0.13/x86_64 initialized
17:19:30.491 pjsua_app.c .Turning sound device -99 -99 ON
17:19:30.491 main.c Ready: Success
17:19:30.493 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.7:5060>: does not register
Online status: Online
*[ 1] sip:[email protected]:5060: 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]
>>> 17:19:30.531 pjsua_app.c .....Call 0 is DISCONNECTED [reason=403 (Forbidden)]
17:19:30.531 pjsua_app_common.c .....
[DISCONNCTD] To: sip:[email protected]
Call time: 00h:00m:00s, 1st res in 40 ms, conn in 0ms
17:19:31.491 pjsua_app.c .Turning sound device -99 -99 OFF
No current call
[Info] Call ended...
[Info] Listening for messages via stdin service call...
sip_parameters:
caller_id_uri: 'sip:[email protected]:5060'
realm: '*'
username: '111111111'
password: '111111111'
sip_server_uri: 'sip:telefonica.net:5060'
pjsua_custom_options: '--no-tcp --proxy=sip:10.31.255.134:5070;lr'
Also tried with : pjsua_custom_options: ‘–no-tcp --outbound=sip:10.31.255.134:5070;lr’
addon: 89275b70_dss_voip
input: {"call_sip_uri":"sip:[email protected]","message_tts":"Example Message"}
Other data:
County Code spain: +34
111111111 = Sip account number
22222222 = Destination call number
base_url in google tts setup in your configuration.yaml
for some reason during addon install your docker cannot resolve DNS for alpinelinux.org packages repository.
Please try to change DNS on your router to 8.8.8.8, reboot your router and also your HA host machine
disable also firewall/ad remover to try
i have hidden the numbers by XX, log says audio is created successfully, tts service works ok for me… do i have to add base_url ?
Nothing, same issue.
I have added base_url to tts in configuration.yaml
Log here:
[Info] Starting addon...
PJSUA_CUSTOM_OPTIONS = '--no-tcp --proxy=sip:10.31.255.134:5070;lr'
[Info] Registering as SIP Client...
PJSUA_CUSTOM_OPTIONS = '--no-tcp --proxy=sip:10.31.255.134:5070;lr'
-----------------------------------------------------------
SIP Client registered.
Call <sip:[email protected]>/VoIP phone number
to check system status.
You'll find logs in /share/dss_voip/dss_autoanswer.log
-----------------------------------------------------------
[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://192.168.1.18:8123/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_es_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '50' seconds.
18:34:12.941 os_core_unix.c !pjlib 2.9 for POSIX initialized
18:34:12.941 sip_endpoint.c .Creating endpoint instance...
18:34:12.942 pjlib .select() I/O Queue created (0x7fccbf2530f0)
18:34:12.942 sip_endpoint.c .Module "mod-msg-print" registered
18:34:12.942 sip_transport.c .Transport manager created.
18:34:12.942 pjsua_core.c .PJSUA state changed: NULL --> CREATED
18:34:12.962 pjsua_core.c .pjsua version 2.9 for Linux-4.19.0.13/x86_64 initialized
18:34:12.965 pjsua_app.c .Turning sound device -99 -99 ON
18:34:12.965 main.c Ready: Success
18:34:12.966 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.7:5060>: 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]
>>> 18:34:13.002 pjsua_app.c .....Call 0 is DISCONNECTED [reason=403 (Forbidden)]
18:34:13.002 pjsua_app_common.c .....
[DISCONNCTD] To: sip:[email protected]
Call time: 00h:00m:00s, 1st res in 37 ms, conn in 0ms
18:34:13.966 pjsua_app.c .Turning sound device -99 -99 OFF
No current call
[Info] Call ended...
[Info] Listening for messages via stdin service call...
error is [reason=403 (Forbidden)]… Is your login details correct?
also try with this
tried with outbound again, nothing…
login is correct, username = sipnumber password = sipnumber
