Strange. I have join the Beta Channel and update to core-2021.2.0b1 and supervisor-2021.01.8 .
Latest coreutils for me is this version:
So I get the same install error message as before.
try to execute command
apk add --nocache coreutils=8.32-r0
I guess it fails…
It will fail because you are wrong: you are trying to use apk add within a SSH shell but that command is for DOCKERFILE!!!
Unfortunately, I don’t know what else I can change. I was still able to install the earlier version of the VOIP plugin (3.4.1). I can also install other plug-ins from the add-on store without any problems. Perhaps further updates of the Docker file for my platform are necessary. Maybe it will work in the next few days. We will see.
You can use this previuos version. Code It’s the same but I’ve upgraded config.json, build.json e Dockerfile to meet new base addon template and Hassio requirements…
Yesterday my calls were successfull, today I’m struggeling with the following error message:
Starting SIP Client and calling ‘sip:[email protected]:5060’…
This call will be terminated after ‘50’ seconds.
11:12:51.471 os_core_unix.c !pjlib 2.9 for POSIX initialized
18:41:07.472 sip_endpoint.c .Creating endpoint instance…
01:09:23.473 pjlib .select() I/O Queue created (0xb66990b8)
01:09:23.473 sip_endpoint.c .Module “mod-msg-print” registered
01:09:23.473 sip_transport. .Transport manager created.
01:09:23.473 pjsua_core.c .PJSUA state changed: NULL → CREATED
06:52:35.500 pjsua_core.c .pjsua version 2.9 for Linux-5.4.79/armv7l initialized
22:42:11.506 pjsua_app.c .Turning sound device -99 -99 ON
22:42:11.506 main.c Ready: Success
> 18:32:35.650 tsx0xb6511094 …Failed to send Request msg INVITE/cseq=32072 (tdta0xb6514474)! err=70018 (gethostbyname() has returned error (PJ_ERESOLVE))
18:32:35.650 pjsua_app.c …Call 0 is DISCONNECTED [reason=502 (gethostbyname() has returned error (PJ_ERESOLVE))]
18:32:35.650 pjsua_app_comm …
[DISCONNCTD] To: sip:[email protected]
Call time: 00h:00m:00s, 1st res in 144 ms, conn in 0ms
Any idea what this means?
Thank you!!
Can you post your add-on’s configuration and service invocation?
And please… Full add-on’s logs
Thank you for your support. I just found the problem. It wasn’t HA. I restarted my fritz.box (including the SIP Server) and it works again.
On my Raspberry 3b+ with
Home Assistant OS 5.10
core-2021.2.1
supervisor-2021.01.7
I was able to upgrade from armv7-addon-dss_voip:3.4.2 to armv7-addon-dss_voip:3.5.5
I can’t figure how to help who is experiencing problems. Seems that only users on ARM (Raspberry) still experiencing problems.
Now I’ve upgraded addon to 3.5.6 for use base images 9.1.2… Let me know if you can upgrade
Hi sdesalve.
I can resume @Luisico’s work. I have got this working once, but no luck anymore. I don’t know why…
My configuration is:
sip_parameters:
caller_id_uri: 'sip:[email protected]'
realm: telefonica.net
username: 9xxxxxxxx
password: 9xxxxxxxx
sip_server_uri: 'sip:10.31.255.134:5070'
pjsua_custom_options: >-
--no-tcp --srtp-secure=0 --use-srtp=0 --proxy=sip:10.31.255.134:5070;lr
--no-vad --add-codec=PCMA/8000
The execution part is:
addon: 89275b70_dss_voip34
input: {"call_sip_uri":"sip:[email protected]","message_tts":"Esto es una prueba"}
If I call from my mobile phone to the DSS VoIP registered number, I can listen a voice that says: “DSS VoIP Notifier is running”.
The pjsua’s log is:
ALSA lib confmisc.c:767:(parse_card) cannot find card '0'
ALSA lib conf.c:4693:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory
ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings
ALSA lib conf.c:4693:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
ALSA lib confmisc.c:1246:(snd_func_refer) error evaluating name
ALSA lib conf.c:4693:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
ALSA lib conf.c:5181:(snd_config_expand) Evaluate error: No such file or directory
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM sysdefault
ALSA lib confmisc.c:767:(parse_card) cannot find card '0'
ALSA lib conf.c:4693:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory
ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings
ALSA lib conf.c:4693:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
ALSA lib confmisc.c:1246:(snd_func_refer) error evaluating name
ALSA lib conf.c:4693:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
ALSA lib conf.c:5181:(snd_config_expand) Evaluate error: No such file or directory
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM sysdefault
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.front
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround21
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround21
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround40
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround41
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround50
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround51
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround71
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.iec958
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.iec958
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.iec958
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.modem
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.modem
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline
ALSA lib confmisc.c:767:(parse_card) cannot find card '0'
ALSA lib conf.c:4693:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory
ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings
ALSA lib conf.c:4693:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
ALSA lib confmisc.c:1246:(snd_func_refer) error evaluating name
ALSA lib conf.c:4693:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
ALSA lib conf.c:5181:(snd_config_expand) Evaluate error: No such file or directory
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM default
ALSA lib confmisc.c:767:(parse_card) cannot find card '0'
ALSA lib conf.c:4693:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory
ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings
ALSA lib conf.c:4693:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
ALSA lib confmisc.c:1246:(snd_func_refer) error evaluating name
ALSA lib conf.c:4693:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
ALSA lib conf.c:5181:(snd_config_expand) Evaluate error: No such file or directory
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM default
ALSA lib confmisc.c:767:(parse_card) cannot find card '0'
ALSA lib conf.c:4693:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory
ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings
ALSA lib conf.c:4693:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
ALSA lib confmisc.c:1246:(snd_func_refer) error evaluating name
ALSA lib conf.c:4693:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
ALSA lib conf.c:5181:(snd_config_expand) Evaluate error: No such file or directory
ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM dmix
Cannot connect to server socket err = No such file or directory
Cannot connect to server request channel
jack server is not running or cannot be started
JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock
JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock
The addon’s log is:
[Info] Received messages {"call_sip_uri": "sip:[email protected]", "message_tts": "Esto es una prueba"}
Converting audio file 'http://192.168.33.30:8123/api/tts_proxy/a609295021744033e92fd3fb5c758fc7f19ed697_es_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '50' seconds.
>>> 15:08:26.077 os_core_unix.c !pjlib 2.9 for POSIX initialized
15:08:26.078 sip_endpoint.c .Creating endpoint instance...
15:08:26.078 pjlib .select() I/O Queue created (0x1b30e78)
15:08:26.078 sip_endpoint.c .Module "mod-msg-print" registered
15:08:26.078 sip_transport. .Transport manager created.
15:08:26.078 pjsua_core.c .PJSUA state changed: NULL --> CREATED
15:08:26.094 pjsua_core.c .pjsua version 2.9 for Linux-5.10.11/armv7l initialized
15:08:26.097 pjsua_app.c .Turning sound device -99 -99 ON
15:08:26.098 main.c Ready: Success
15:08:26.099 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.10:5060>: 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]
>>> 15:08:26.106 pjsua_app.c .....Call 0 is DISCONNECTED [reason=403 (Forbidden)]
15:08:26.106 pjsua_app_comm .....
[DISCONNCTD] To: sip:[email protected]
Call time: 00h:00m:00s, 1st res in 8 ms, conn in 0ms
15:08:27.097 pjsua_app.c .Turning sound device -99 -99 OFF
No current call
[Info] Call ended...
[Info] Listening for messages via stdin service call...
have you tried this solutions?
The first step has been tested without success. I havent tried the second step, i will do it!
[quote="[New Addon] DSS VoIP Notifier: Hassio makes phone calls to you! ️ Share your Projects!, post:256, topic:130993"]
telefonica.net
[/quote]
No luck 
This is not required. If you add this option add-on will be registered on sip server and answer to all incoming calls. Where come from that IP? Have you tried to use this ip instead server name telefonica.net?
Hey @sdesalve,
thank you so much for developing this awesome add-on and answering so many questions on this forum.
I have been struggling with setting it up with my FritzBox for a few days and I have tried pretty much everything. I read every single comment in this thread and on github but simply could not get it to work, even though I set everything up correctly. The phone would ring, but there was never any audio.
I was actually just about to ask for your help when I discovered the problem myself. My raspberry has two ip addresses for some reason (192.168.178.136 and 192.168.178.140). Both work when accessing the web interface, but only one of them works in the add-on config.
I thought, I’d still share the solution to my problem here in case someone else is having the same issue. Thanks again for sharing this project with everyone. Enjoy your coffee! 
1 Like
I think you have both WiFi connection and LAN cable… Remove from config.txt your wireless data
For everyone who wants to use this great addon within Node-RED:
Add a “call service”-node to your flow with the following settings:
- Domain: hassio
- Service: addon_stdin
- Data (JSON):
{
“addon”: “89275b70_dss_voip”,
“input”: {
“call_sip_uri”: “sip:<phone-number>@sip.freevoipdeal.com”,
“message_tts”: “This is a voice message and optionally you can use {{payload}} to add some dynamic text as input to this node.”
}
}
3 Likes
I don’t know if this is the right place to post any potential ideas, but any chance to extend the addon with a sensor that indicates if someone actually picked up the phone when making a call? This could be useful with alarm systems for example; when a phone number is called but not answered within x rings, the next one on the list will be called.
Is this even possible or is there another way of monitoring/solving this?
unfortunately it’s not possible to know when attendee pickup a call or not.
I use a PBX, pbxes.org to make a group destination that make ringing at same time all attendee
1 Like
Hello,
i hope so some peaple can help me.
I Have install:
Add-on version: 3.4.3
You are running the latest version of this add-on.
System: Home Assistant OS 5.12 (armv7 / raspberrypi3)
Home Assistant Core: 2021.3.4
Home Assistant Supervisor: 2021.03.4
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 = ‘–ip-addr=192.168.178.63’
[Info] Registering as SIP Client…
PJSUA_CUSTOM_OPTIONS = ‘–ip-addr=192.168.178.63’
SIP Client registered.
Call sip:[email protected]:5060/VoIP phone number
to check system status.
You’ll find logs in /share/dss_voip34/dss_autoanswer.log
[Info] Listening for messages via stdin service call…
…
- service: hassio.addon_stdin
data_template:
addon: 89275b70_dss_voip34
input: {“call_sip_uri":"sip:[email protected]”,“message_tts”:“Write here your message”}
…
But I dont understand to creat:
- service: hassio.addon_stdin
data_template:
addon: 89275b70_dss_voip34
input: {"call_sip_uri":"sip:[email protected]","message_tts":"Write here your message"}
Can me that explain with Picture ?.
Thanks