Great add on !
Work perfectly with HA 2024.3 + nuc + remote 3cx
1 Like
Do I understand correctly the message will just keep looping? No way to get it just to play once when the call is answered?
(Why? Because I was hoping to use this with intercom/paging)
yes but you can set this
https://github.com/sdesalve/hassio-addons/tree/master/dss_voip#option-max_call_time-optional
or this
https://github.com/sdesalve/hassio-addons/tree/master/dss_voip#option-call_duration-optional
I have problem with my dss voip notifier arm
[Info] Received messages {"call_sip_uri":"sip:[email protected]","message_tts":"abc"}
Converting audio file 'http://192.168.178.36:8123/api/tts_proxy/a9993e364706816aba3e25717850c26c9cd0d89d_it_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '50' seconds.
20:45:59.672 os_core_unix.c !pjlib 2.9 for POSIX initialized
20:45:59.676 sip_endpoint.c .Creating endpoint instance...
20:45:59.677 pjlib .select() I/O Queue created (0x55b0dc1c90)
20:45:59.677 sip_endpoint.c .Module "mod-msg-print" registered
20:45:59.677 sip_transport.c .Transport manager created.
20:45:59.677 pjsua_core.c .PJSUA state changed: NULL --> CREATED
20:45:59.722 pjsua_core.c .pjsua version 2.9 for Linux-6.1.73/aarch64 initialized
20:45:59.726 pjsua_app.c .Turning sound device -99 -99 ON
20:45:59.727 main.c Ready: Success
20:45:59.740 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:192.168.178.36:5060>: does not register
Online status: Online
[ 1] <sip:192.168.178.36:5060;transport=TCP>: does not register
Online status: Online
*[ 2] 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]
>>> 20:46:00.726 pjsua_app.c .Turning sound device -99 -99 OFF
20:46:31.740 pjsua_app.c ....Call 0 is DISCONNECTED [reason=408 (Request Timeout)]
20:46:31.740 pjsua_app_common.c ....
[DISCONNCTD] To: sip:[email protected]
Call time: 00h:00m:00s, 1st res in 32013 ms, conn in 0ms
No current call
[Info] Call ended...
[Info] Listening for messages via stdin service call...
Full addon log?
Addon config?
Have you disabled Fritz VoIP filter?
You are italian? My english languagebits vwry bad 
sip_parameters:
caller_id_uri: sip:**[email protected]
realm: "*"
username: myuser
password: mypsw
pjsua_custom_options: "--ip-addr=192.168.178.36"
Mine It’s worst… 
But here we need to write in English
(At least, I try to write something that seems English…)
Ok useró google translate:
I configured the addon and script as indicated in the guide.
I think the problem is in the router, but I don’t receive the call on my smartphone.
I created a voip number on fritzbox with username and password but nothing happens. What can it be? Would you be kind to help me?
I would like to use your service to notify me that the house alarm goes off.
Do you have a telegram contact where you can help me so I can solve the problem and then post the solution on the forum?
My telegram contact is @PakyITA
Please check private messages inbox
Setting -1 for max_call_time did the trick! 
1 Like
Thanks for this, my supervisor watchdog keeps finding that the addon has failed, is it something you’re aware of? 
If not the watchdog catching the failed state, I get this error:
[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...
MAX_CALL_TIME = '-1'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:[redacted]@[redacted]","message_tts":"Hello"}
Converting audio file 'https://[redacted].org:8123/api/tts_proxy/0f7689264bc63373cb9727382a2241632bc1c3d4_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:sip:[redacted]@[redacted]'...
This call will be terminated after '3.18' seconds.
20:59:57.103 os_core_unix.c !pjlib 2.9 for POSIX initialized
./run: line 337: echo: write error: Broken pipe
./run: line 337: echo: write error: Broken pipe
./run: line 337: 413 Exit 1 ( sleep $MAX_CALL_TIME_VALUE; echo h; sleep 0.5; echo q )
414 Segmentation fault (core dumped) | ( pjsua --app-log-level=3 --config-file '/share/dss_voiparm/dss_pjsua.conf' $CALL_SIP_URI_VALUE 2> /share/dss_voiparm/dss_pjsua.log )
[Error] pjsua Exit code: 139
[Info] Call ended...
[Info] Listening for messages via stdin service call...
Sometimes it works though!
In the end I succeeded thanks to a user. It works perfectly! In the caller_id_uri I put the username I created when activating the voip on the fritz and not the voip number so the syntax is sip:username@ipfritzbox:5060
Please rephrase in English so everyone can follow the conversation
2 Likes
Sometimes pjsua will raise a segmentation fault. Seems a bug on the bin.
I’ve already searched and google is full for query pjsua segmentation fault, but I haven’t found any solution…
Try to leave some time between each call
Hello and congratulations on this fantastic add-on! I was trying to get it to work to manage the house alarm but unfortunately I can’t receive any calls. I use the add-on with the Irideos-Orchestra VoIP service, I followed the specific guide but was unable to make it work. My configuration is Hassio.os+proxmox+mini pc. The addo-on installs correctly but unfortunately crashes while making the call. I attach the logs to see if you have any advice to give me.
A thousand thanks
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:+3934733476**@voip.eutelia.it","message_tts":"Prova messaggio"}
Converting audio file 'http://192.168.1.90:8123/api/tts_proxy/772c970523ed7389b60b2ed1b674b3705d965e37_it_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:+3934733476**@voip.eutelia.it'...
This call will be terminated after '50' seconds.
23:54:37.704 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
23:54:37.705 sip_endpoint.c .Creating endpoint instance...
23:54:37.705 pjlib .select() I/O Queue created (0x7fc7ab008100)
23:54:37.705 sip_endpoint.c .Module "mod-msg-print" registered
23:54:37.705 sip_transport.c .Transport manager created.
23:54:37.705 pjsua_core.c .PJSUA state changed: NULL --> CREATED
23:54:37.713 pjsua_core.c .pjsua version 2.11.1 for Linux-6.6.20/x86_64 initialized
23:54:37.714 pjsua_app.c .Turning sound device -99 -99 ON
23:54:37.714 main.c Ready: Success
23:54:37.716 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.8:5060>: does not register
Online status: Online
[ 1] <sip:172.30.33.8:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:04186277**@voip.eutelia.it: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:+3934733476**@voip.eutelia.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:+3934733476**@voip.eutelia.it [CALLING]
>>> 23:54:38.715 pjsua_app.c .Turning sound device -99 -99 OFF
23:54:44.998 tcpc0x7fc7aae61538 TCP connect() error: [code=120113]: Host is unreachable
23:54:44.998 tsx0x7fc7aae776e8 Temporary failure in sending Request msg INVITE/cseq=17067 (tdta0x7fc7aae6eaa8), will try next server: Host is unreachable
23:54:44.998 pjsua_app.c SIP TCP transport is disconnected from 83.211.227.21:5060: Host is unreachable [status=120113]
23:55:16.999 pjsua_app.c ....Call 0 is DISCONNECTED [reason=408 (Request Timeout)]
No current call
>>> 23:55:29.236 timer.c .Dumping timer heap:
23:55:29.236 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...
Questa è la mia configurazione dell’add-on
sip_parameters:
caller_id_uri: sip:04186277**@voip.eutelia.it
realm: "*"
username: "04186277**"
password: px************
Add
--no-tcp
Option. See above