Sorry,
here are the logs out of the share folder:
[Info] Received messages '{"call_sip_uri":"[email protected]:5060","message_tts":"TESTANRUF"}'
CALL_SIP_URI_VALUE = '[email protected]:5060'
MESSAGE_TTS_VALUE = 'TESTANRUF'
DATA_JSON = '{
"message": "TESTANRUF",
"platform": "google_translate"
}'
JSONGOOGLETTS = '500: Internal Server Error'
Thanks
Merc
as expected…
have you TTS in your hassio setup as stated in docs and in start guide?
ok, I have changed. Now I made the call like this:
action: hassio.addon_stdin
data:
addon: 89275b70_dss_voip
input:
call_sip_uri: [email protected]:9060
message_tts: Esto es una prueba
and this is the result:
[Info] Received messages {"call_sip_uri":"[email protected]:9060","message_tts":"Esto es una prueba"}
Converting audio file 'https://yyyyyyyyyyyyyyyyy:tttttttt/api/tts_proxy/nRDr-4gVFifE6W5jLx2aQA.mp3'...
Audio succesfully converted...
Starting SIP Client and calling '[email protected]:9060'...
This call will be terminated after '50' seconds.
01:47:09.887 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
01:47:09.887 sip_endpoint.c .Creating endpoint instance...
01:47:09.888 pjlib .select() I/O Queue created (0x7f2183b1c100)
01:47:09.888 sip_endpoint.c .Module "mod-msg-print" registered
01:47:09.888 sip_transport.c .Transport manager created.
01:47:09.888 pjsua_core.c .PJSUA state changed: NULL --> CREATED
01:47:09.888 pjsua_app_config.c Argument "zzzzzzzzzzzzzzz" is not valid. Use --help to see help
01:47:09.888 pjsua_core.c Shutting down, flags=0...
01:47:09.888 pjsua_core.c PJSUA state changed: CREATED --> CLOSING
01:47:09.888 pjsua_call.c .Hangup all calls..
01:47:09.888 pjsua_media.c .Call 0: deinitializing media..
01:47:09.888 pjsua_media.c .Call 1: deinitializing media..
01:47:09.888 pjsua_media.c .Call 2: deinitializing media..
01:47:09.888 pjsua_media.c .Call 3: deinitializing media..
01:47:09.888 pjsua_pres.c .Shutting down presence..
01:47:10.894 pjsua_core.c .Destroying...
01:47:10.894 pjsua_media.c .Shutting down media..
01:47:10.894 sip_endpoint.c .Destroying endpoint instance..
01:47:10.894 sip_endpoint.c .Module "mod-msg-print" unregistered
01:47:10.894 sip_transport.c .Destroying transport manager
01:47:10.895 timer.c .Dumping timer heap:
01:47:10.895 timer.c . Cur size: 0 entries, max: 3070
01:47:10.895 sip_endpoint.c .Endpoint 0x7f2183b610e8 destroyed
01:47:10.895 pjsua_core.c .PJSUA state changed: CLOSING --> NULL
01:47:10.895 pjsua_core.c .PJSUA destroyed...
./run: line 337: echo: write error: Broken pipe
./run: line 337: echo: write error: Broken pipe
[Error] pjsua Exit code: 1
[Info] Call ended...
in relation to the line
01:47:09.888 pjsua_app_config.c Argument "zzzzzzzzzzzzzzzz" is not valid. Use --help to see help
the parameter that gives error is the password of the configuration. the password that you put here:
sip_parameters:
caller_id_uri: sip:[email protected]:9060
realm: "*"
username: "eeeeeeeeeeeeeeee"
password: zzzzzzzzzzzzzzz
pjsua_custom_options: "-–no-tcp --outbound=sip:sip-eu.netelip.com:9060;lr"
but the password is correct according to the netelip platform. And I have tested it on an ATA CISCO SPA112 and the password is ok.
sip_parameters:
caller_id_uri: sip:[email protected]:9060
realm: "*"
username: "eeeeeeeeeeeeeeee"
password: "zzzzzzzzzzzzzzz"
pjsua_custom_options: "--no-tcp --outbound=sip:sip-eu.netelip.com:9060;lr"
put your password between simple double quotes"
custom options start with double minus signs –
your password can be correct and it can work with all other programs, but if you don’t write correctly yaml as a Linux program will expect them… it doesn’t will run and give you that self explaining error
sorry! i didn’t see that double minus sign mistake! thanks!
On the subject of the password I put it with double quotes but if the password contains numbers only the double quotes are kept. But if it contains any letter the double quotes are removed by themselves when saving the configuration.
[Info] Received messages {"call_sip_uri":"[email protected]:9060","message_tts":"Esto es una prueba"}
Converting audio file 'https://xxxxxxxxxxx:yyyy/api/tts_proxy/nRDr-4gVFifE6W5jLx2aQA.mp3'...
Audio succesfully converted...
Starting SIP Client and calling '[email protected]:9060'...
This call will be terminated after '50' seconds.
03:54:41.182 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
03:54:41.183 sip_endpoint.c .Creating endpoint instance...
03:54:41.183 pjlib .select() I/O Queue created (0x7fdecb343100)
03:54:41.183 sip_endpoint.c .Module "mod-msg-print" registered
03:54:41.183 sip_transport.c .Transport manager created.
03:54:41.183 pjsua_core.c .PJSUA state changed: NULL --> CREATED
03:54:41.184 pjsua_app_config.c Invalid SIP URI [email protected]:9060
03:54:41.184 pjsua_core.c Shutting down, flags=0...
03:54:41.184 pjsua_core.c PJSUA state changed: CREATED --> CLOSING
03:54:41.184 pjsua_call.c .Hangup all calls..
03:54:41.184 pjsua_media.c .Call 0: deinitializing media..
03:54:41.184 pjsua_media.c .Call 1: deinitializing media..
03:54:41.184 pjsua_media.c .Call 2: deinitializing media..
03:54:41.184 pjsua_media.c .Call 3: deinitializing media..
03:54:41.184 pjsua_pres.c .Shutting down presence..
03:54:42.186 pjsua_core.c .Destroying...
03:54:42.186 pjsua_media.c .Shutting down media..
03:54:42.186 sip_endpoint.c .Destroying endpoint instance..
03:54:42.186 sip_endpoint.c .Module "mod-msg-print" unregistered
03:54:42.186 sip_transport.c .Destroying transport manager
03:54:42.186 timer.c .Dumping timer heap:
03:54:42.186 timer.c . Cur size: 0 entries, max: 3070
03:54:42.187 sip_endpoint.c .Endpoint 0x7fdecb3880e8 destroyed
03:54:42.187 pjsua_core.c .PJSUA state changed: CLOSING --> NULL
03:54:42.187 pjsua_core.c .PJSUA destroyed...
./run: line 337: echo: write error: Broken pipe
./run: line 337: echo: write error: Broken pipe
[Error] pjsua Exit code: 1
[Info] Call ended...
sorry my mistake
please add sip: before
like I’ve stated before: if it don’t get input as expected, it will raise error…
I’ve told you to remove sip: I was wrong
ok, I test with SIP like this:
action: hassio.addon_stdin
data:
addon: 89275b70_dss_voip
input:
call_sip_uri: sip:[email protected]:9060
message_tts: Esto es una prueba
my log:
-----------------------------------------------------------
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 14.1 (amd64 / qemux86-64)
Home Assistant Core: 2024.12.4
Home Assistant Supervisor: 2024.12.0
-----------------------------------------------------------
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 --outbound=sip:sip-eu.netelip.com:9060;lr'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:[email protected]:9060","message_tts":"Esto es una prueba"}
Converting audio file 'https://xxxxxxxxxxxxxx:yyyy/api/tts_proxy/nRDr-4gVFifE6W5jLx2aQA.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]:9060'...
This call will be terminated after '50' seconds.
04:06:18.463 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
04:06:18.464 sip_endpoint.c .Creating endpoint instance...
04:06:18.464 pjlib .select() I/O Queue created (0x7f63db289100)
04:06:18.464 sip_endpoint.c .Module "mod-msg-print" registered
04:06:18.464 sip_transport.c .Transport manager created.
04:06:18.464 pjsua_core.c .PJSUA state changed: NULL --> CREATED
04:06:18.485 pjsua_core.c .pjsua version 2.11.1 for Linux-6.6.66/x86_64 initialized
04:06:18.489 pjsua_app.c .Turning sound device -99 -99 ON
04:06:18.489 main.c Ready: Success
04:06:18.492 tsx0x7f63db0f7be8 ....Temporary failure in sending Request msg INVITE/cseq=3751 (tdta0x7f63db0f1908), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
04:06:18.492 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.4:5060>: does not register
Online status: Online
*[ 1] sip:[email protected]:9060: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:[email protected]:9060
+=============================================================================+
| 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]:9060 [CALLING]
>>> 04:06:19.489 pjsua_app.c .Turning sound device -99 -99 OFF
04:06:50.492 pjsua_app.c ....Call 0 is DISCONNECTED [reason=408 (Request Timeout)]
No current call
>>> 04:07:10.001 timer.c .Dumping timer heap:
04:07:10.001 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...
try to remove --no-tcp
sip_parameters:
caller_id_uri: sip:[email protected]:9060
realm: "*"
username: "xxxxxxxxxxxx"
password: "xxxxxxxxxx"
pjsua_custom_options: "--outbound=sip:sip-eu.netelip.com:9060;lr"
not wotk…
-----------------------------------------------------------
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 14.1 (amd64 / qemux86-64)
Home Assistant Core: 2024.12.4
Home Assistant Supervisor: 2024.12.0
-----------------------------------------------------------
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 = '--outbound=sip:sip-eu.netelip.com:9060;lr'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:[email protected]:9060","message_tts":"Esto es una prueba"}
Converting audio file 'https://xxxxxxxxxxxx:yyyy/api/tts_proxy/nRDr-4gVFifE6W5jLx2aQA.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]:9060'...
This call will be terminated after '50' seconds.
04:17:35.654 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
04:17:35.654 sip_endpoint.c .Creating endpoint instance...
04:17:35.655 pjlib .select() I/O Queue created (0x7f707cb83100)
04:17:35.655 sip_endpoint.c .Module "mod-msg-print" registered
04:17:35.655 sip_transport.c .Transport manager created.
04:17:35.655 pjsua_core.c .PJSUA state changed: NULL --> CREATED
04:17:35.674 pjsua_core.c .pjsua version 2.11.1 for Linux-6.6.66/x86_64 initialized
04:17:35.679 pjsua_app.c .Turning sound device -99 -99 ON
04:17:35.679 main.c Ready: Success
04:17:35.683 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.4:5060>: does not register
Online status: Online
[ 1] <sip:172.30.33.4:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:[email protected]:9060: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:[email protected]:9060
+=============================================================================+
| 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]:9060 [CALLING]
>>> 04:17:35.707 tcpc0x7f707c9de388 !TCP connect() error: [code=120113]: Host is unreachable
04:17:35.707 tsx0x7f707c9f1be8 Temporary failure in sending Request msg INVITE/cseq=10871 (tdta0x7f707c9e9918), will try next server: Host is unreachable
04:17:35.708 pjsua_app.c SIP TCP transport is disconnected from 185.8.244.80:9060: Host is unreachable [status=120113]
04:17:36.679 pjsua_app.c .Turning sound device -99 -99 OFF
04:18:07.708 pjsua_app.c ....Call 0 is DISCONNECTED [reason=408 (Request Timeout)]
No current call
>>> 04:18:27.196 timer.c .Dumping timer heap:
04:18:27.196 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...```sorry
try to use other SIP providers
good luck
Hello everyone! I’m sorry to announce that I will no give more support for this addon.
If you can find answer to your issue in all this thread you’re welcome. Otherwise please find other solution to make VoIP calls
It was great, bye
Salvatore
thank you Salvatore it has been a pleasure. don’t you have anyone interested in continuing with your project? a fork of it and keep it on github? have you explained it and offered it to anyone?
Dear sdesalve,
after loads of going back and forth through this thread and picking the individual pieces required I managed to make your addon work.
It now makes the call and reads out the text.
This is amazing.
I just wanted to thank you for making this awesome addon available. And of course thanks as well for your extremely fast response and help.
Would be great if someone with the appropriate knowledge would be able to maintain this and keep it alive. Otherwise it would be a waste of all the time and energy you have invested.
Thanks again and merry X-mas.
Merc
thank you for your great work
I hope someone will continue your work and maintain the project
wish you the best
Hey all,
I’ve been using dss_voip addon for more than 1yr so far (with Eutelia/Orchestra VOIP provider in Italy) and I noticed that after a long time where it was running just fine, it recently started to fail.
In particular if I trigger a phone call, I get some generic “request timeout” error:
You have 1 active call
Current call id=0 to sip:<OMITTED>@voip.eutelia.it [CALLING]
>>> 00:39:30.739 pjsua_app.c .Turning sound device -99 -99 OFF
00:40:01.761 pjsua_app.c ....Call 0 is DISCONNECTED [reason=408 (Request Timeout)]
No current call
>>> 00:40:21.248 timer.c .Dumping timer heap:
00:40:21.248 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...
Do others noticed the same issue?
I’m exploring the idea of rolling my own HA addon (perhaps using “baresip” instead of “pjsua” as VOIP client)… but before that: is anybody aware of an HA alternative addon given that dss_voip has been abandoned?
thanks!
This could be a replacement, but the main problem of both of them is that there is not a direct control of the process.
I’m on the same crossway: develop my own component or try to continue to support dss…
thanks @crug80 , I actually tried to use ha-sip but somehow it does not work with my VOIP provider (I get a SIP 513 Message Too big as error, all details at SIP INVITE fails with SIP/2.0 513 Message too big · Issue #134 · arnonym/ha-plugins · GitHub).
I also investigated in the ha-sip source code, which uses pjsua under the hood (just like dss_voip), but I couldn’t find any obvious way to fix the issue.
Moreover I’m not getting any reply from ha-sip authors (above ticket is unanswered).
I tried to reach out to the pjsip project directly and my issue was closed without much discussion: Reduce SIP INVITE size from Python interface · Issue #4457 · pjsip/pjproject · GitHub
At this point I’ve been able to use baresip successfully with my VOIP provider and I will probably start my own addon based on that shortly…