Have you copied and pasted your addon config?
check that you have used standard double quotes
"
and simple minus
-
chars.
–no-tcp option is written as: --no-tcp
no copy paste. as written, it is working fine.
but If I remove the hash from the code below the addon is crashing with the above error in the moment when I trigger the phonecall via automation (which was all fine before)
tts:
- platform: google_translate
service_name: google_translate_say
language: 'de'
cache: true
cache_dir: /config/tts
time_memory: 300
# base_url: "http://HOMEASSISTANT-IP:8123"
there was is no change in addon config:
sip_parameters:
caller_id_uri: sip:[email protected]:5060
realm: "*"
username: username
password: password
pjsua_custom_options: "--ip-addr=HOMEASSISTANT-IP"
max_call_time: 20
Ok leave this line commented
Have you disabled Fritz box VoIP LAN lock?
If you click here, you can hear your TTS, right?
fritzbox Voip-LAN is locked. there is no reason to disable it from my point of view.
yes the mp3 is there, and available and i can hear what i expect
Here, before base_url option deprecation, I have my FQN url (eg: https://sssss.duckdns.org)
Prevent the use of Internet telephony from the local network
IP phones and applications cannot receive and make calls from the local network. This option activates a filter for outgoing SIP packets in the FRITZ!Box, in particular to protect against malware. Telephony devices configured on the FRITZ!Box can continue to be used without restrictions.
If you leave this option ticked, It will not work. Sorry
Let me know if you can solve your issue without this mandatory config
ah. this is the error
this was a misunderstanding.
the option I was mentioned which is not needed was this
it means that it is not allowed for this IP-Phone(user) to connect from internet to the fritzbox.
but your option was still missing, even if I do not understand why it was possible that the call itself was working but not to transfer the “voice”-data… thanks for the hint, I will try asap
your option was not recognized. If you had place here your full addon log (as requested) I’ve recognized the mistake early. sorry
great, it works. thanks a lot
I have read the entire thread and I have not been able to configure the addon:
CONFIG:
sip_parameters:
caller_id_uri: sip:[email protected]
realm: "*"
username: xxxxxxx
password: xxxxxxx
pjsua_custom_options: "--no-tcp"
max_call_time: 30
SCRIPT:
alias: call_nicojmb_phone
sequence:
- service: hassio.addon_stdin
data_template:
addon: 89275b70_dss_voip
input:
call_sip_uri: sip:[email protected]
message_tts: Test of message
mode: single
icon: mdi:phone
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 11.2 (amd64 / qemux86-64)
Home Assistant Core: 2023.12.0
Home Assistant Supervisor: 2023.11.6
-----------------------------------------------------------
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'
MAX_CALL_TIME = '30'
[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://192.168.86.39:8123/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '30' seconds.
16:25:48.324 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
16:25:48.325 sip_endpoint.c .Creating endpoint instance...
16:25:48.326 pjlib .select() I/O Queue created (0x7f31f5c75100)
16:25:48.326 sip_endpoint.c .Module "mod-msg-print" registered
16:25:48.326 sip_transport.c .Transport manager created.
16:25:48.326 pjsua_core.c .PJSUA state changed: NULL --> CREATED
16:25:48.343 pjsua_core.c .pjsua version 2.11.1 for Linux-6.1.63/x86_64 initialized
16:25:48.346 pjsua_app.c .Turning sound device -99 -99 ON
16:25:48.346 main.c Ready: Success
16:25:48.372 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.3:5060>: 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]
>>> 16:25:49.346 pjsua_app.c .Turning sound device -99 -99 OFF
16:26:18.313 pjsua_app.c .Call 0 is DISCONNECTED [reason=487 (Request Terminated)]
>>> 16:26:19.852 timer.c .Dumping timer heap:
16:26:19.852 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...
@sdesalve can you helpme?
1 Like
Hi,
Thank @sdesalve , but we tried with all of possibilities and do not work.
Just configured microsip and work fine with this parameters:
I don’t know what else to look at, here is my log:
[Info] Call ended...
[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://192.168.86.39:8123/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '30' seconds.
21:24:10.347 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
21:24:10.348 sip_endpoint.c .Creating endpoint instance...
21:24:10.349 pjlib .select() I/O Queue created (0x7f6b473ee100)
21:24:10.349 sip_endpoint.c .Module "mod-msg-print" registered
21:24:10.349 sip_transport.c .Transport manager created.
21:24:10.349 pjsua_core.c .PJSUA state changed: NULL --> CREATED
21:24:10.366 pjsua_core.c .pjsua version 2.11.1 for Linux-6.1.63/x86_64 initialized
21:24:10.370 pjsua_app.c .Turning sound device -99 -99 ON
21:24:10.370 main.c Ready: Success
21:24:10.396 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.3:5060>: 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]
>>> 21:24:11.371 pjsua_app.c .Turning sound device -99 -99 OFF
21:24:40.337 pjsua_app.c .Call 0 is DISCONNECTED [reason=487 (Request Terminated)]
>>> 21:24:41.872 timer.c .Dumping timer heap:
21:24:41.872 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...```Full addon logs…
If you make mistakes on your addon config (as I think) and you putted pjsua options in sip_setting field I can know this reading FULL addon logs, from start to call end
Hi @sdesalve,
If i put the paramenter mentioned in github thread, i’ve get 487 error (Request Terminated).
CONFIG
sip_parameters:
caller_id_uri: sip:[email protected]
realm: "*"
username: xxxxxxxxx
password: xxxxxxxxx
max_call_time: 30
pjsua_custom_options: "--no-tcp --proxy=sip:myurlof3cx.my3cx.es:5060;lr"
LOG
[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: 4.0.0
You are running the latest version of this add-on.
System: Home Assistant OS 11.2 (amd64 / qemux86-64)
Home Assistant Core: 2023.12.0
Home Assistant Supervisor: 2023.11.6
-----------------------------------------------------------
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:myurlof3cx.my3cx.es:5060;lr'
MAX_CALL_TIME = '30'
[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://192.168.86.39:8123/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '30' seconds.
21:39:26.501 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
21:39:26.502 sip_endpoint.c .Creating endpoint instance...
21:39:26.502 pjlib .select() I/O Queue created (0x7f1f1aca2100)
21:39:26.502 sip_endpoint.c .Module "mod-msg-print" registered
21:39:26.503 sip_transport.c .Transport manager created.
21:39:26.503 pjsua_core.c .PJSUA state changed: NULL --> CREATED
21:39:26.521 pjsua_core.c .pjsua version 2.11.1 for Linux-6.1.63/x86_64 initialized
21:39:26.525 pjsua_app.c .Turning sound device -99 -99 ON
21:39:26.525 main.c Ready: Success
21:39:26.574 tsx0x7f1f1ab10c28 ....Temporary failure in sending Request msg INVITE/cseq=28300 (tdta0x7f1f1ab0aab8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
21:39:26.574 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.3:5060>: 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]
>>> 21:39:27.525 pjsua_app.c .Turning sound device -99 -99 OFF
21:39:56.490 pjsua_app.c .Call 0 is DISCONNECTED [reason=487 (Request Terminated)]
>>> 21:39:58.028 timer.c .Dumping timer heap:
21:39:58.028 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...```Ok, pjsua options was correctly recognised
But are you trying to call the same extension of hassio? 41?
It’s removed for security…
I realized that when I add the parameters, I have this error:
21:55:03.687 tsx0x7f8d3b58ec28 ....Temporary failure in sending Request msg INVITE/cseq=12394 (tdta0x7f8d3b588ab8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
LOG
[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: 4.0.0
You are running the latest version of this add-on.
System: Home Assistant OS 11.2 (amd64 / qemux86-64)
Home Assistant Core: 2023.12.0
Home Assistant Supervisor: 2023.11.6
-----------------------------------------------------------
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:myurlof3cx.my3cx.es:5060;lr'
MAX_CALL_TIME = '30'
[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://192.168.86.39:8123/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '30' seconds.
21:55:03.659 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
21:55:03.660 sip_endpoint.c .Creating endpoint instance...
21:55:03.660 pjlib .select() I/O Queue created (0x7f8d3b720100)
21:55:03.660 sip_endpoint.c .Module "mod-msg-print" registered
21:55:03.660 sip_transport.c .Transport manager created.
21:55:03.660 pjsua_core.c .PJSUA state changed: NULL --> CREATED
21:55:03.679 pjsua_core.c .pjsua version 2.11.1 for Linux-6.1.63/x86_64 initialized
21:55:03.684 pjsua_app.c .Turning sound device -99 -99 ON
21:55:03.684 main.c Ready: Success
21:55:03.687 tsx0x7f8d3b58ec28 ....Temporary failure in sending Request msg INVITE/cseq=12394 (tdta0x7f8d3b588ab8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
21:55:03.687 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.3:5060>: 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]
>>> 21:55:04.683 pjsua_app.c .Turning sound device -99 -99 OFF
21:55:33.647 pjsua_app.c .Call 0 is DISCONNECTED [reason=487 (Request Terminated)]
>>> 21:55:35.186 timer.c .Dumping timer heap:
21:55:35.186 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...
You cannot call your same extension. But if you have edited this log for your privacy and it’s not same extension it’s ok.
Try to remove this
Yessss, removing “–no-tcp” works!:
Here is my config for 3CX
sip_parameters:
caller_id_uri: sip:extension_number@host
realm: "*"
username: xxxxxxxxx
password: xxxxxxxxxxxxxx
pjsua_custom_options: "--proxy=sip:host:port;lr"
Thanks so much @sdesalve !!
1 Like