Thank you. Where could I find more information about your mentioned python library?
Hi guys!
First of all thanks to @sdesalve for the addon and, moreover, for the great supporto here.
I tried to setup the environment and to read the whole topic, but I don’t understand how to call the service to make a phone call to my mobile or if it is possible.
I’m using a Fritz, I’ve configured the IP phone on the interface and I’ve configured the addon and the tts (Picotts).
Just created a test script… but… What should I write as call_sip_uri for a mobile phone number?
Here the log of the addon generated after my first test
[Info] Starting addon...
PJSUA_CUSTOM_OPTIONS = '--ip-addr=192.168.1.10'
PLATFORM_TTS = 'picotts'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:[email protected]:5060","message_tts":"Ciao Sono Io!"}
Converting audio file 'http://192.168.1.10:8123/api/tts_proxy/aebc66490ae73df55e0d940189d10acab54a38d6_it-it_-_picotts.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]:5060'...
This call will be terminated after '50' seconds.
16:04:47.707 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
16:04:47.707 sip_endpoint.c .Creating endpoint instance...
16:04:47.707 pjlib .select() I/O Queue created (0x7fa2292100)
16:04:47.707 sip_endpoint.c .Module "mod-msg-print" registered
16:04:47.707 sip_transport.c .Transport manager created.
16:04:47.707 pjsua_core.c .PJSUA state changed: NULL --> CREATED
16:04:47.718 pjsua_core.c .pjsua version 2.11.1 for Linux-6.1.63/aarch64 initialized
16:04:47.718 pjsua_app.c .Turning sound device -99 -99 ON
16:04:47.718 main.c Ready: Success
16:04:47.719 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:192.168.1.10:5060>: does not register
Online status: Online
[ 1] <sip:192.168.1.10:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:**[email protected]:5060: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:[email protected]:5060
+=============================================================================+
| 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]:5060 [CALLING]
>>> 16:04:48.718 pjsua_app.c .Turning sound device -99 -99 OFF
16:05:19.719 pjsua_app.c ....Call 0 is DISCONNECTED [reason=408 (Request Timeout)]
No current call
>>> 16:05:39.225 timer.c .Dumping timer heap:
16:05:39.225 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
add
--no-tcp
Is your Fritz box’s IP 192.168.1.1?
Yes, 192.168.1.1 is the fritz and 10 is the rasp.
Here the addon config (after your suggestion for --no-tcp):
sip_parameters:
caller_id_uri: sip:**[email protected]:5060
realm: "*"
username: user
password: passUser
pjsua_custom_options: "--ip-addr=192.168.1.10 --no-tcp"
platform_tts: picotts
I tried also to replace **620 with username
Here the script:
test_chiamata:
alias: Test_Chiamata
sequence:
- service: hassio.addon_stdin
data:
addon: 89275b70_dss_voip
input:
call_sip_uri: "sip:[email protected]:5060"
message_tts: "Ciao Sono io"
mode: single
icon: mdi:cellphone
Have you used _ARM version? Can you try to switch beetween ARM version/standard version?
Hi @sdesalve,
thanks for the addon. I’m having issues with setting it up for my FritzBox.
After I start the addon and execute the service call, the addon stops before the call is started.
Im running hassio on a Raspberry Pi 4 and am using the ARM version of the addon. Fritzbox VoIP call filters are deactivated and I have setup an internal phone number for home assistant.
IP FritzBox: 192.168.178.1
IP Raspberry: 192.168.178.69
Addon config:
sip_parameters:
caller_id_uri: sip:[email protected]:5060
realm: "*"
username: homeassistant
password: xxxxx
pjsua_custom_options: "--ip-addr=192.168.178.69 --no-tcp"
Service call:
service: hassio.addon_stdin
data:
addon: 89275b70_dss_voiparm
input:
call_sip_uri: "sip:**[email protected]:5060"
audio_file_url: "http:192.168.178.69:8123/local/notification.mp3"
Addon logs
[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 for ARM
VoIP Notifier for Home Assistant (for ARM processors)
-----------------------------------------------------------
Add-on version: 4.1.0
You are running the latest version of this add-on.
System: Home Assistant OS 11.5 (aarch64 / raspberrypi4-64)
Home Assistant Core: 2024.2.1
Home Assistant Supervisor: 2024.01.1
-----------------------------------------------------------
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=192.168.178.69 --no-tcp'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:**[email protected]:5060","audio_file_url":"http:192.168.178.69:8123/local/notification.mp3"}
Converting audio file 'http:192.168.178.69:8123/local/notification.mp3'...
[cont-finish.d] executing container finish scripts...
[cont-finish.d] 99-message.sh: executing...
[cont-finish.d] 99-message.sh: exited 0.
[cont-finish.d] done.
[s6-finish] waiting for services.
[s6-finish] sending all processes the TERM signal.
**622 is the internal number I am trying to call. I have also tried with external phone numbers. The audio file url also works when opening it in a browser.
The SIP user and password are also correct. When I have tried with the sip_server_uri option, I was able to call the call the internal number of the addon to hear the confirmation message that it is running.
Anything else I can try? Thank you!
there is some issue with your TTS config…
please check your configuration.yaml
yes please…
@sdesalve Almost the same log also with ARM version
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]:5060'...
This call will be terminated after '50' seconds.
21:30:52.501 os_core_unix.c !pjlib 2.9 for POSIX initialized
21:30:52.501 sip_endpoint.c .Creating endpoint instance...
21:30:52.502 pjlib .select() I/O Queue created (0x557d40dc90)
21:30:52.502 sip_endpoint.c .Module "mod-msg-print" registered
21:30:52.502 sip_transport.c .Transport manager created.
21:30:52.502 pjsua_core.c .PJSUA state changed: NULL --> CREATED
21:30:52.511 pjsua_core.c .pjsua version 2.9 for Linux-6.1.63/aarch64 initialized
21:30:52.511 pjsua_app.c .Turning sound device -99 -99 ON
21:30:52.511 main.c Ready: Success
21:30:52.511 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:192.168.1.10:5060>: does not register
Online status: Online
*[ 1] sip:**[email protected]:5060: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:[email protected]:5060
+=============================================================================+
| 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]:5060 [CALLING]
>>> 21:30:53.511 pjsua_app.c .Turning sound device -99 -99 OFF
21:31:24.511 pjsua_app.c ....Call 0 is DISCONNECTED [reason=408 (Request Timeout)]
21:31:24.511 pjsua_app_common.c ....
[DISCONNCTD] To: sip:[email protected]
Call time: 00h:00m:00s, 1st res in 32000 ms, conn in 0ms
No current call
[Info] Call ended...
[Info] Listening for messages via stdin service call...
P.S. it’s a rasp 5, but I don’t if it could be the problem
!!!
you have a typo!!!
http:**//**192.168.178.69:8123/local/notification.mp3
Nothing to do… I see the request sniffing the traffic on my fritz, but same error. @sdesalve any idea?
Could be useful to analyze the sniffing log of the fritz?
Seems that your config it’s not correct
sip_parameters:
caller_id_uri: sip:[email protected]:5060
realm: "*"
username: user
password: passUser
pjsua_custom_options: "--ip-addr=192.168.1.10 --no-tcp"
platform_tts: picotts
Also
I don’t know how, but the call is started… The only change I did is to increase the lenght of the username…
The problem now is that the call says wrong thing, It seems an error of the text To speech.
@sdesalve Here the new log:
PJSUA_CUSTOM_OPTIONS = '--ip-addr=192.168.1.10 --no-tcp'
PLATFORM_TTS = 'picotts'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:[email protected]:5060","message_tts":"Ciao Sono Io"}
Converting audio file 'http://192.168.1.10:8123/api/tts_proxy/aebc66490ae73df55e0d940189d10acab54a38d6_it-it_-_picotts.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]:5060'...
This call will be terminated after '50' seconds.
21:59:25.986 os_core_unix.c !pjlib 2.9 for POSIX initialized
21:59:25.986 sip_endpoint.c .Creating endpoint instance...
21:59:25.987 pjlib .select() I/O Queue created (0x55be111c90)
21:59:25.987 sip_endpoint.c .Module "mod-msg-print" registered
21:59:25.987 sip_transport.c .Transport manager created.
21:59:25.987 pjsua_core.c .PJSUA state changed: NULL --> CREATED
21:59:25.996 pjsua_core.c .pjsua version 2.9 for Linux-6.1.63/aarch64 initialized
21:59:25.996 pjsua_app.c .Turning sound device -99 -99 ON
21:59:25.996 main.c Ready: Success
21:59:25.996 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:192.168.1.10:5060>: does not register
Online status: Online
*[ 1] sip:[email protected]:5060: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:[email protected]:5060
+=============================================================================+
| 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]:5060 [CALLING]
>>> 21:59:26.006 tsx0x55be161aa8 .......Temporary failure in sending Request msg INVITE/cseq=2406 (tdta0x55be158da8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
21:59:26.038 pjsua_app.c .....Call 0 state changed to EARLY (183 Session Progress)
21:59:37.747 pjsua_app.c .....Call 0 state changed to CONNECTING
21:59:37.748 pjsua_app.c .....Call 0 state changed to CONFIRMED
>>> 22:00:15.982 pjsua_app.c .....Call 0 is DISCONNECTED [reason=200 (Normal call clearing)]
22:00:15.983 pjsua_app_common.c .....
[DISCONNCTD] To: sip:[email protected];tag=B8BDA82DABEF8ACE
Call time: 00h:00m:38s, 1st res in 42 ms, conn in 11752ms
#0 audio iLBC @8kHz, sendrecv, peer=192.168.1.1:7078
SRTP status: Not active Crypto-suite:
RX pt=104, last update:00h:00m:03.945s ago
total 963pkt 48.1KB (86.6KB +IP hdr) @avg=10.0Kbps/18.1Kbps
pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 1.858 2.000 1.875 0.175
TX pt=104, ptime=30, last update:00h:00m:10.852s ago
total 1.0Kpkt 53.1KB (95.6KB +IP hdr) @avg=11.1Kbps/20.0Kbps
pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 8.125 9.725 10.125 10.125 0.800
RTT msec : 0.732 0.835 0.885 0.732 0.062
22:00:16.983 pjsua_app.c ..Turning sound device -99 -99 OFF
[Info] Call ended...
anyway, the configuration seems not stable. 10 tests, 5 failures
>>> 22:43:17.727 tsx0x55a2833aa8 .......Temporary failure in sending Request msg INVITE/cseq=21347 (tdta0x55a282ada8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
22:43:17.756 pjsua_app.c .....Call 0 state changed to EARLY (183 Session Progress)
>>> 22:44:07.706 pjsua_app.c .....Call 0 is DISCONNECTED [reason=487 (Request Terminated)]
22:44:07.706 pjsua_app_common.c .....
and I have to restart the component
Try Google TTS
On NUC it run well.
raspy has less computational power
1 Like
I don’t know if that is the problem. Monitoring the resouces, the CPU never exceeds 5% and RAM is lower than 2Gb of 8 available…