Yes, it’s checked
thank you for your great efforts and this amazing add-on.
i want to ask you can this add-on start a voice call between two peers not just playing audio file
Hi Mustafa,
I don’t think it’s possible. I think you must use a PBX and conference rooms. I use pbxes.com but I don’t know if it can help you.
I’ve created a thread on their forum. Please follow it and you’ll updated on my request
https://www1.pbxes.com/forum/thread.php?threadid=1625776132&sid=0bbc65dc09b2c2682e1d52765ebc083d
1 Like
Thanks for very usefull addon. I tring set call duration more then 120 sec (my mp3-file’s duration is 48 minutes). max_call_time: ‘-1’ in addon settings and call_duration: ‘-1’ in script are haven’t effect.
addon settings:
sip_parameters:
caller_id_uri: sip:[email protected]
sip_server_uri: sip:192.168.191.20
realm: '*'
username: '76304'
password: '123456'
pjsua_custom_options: >-
--no-tcp --srtp-secure=0 --use-srtp=0 --proxy=sip:192.168.191.20;lr --no-vad
--add-codec=PCMA/8000
max_call_time: '-1'
scripts.yaml:
sipprivet:
sequence:
- service: hassio.addon_stdin
data_template:
addon: 89275b70_dss_voip
input:
call_sip_uri: 'sip:[email protected]'
audio_file_url: 'http://192.168.201.233:8123/local/mp3/accr3_1.mp3'
call_duration: '-1'
addon log:
Add-on version: 3.5.7
You are running the latest version of this add-on.
System: Home Assistant OS 6.1 (amd64 / qemux86-64)
Home Assistant Core: 2021.7.0
Home Assistant Supervisor: 2021.06.8
-----------------------------------------------------------
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 --srtp-secure=0 --use-srtp=0 --proxy=sip:192.168.191.20;lr --no-vad --add-codec=PCMA/8000'
[Info] Registering as SIP Client...
PJSUA_CUSTOM_OPTIONS = '--no-tcp --srtp-secure=0 --use-srtp=0 --proxy=sip:192.168.191.20;lr --no-vad --add-codec=PCMA/8000'
-----------------------------------------------------------
SIP Client registered.
Call <sip:[email protected]>/VoIP phone number
to check system status.
You'll find logs in /share/dss_voip/dss_autoanswer.log
-----------------------------------------------------------
MAX_CALL_TIME = '-1'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri": "sip:[email protected]", "audio_file_url": "http://192.168.201.233:8123/local/mp3/accr3_1.mp3", "call_duration": "-1"}
CALL_DURATION = '-1'
Converting audio file 'http://192.168.201.233:8123/local/mp3/accr3_1.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '2978.91' seconds.
09:22:15.218 os_core_unix.c !pjlib 2.9 for POSIX initialized
09:22:15.222 sip_endpoint.c .Creating endpoint instance...
09:22:15.222 pjlib .select() I/O Queue created (0x7f0c294690f0)
09:22:15.222 sip_endpoint.c .Module "mod-msg-print" registered
09:22:15.222 sip_transport.c .Transport manager created.
09:22:15.222 pjsua_core.c .PJSUA state changed: NULL --> CREATED
09:22:15.235 pjsua_core.c .pjsua version 2.9 for Linux-5.10.45/x86_64 initialized
09:22:15.236 pjsua_app.c .Turning sound device -99 -99 ON
09:22:15.236 main.c Ready: Success
09:22:15.237 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.0:5060>: 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]
>>> 09:22:15.339 pjsua_app.c .....Call 0 state changed to EARLY (180 Ringing)
09:22:19.533 pjsua_app.c .....Call 0 state changed to CONNECTING
09:22:19.534 pjsua_app.c .....Call 0 state changed to CONFIRMED
09:24:19.534 pjsua_app.c !Duration (120 seconds) has been exceeded for call 0, disconnecting the call
09:24:19.578 pjsua_app.c .....Call 0 is DISCONNECTED [reason=200 (Normal call clearing)]
09:24:19.578 pjsua_app_common.c .....
[DISCONNCTD] To: sip:[email protected];tag=9a76dcadee95507e
Call time: 00h:02m:00s, 1st res in 103 ms, conn in 4298ms
#0 audio PCMA @8kHz, sendrecv, peer=192.168.191.20:50134
SRTP status: Not active Crypto-suite:
RX pt=8, last update:00h:00m:04.574s ago
total 5.9Kpkt 959.5KB (1.19MB +IP hdr) @avg=63.9Kbps/79.9Kbps
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 0.127 0.750 0.250 0.125
TX pt=8, ptime=20, last update:00h:00m:00.004s ago
total 6.0Kpkt 960.3KB (1.20MB +IP hdr) @avg=63.9Kbps/79.9Kbps
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 : 0.000 0.375 2.500 0.000 0.781
RTT msec : 50.399 173.640 372.087 372.087 54.894
Accept value between 10 and 120 seconds
hassio-addons/dss_voip at master · sdesalve/hassio-addons · GitHub
I’ve readed that max call time allowed is 120 seconds so I’ve hardcoded this limit in my addon.
I’ll remove this limit on a next update… Will you try and report me if duration allowed for a call can be greather than 120 secs?
Thank you very much! I will waiting next update and report result here surely.
1 Like
Excuse me, desalve. Can I ask you some questions. I’d like to terminate call by script and make multiple different calls at the same time. Is it possible those functionality in your addon?
No, it’s not possible pjsua calls cannot be controlled from HA. I suggest you to use an external PBX, try pbxes.com
bye
Install this addon and try it!
Be aware that addon id will change to 89275b70_dss_voip_test
sipprivet:
sequence:
- service: hassio.addon_stdin
data_template:
addon: 89275b70_dss_voip_test
input:
call_sip_uri: 'sip:[email protected]'
audio_file_url: 'http://192.168.201.233:8123/local/mp3/accr3_1.mp3'
call_duration: '-1'
Thanks! But, addon starting with error:
[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 - TEST VERSION - DO NOT USE IN PRODUCTION
VoIP Notifier for Home Assistant - TEST VERSION - DO NOT USE IN PRODUCTION
-----------------------------------------------------------
Add-on version: 0.0.1
You are running the latest version of this add-on.
System: Home Assistant OS 6.1 (amd64 / qemux86-64)
Home Assistant Core: 2021.7.0
Home Assistant Supervisor: 2021.06.8
-----------------------------------------------------------
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...
./run: line 178: syntax error near unexpected token `else'
[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.
Please try now
Hello sdesalve,
firstly: thank you for this great addon!
I already used calls on my Pi4 with freevoipdeal. Last week i changed to a X86-based NUC System and migrated my configuration. But since i moved to the new system, phone calls did not working anymore.
I tried to de- and install the addon again. Make the configuration new, but nothing helped.
Here is my config:
sip_parameters:
caller_id_uri: sip:[email protected]
realm: '*'
username: my.username
password: XXXXXXXXX
pjsua_custom_options: '--no-tcp'
The way to call is that one:
service: hassio.addon_stdin
data:
addon: 89275b70_dss_voip
input:
call_sip_uri: sip:[email protected]
message_tts: Some text
And this is in the protocol:
11:39:56.403 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
11:39:56.403 sip_endpoint.c .Creating endpoint instance...
11:39:56.403 pjlib .select() I/O Queue created (0x7f8af0cf3100)
11:39:56.403 sip_endpoint.c .Module "mod-msg-print" registered
11:39:56.403 sip_transport.c .Transport manager created.
11:39:56.403 pjsua_core.c .PJSUA state changed: NULL --> CREATED
11:39:56.410 pjsua_core.c .pjsua version 2.11.1 for Linux-5.10.61/x86_64 initialized
11:39:56.412 pjsua_app.c .Turning sound device -99 -99 ON
11:39:56.412 main.c Ready: Success
11:39:56.437 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]
>>> 11:39:56.462 tsx0x7f8af0b626d8 .......Temporary failure in sending Request msg INVITE/cseq=19660 (tdta0x7f8af0b5baa8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
11:39:57.412 pjsua_app.c .Turning sound device -99 -99 OFF
11:40:28.462 pjsua_app.c ....Call 0 is DISCONNECTED [reason=408 (Request Timeout)]
No current call
>>> 11:40:47.928 timer.c .Dumping timer heap:
11:40:47.928 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...
Do you have any idea to help me?
pjsua_custom_options: '--no-tcp --ip-addr=192.168.xxx.xxx'
Where 192.168.xxx.xxx is the IP address of your NUC…
Please post FULL log if this will not solve your problem…
/share/dss_voip/dss_pjsua.log:
ALSA lib confmisc.c:855:(parse_card) cannot find card '0'
ALSA lib conf.c:5111:(_snd_config_evaluate) function snd_func_card_inum returned error: No such file or directory
ALSA lib confmisc.c:422:(snd_func_concat) error evaluating strings
ALSA lib conf.c:5111:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
ALSA lib confmisc.c:1334:(snd_func_refer) error evaluating name
ALSA lib conf.c:5111:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
ALSA lib conf.c:5599:(snd_config_expand) Evaluate error: No such file or directory
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM sysdefault
ALSA lib confmisc.c:855:(parse_card) cannot find card '0'
ALSA lib conf.c:5111:(_snd_config_evaluate) function snd_func_card_inum returned error: No such file or directory
ALSA lib confmisc.c:422:(snd_func_concat) error evaluating strings
ALSA lib conf.c:5111:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
ALSA lib confmisc.c:1334:(snd_func_refer) error evaluating name
ALSA lib conf.c:5111:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
ALSA lib conf.c:5599:(snd_config_expand) Evaluate error: No such file or directory
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM sysdefault
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.front
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround21
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround21
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround40
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround41
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround50
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround51
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround71
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.iec958
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.iec958
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.iec958
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.modem
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.modem
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline
ALSA lib confmisc.c:855:(parse_card) cannot find card '0'
ALSA lib conf.c:5111:(_snd_config_evaluate) function snd_func_card_inum returned error: No such file or directory
ALSA lib confmisc.c:422:(snd_func_concat) error evaluating strings
ALSA lib conf.c:5111:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
ALSA lib confmisc.c:1334:(snd_func_refer) error evaluating name
ALSA lib conf.c:5111:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
ALSA lib conf.c:5599:(snd_config_expand) Evaluate error: No such file or directory
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM default
ALSA lib confmisc.c:855:(parse_card) cannot find card '0'
ALSA lib conf.c:5111:(_snd_config_evaluate) function snd_func_card_inum returned error: No such file or directory
ALSA lib confmisc.c:422:(snd_func_concat) error evaluating strings
ALSA lib conf.c:5111:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
ALSA lib confmisc.c:1334:(snd_func_refer) error evaluating name
ALSA lib conf.c:5111:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
ALSA lib conf.c:5599:(snd_config_expand) Evaluate error: No such file or directory
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM default
ALSA lib confmisc.c:855:(parse_card) cannot find card '0'
ALSA lib conf.c:5111:(_snd_config_evaluate) function snd_func_card_id returned error: No such file or directory
ALSA lib confmisc.c:422:(snd_func_concat) error evaluating strings
ALSA lib conf.c:5111:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
ALSA lib confmisc.c:1334:(snd_func_refer) error evaluating name
ALSA lib conf.c:5111:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
ALSA lib conf.c:5599:(snd_config_expand) Evaluate error: No such file or directory
ALSA lib pcm.c:2660:(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
/share/dss_voip/dss_sox.log
sox: SoX v14.4.2
Input File : 'http://192.168.2.40:8123/api/tts_proxy/86d47069c11cd37f01092c8be6dcf7334f34eed7_de_-_google_translate.
Channels : 1
Sample Rate : 24000
Precision : 16-bit
Sample Encoding: MPEG audio (layer I, II or III)
Output File : '/share/dss_voip/dss_message_tts.wav'
Channels : 1
Sample Rate : 24000
Precision : 16-bit
Sample Encoding: 16-bit Signed Integer PCM
Endian Type : little
Reverse Nibbles: no
Reverse Bits : no
Comment : 'Processed by SoX'
sox INFO sox: effects chain: input 24000Hz 1 channels
sox INFO sox: effects chain: pad 24000Hz 1 channels
sox INFO sox: effects chain: output 24000Hz 1 channels
[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: 3.5.9
You are running the latest version of this add-on.
System: Home Assistant OS 6.3 (amd64 / generic-x86-64)
Home Assistant Core: 2021.9.6
Home Assistant Supervisor: 2021.09.0
-----------------------------------------------------------
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 --ip-addr=192.168.2.40'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri": "sip:[email protected]", "message_tts": "Alarmanlage wurde ausgelöst"}
Converting audio file 'http://192.168.2.40:8123/api/tts_proxy/86d47069c11cd37f01092c8be6dcf7334f34eed7_de_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '50' seconds.
13:25:35.243 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
13:25:35.243 sip_endpoint.c .Creating endpoint instance...
13:25:35.243 pjlib .select() I/O Queue created (0x7f4509618100)
13:25:35.243 sip_endpoint.c .Module "mod-msg-print" registered
13:25:35.243 sip_transport.c .Transport manager created.
13:25:35.243 pjsua_core.c .PJSUA state changed: NULL --> CREATED
13:25:35.250 pjsua_core.c .pjsua version 2.11.1 for Linux-5.10.61/x86_64 initialized
13:25:35.252 pjsua_app.c .Turning sound device -99 -99 ON
13:25:35.252 main.c Ready: Success
13:25:35.253 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:192.168.2.40:5060>: 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]
>>> 13:25:35.281 tsx0x7f45094876d8 .......Temporary failure in sending Request msg INVITE/cseq=14297 (tdta0x7f4509480aa8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
13:25:36.252 pjsua_app.c .Turning sound device -99 -99 OFF
13:26:07.282 pjsua_app.c ....Call 0 is DISCONNECTED [reason=408 (Request Timeout)]
No current call
>>> 13:26:26.772 timer.c .Dumping timer heap:
13:26:26.772 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...
The call was placed but you can’t hear anything after pick up?
You’ll need to specify base-url on Google TTS config.
No, there is no incoming call. The sound file seems to be ok, i can hear the voice at my pc. I don’t think, that tts is the problem.
I think this is the problem:
>>> 13:25:35.281 tsx0x7f45094876d8 .......Temporary failure in sending Request msg INVITE/cseq=14297 (tdta0x7f4509480aa8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
It’s very strange because 408 error is related to --no-tcp absence but your config has it
Have you a firewall on your LAN? (like Fritzbox SIP filter)
Have you enough credit on your Freevoipdeal account?
Have you checked that SIP call was enabled on your account?
Have you tried same configuration on your RPI3b+ today?
Yes, there is a firewall. But it makes no sense, because with the pi and the same ip everything works fine.
About 10 EUR are avaible…
SIP is also enabled, it works with the Pi fine.
Yes, with the Pi4 2GB and it works directly. The configuration seems to be the same… Ok Hass OS and so on aren’t on the latest versions, and there are different addons active. In my point of view, no of these addons should make these problems.
[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: 3.5.9
You are running the latest version of this add-on.
System: Home Assistant OS 6.2 (aarch64 / raspberrypi4-64)
Home Assistant Core: 2021.8.8
Home Assistant Supervisor: 2021.09.0
-----------------------------------------------------------
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'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri": "sip:[email protected]", "message_tts": "Alarmanlage wurde ausgelöst"}
Converting audio file 'http://192.168.2.40:8123/api/tts_proxy/86d47069c11cd37f01092c8be6dcf7334f34eed7_de_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '50' seconds.
13:30:59.687 os_core_unix.c !pjlib 2.9 for POSIX initialized
13:30:59.689 sip_endpoint.c .Creating endpoint instance...
13:30:59.689 pjlib .select() I/O Queue created (0x7fa33160f0)
13:30:59.689 sip_endpoint.c .Module "mod-msg-print" registered
13:30:59.689 sip_transport.c .Transport manager created.
13:30:59.689 pjsua_core.c .PJSUA state changed: NULL --> CREATED
13:30:59.728 pjsua_core.c .pjsua version 2.9 for Linux-5.10.17/aarch64 initialized
13:30:59.733 pjsua_app.c .Turning sound device -99 -99 ON
13:30:59.733 main.c Ready: Success
13:31:00.774 pjsua_app.c .......Call 0 state changed to CALLING
13:31:00.774 pjsua_app.c .Turning sound device -99 -99 OFF
>>>>
Account list:
[ 0] <sip:172.30.33.4:5060>: 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]
>>> 13:31:00.802 tsx0x7fa25c86b8 .......Temporary failure in sending Request msg INVITE/cseq=25808 (tdta0x7fa25c1a88), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
13:31:00.880 pjsua_app.c .....Call 0 state changed to EARLY (183 Session progress)
13:31:00.883 pjsua_app.c ........Turning sound device -99 -99 ON
13:31:12.696 pjsua_app.c .....Call 0 state changed to CONNECTING
13:31:12.696 pjsua_app.c .....Call 0 state changed to CONFIRMED
13:31:14.087 pjsua_app.c ......Call 0 is DISCONNECTED [reason=200 (Normal call clearing)]
13:31:14.087 pjsua_app_common.c ......
[DISCONNCTD] To: sip:[email protected];tag=9b0313ac611cfb5e52248
Call time: 00h:00m:01s, 1st res in 1147 ms, conn in 12963ms
#0 audio iLBC @8kHz, sendrecv, peer=195.219.64.76:20556
SRTP status: Not active Crypto-suite:
RX pt=104, last update:00h:00m:02.360s ago
total 436pkt 21.8KB (39.2KB +IP hdr) @avg=13.2Kbps/23.7Kbps
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 0.268 0.750 0.250 0.119
TX pt=104, ptime=30, last update:never
total 306pkt 15.3KB (27.5KB +IP hdr) @avg=9.2Kbps/16.6Kbps
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 : 0.000 0.000 0.000 0.000 0.000
RTT msec : 0.000 0.000 0.000 0.000 0.000
Please try to add :5060 port to called number and configuration…
On my rpi3b+ and On my NUC freevoipdeal has working.
I’ll try today to setup it on my Proxmox+hassio NUC