sdesalve
commented
1 year ago [Info] Listening for messages via stdin service call...
please hide your number
have you enabled SIP calls within FreeVoipDeal settings?
Closed wptracy closed 1 year ago
sdesalve
commented
1 year ago [Info] Listening for messages via stdin service call...
please hide your number
have you enabled SIP calls within FreeVoipDeal settings?
wptracy
commented
1 year ago Yes, It was already enabled, but I pushed save settings anyway, because I read a previous similar issue where you said check that setting.
Does this app need a stun server setting option?
n: detect NAT type ?
I don't think I'm registered on their sip server.
sdesalve
commented
1 year ago Does this app need a stun server setting option?
Only if you can't hear anything after call was picked up
I don't think I'm registered on their sip server.
You will not be registered with that configuration. And definitely you don't need to be registered to place a call
System: Home Assistant OS 9.4 (aarch64 / raspberrypi4-64)
Damns! Please install ARM version
wptracy
commented
1 year ago I installed Home Assistant OS version arm7l on the Rpi.
https://github.com/home-assistant/operating-system/releases/download/9.4/haos_rpi4-64-9.4.img.xz
I added your repository, but found no App for DSS VoIP Notifier.
OS Version: Home Assistant OS 9.4
Home Assistant Core: 2023.1.7
Home Assistant URL: http://homeassistant.local:8123
Observer URL: http://homeassistant.local:4357
[core-ssh ~]$ arch
armv7l
SDeSalve add-ons for Hassio
SDeSalve <me@sdesalve.it>
https://github.com/sdesalve/hassio-addons
why did you close this. I've tried both arm 32 and 64 bit.
I see you have dss_voip and dss_voipARM
Is sdd_voip for 64 bit and dss_voipARM for 32 bit ?
I'm still having trouble getting this working.
sdesalve
commented
1 year ago Arm version is with old Hassio image base. It's same script. For some unknown reason new base image don't place calls in arm.
Where did you add my repository? It's for Supervised Home Assistant and should add it from Hassio GUI...
wptracy
commented
1 year ago I get a SIP 404 Unauthorized when I try to make a call from my IP phone to my home land line. I have my freevoipdeal user and password configured.
SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 98.43.186.186:3073;branch=z9hG4bK-yrp3m4kqjnc2;rport From: "freevoipdeal" sip:xxx@sip.freevoipdeal.com;tag=uqttkatfjc
I did an ARP Spoof on home assistant and captured the SIP messages going from home assistant to the router and back.
Your app is sending out the SIP messages and freevoipdeal is sending 401 unauthorized.
I have the correct user and password configured in both my IP phone and your App.
I'm using the same user and password of my login account at freevoipdeal.
Is it the USA phone number that's unauthorized, or my user password that's unauthorized.
My IP phone is registered with freevoipdeal so I must be using the correct user and password.
I can send you the SIP trace if you like.
I found these settings on their web site, I tried them, It still doesn't work.
| SIP port : 5060
| Registrar : sip.freevoipdeal.com
| Proxy server : sip.freevoipdeal.com
| Outbound proxy server : leave empty
| Account name : your FreeVoipDeal username
| Password : your FreeVoipDeal password
| Display name/number : your FreeVoipDeal username or voipnumber
| Stunserver (option) : stun.freevoipdeal.comI gave up trying to use freevoipdeal. I submitted 4 trouble tickets and they never responded.
I got my money back and went elsewhere.
I found another SIP provider and I'm trying to make the app work there.
The call won't go through. Here is the SIP INVITE message, my App configuration, hassio.addon_stdin, and log file
If I send a call_sip_uri: without a port, I see SIP messages.
service: hassio.addon_stdin
data:
addon: 89275b70_dss_voip
input:
call_sip_uri: sip:13039339035@sip.provider.com
message_tts: Write here your message
If I send a call_sip_uri: with a port, I don't see SIP messages.
service: hassio.addon_stdin
data:
addon: 89275b70_dss_voip
input:
call_sip_uri: 'sip:13039339035@sip.provider.com:5008'
message_tts: Write here your message
Here is my configuration file:
caller_id_uri: sip:user@sip.provider.com:5008
realm: "*"
username: "user"
password: 'password'
pjsua_custom_options: "--no-tcp"
The app doesn't send the required port on the URL, but if I put the credentials and sip provider in my IP phone, it does send the port.
dss_voip App's Invite message:
INVITE sip:13039339035@sip.provider.com SIP/2.0
Via: SIP/2.0/UDP 172.30.33.5:5060;rport;branch=z9hG4bKPjIgp4xPwpn4rxNKnkwiv1rVMFbq9z-2EL
Max-Forwards: 70
From: sip:username@sip.provider.com;tag=YaZG4I-UzV0d6B1fKm5-oEDOobVElWOd
To: sip:13039339035@sip.provider.com
Contact: <sip:username@172.30.33.5:5060;ob>
Call-ID: VPE8OhPfU7EVOTiLG8s1aau0S9vm6UHd
CSeq: 15221 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: PJSUA v2.11.1 Linux-5.15.76/aarch64
Content-Type: application/sdp
Content-Length: 626
My SNOM IP phone User-Agent Invite message.
INVITE [sip:3039339035@sip.provider.com:5008;user=phone](mailto:sip:3039339035@sip.provider.com:5008;user=phone) SIP/2.0
Via: SIP/2.0/UDP 98.43.186.186:3073;branch=z9hG4bK-8e128z6h7a0e;rport
From: "HomeAssistant.sip.provider.com" [<sip:6859810135@sip.provider.com:5008>](mailto:sip:6859810135@sip.provider.com:5008);tag=t2hjnib6bj
To: [<sip:3039339035@sip.provider.com:5008;user=phone>](mailto:sip:3039339035@sip.provider.com:5008;user=phone)
Call-ID: 313637343835303434303338373138-9y7dzebfqzq2
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom821/8.7.5.35
Contact: [<sip:6859810135@98.43.186.186:3073;line=1t35fiwd>](mailto:sip:6859810135@98.43.186.186:3073;line=1t35fiwd);reg-id=1
X-Serialnumber: 00041348D068
P-Key-Flags: resolution="31x13", keys="4"
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600
Min-SE: 90
Content-Type: application/sdp
Content-Length: 516
LOG File:
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:13039339035@sip.provider.com","message_tts":"Write here your message"}
Converting audio file 'http://10.0.0.233:8123/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:13039339035@sip.provider.com'...
This call will be terminated after '50' seconds.
14:59:50.905 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
14:59:50.906 sip_endpoint.c .Creating endpoint instance...
14:59:50.907 pjlib .select() I/O Queue created (0x7f9ef4b100)
14:59:50.907 sip_endpoint.c .Module "mod-msg-print" registered
14:59:50.907 sip_transport.c .Transport manager created.
14:59:50.907 pjsua_core.c .PJSUA state changed: NULL --> CREATED
14:59:50.931 pjsua_core.c .pjsua version 2.11.1 for Linux-5.15.76/aarch64 initialized
14:59:50.940 pjsua_app.c .Turning sound device -99 -99 ON
14:59:50.940 main.c Ready: Success
14:59:50.944 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.5:5060>: does not register
Online status: Online
[ 1] <sip:172.30.33.5:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:6859810135@sip.provider.com:5008: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:13039339035@sip.provider.com
+=============================================================================+
| 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:13039339035@sip.provider.com [CALLING]
>>> 14:59:50.991 tcpc0x7f9e1d0518 !TCP connect() error: [code=120111]: Connection refused
14:59:50.991 tsx0x7f9e1e3c18 Temporary failure in sending Request msg INVITE/cseq=19575 (tdta0x7f9e1dbaa8), will try next server: Connection refused
14:59:50.992 pjsua_app.c SIP TCP transport is disconnected from 67.231.2.13:5060: Connection refused [status=120111]
14:59:51.049 tcpc0x7f9e1ce528 TCP connect() error: [code=120111]: Connection refused
14:59:51.049 tsx0x7f9e1e3c18 Temporary failure in sending Request msg INVITE/cseq=19575 (tdta0x7f9e1dbaa8), will try next server: Connection refused
14:59:51.049 pjsua_app.c SIP TCP transport is disconnected from 216.82.238.135:5060: Connection refused [status=120111]
14:59:51.941 pjsua_app.c .Turning sound device -99 -99 OFF
15:00:23.049 pjsua_app.c ....Call 0 is DISCONNECTED [reason=408 (Request Timeout)]
No current call
>>> 15:00:42.442 timer.c .Dumping timer heap:
15:00:42.442 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...[Info] Starting addon... [Info] Listening for messages via stdin service call... [Info] Received messages {"call_sip_uri":"sip:+xxx@sip.freevoipdeal.com","message_tts":"Write here your message"}
Damns! Here I should have checked that your pjsua_custom_options was not recognized. Sorry
Happy that you've resolved your issues
I can't get it to work.
I have 10 Euro on the account.
I also disabled the firewall: LAN-to-WAN : Allow all. WAN-to-LAN : Allow all.