SIP client card, as intercom

TECHFox · March 13, 2024, 5:23pm

Seems like you put a incorrect person entity in the config. You need to set a actual person entity id in the card config.

As for the RasPBX, I’m not sure if that will work. The current setup does need a valid SSL on your Asterisk server for it to work. If you run HA-OS you can try the asterisk add-on, which has some settings pre-configured for the sip card.

There is a new card coming, that does not require SSL or port forwarding, and works with every PBX. It’s still a WIP and the first proof of concept got released for testing. But it might we worth waiting for. You can follow progress in the discord server.

daneboom · March 13, 2024, 5:26pm

Thanks! So if I put an incorrect person entity in, NOTHING shows up? It is just totally blank? In that case sounds like I have the lovelace bit right it just has no way of handling that error?

I wonder if I used the SIP-Hass Addon whether I could somehow call using IAX between the two asterisk instances…quite keen to still have FreePBX but could maybe bridge connections? Hmmmm

TECHFox · March 17, 2024, 10:05pm

The card crashes when you put in a person entity that doesn’t exist. Check your entity id or use the card editor.

You can connect 2 asterisk servers using a trunk. Try this: Connecting two Asterisk Servers with the use of Examples

daneboom · March 18, 2024, 10:14am

Thanks, in the end I found (for my use case) this addon (DSS Notifier for Arm) does exactly what I need. It’s a liiiiiitle beta-y but it’s great when it works!

tung256 · March 23, 2024, 7:27pm

i believe i got the server running as seen here. nothing fancy. just voice, no vmail nor video

however, i like to use SIPnetic (free android app). set up asks for a server so i put in my HA ip

what do i put in credentials?

i will have 3 tablets in the house. where do i create 3 accounts in Asterisk or HA so these 3 tablets can call each other using this SIPnetic app?

thanks

pergola.fabio · March 23, 2024, 7:30pm

You need to setup extensions for that , afterwards you can use them on any sipphone, Google for it, lots of examples how to create it

tung256 · March 23, 2024, 7:59pm

you mean SIP account is only managed via using the HACS card?

here’s my full code:

using the app, im signed in as 100 as you can see. when i dialed “102” on another tablet, i get “Account is not registered” and it auto hang up

pergola.fabio · March 23, 2024, 8:03pm

Those auto generated extensions are used for the Lovelace card , you need to setup your own extensions first

tung256 · March 23, 2024, 10:38pm

happy to say i got all the extensions by allowing Auto Add:

i see these 2 files have the extensions and passwords and that’s what i use with the SIPnetic app and it almost works

This time, the app is giving me “Network error”
the Add On Log shows

i searched this Post for “No registered publish handler for event presence from 100” but seems nobody has this error before

tung256 · March 23, 2024, 11:06pm

using an app like SIPnetic, what should my network settings be to work with Asterisk? my HAOS machine is not accessible via HTTPS, so i disabled TLS as seen here. I believe the port is 5160.
still the android app on tablet 1 (ext 101) cannot call tablet 2 (ext 102)

djalvin79 · April 9, 2024, 8:48pm

Hi,
I’ve a Dahua doorbell VTO2211 at 192.168.1.8. It has internal SIP server enabled on port 5060.
I want to use my wall tablet with HA to answer to the doorbell, see the video and unlock the gate.
I only want a SIP client card to do this.
Can you help me? Can you also tell me the correct values to insert in the configuration?

Thanks!!

Bernst4 · May 4, 2024, 10:09am

Hi,
Asterisk is having trouble setting up the SSL when using the Nabu Casa Cloud. Can anyone give me some hints how to setup?

I wanted to switch from open ports to the cloud solution because of security reasons. The connection worked perfectly when using duckdns with an open Port.

Best regards,
Bastian

hlalayan · May 6, 2024, 9:38am

Hi,

I generate wildcard ssl from nginix proxy manager, when I try to use this generated ssl I get error “FATAL: Certificate file at /addon_configs/a0d7b954_nginxproxymanager/letsencrypt/live/npm-15/fullchain.pem was not found”, it is the soft link from this path /addon_configs/a0d7b954_nginxproxymanager/letsencrypt/archive/npm-15/

BR

MiAutomations · May 7, 2024, 2:07pm

Hello to all

I’m facing this errors at the addon log, last version installed

I’m also unable to register any physical extensions like a Mobile phone APP or a SIP Phone, but the extension card is registered successfully

Is anyone facing a similar issue?

Cr4z33 · June 20, 2024, 7:23am

I am trying to get sip-hass-card working with my Dahua VTO2202F-P-S2 acting as SIP server, but I am struggling at solving the login issue.

Can you please tell us more what to enter in the integration (in case of typos or my misunderstanding)?

Also what would you suggest to use to solve the SSL issues?

So far I’ve tried mkcert trying to certify the LAN IP of Home Assistant, the external URL and homeassistant.local, but I had to give up with the other Asterisk/SIP addon…

projectdimpreza · July 23, 2024, 8:43am

Thanks for this wonderful work @TECHFox
By the way can somebody point me on what’s wrong with my setup… i installed the addon the integration and the card… but when making a call from tablet to phone nothing happens. I’m using Home Assistant and Cloudflare tunnel to access my HA outside my home network… Do i still need to do a port forward? Unfortunately i cannot forward any port on my router because my ISP prohibit forwarding so maybe theres someone here know an alternative way. Below are my logs and screenshots of my current setup. Hope it help debugs the problem.

Parsing /etc/asterisk/asterisk.conf
gl_pathc 3
UUID system initiated
Unable to find key 'UUID' in family 'pbx'
PBX UUID: 7d3e391f-18c6-490f-a6a3-f1ade03919e4
extract int from [5] in [0, 2147483647] gives [5](0)
extract int from [50] in [0, 2147483647] gives [50](0)
extract int from [20] in [0, 2147483647] gives [20](0)
Parsing /etc/asterisk/stasis.conf
Resetting translation matrix
Increasing threadpool stasis/pool's size by 5
Parsing /etc/asterisk/sorcery.conf
Not changing threadpool size since new size 0 is the same as current 0
Creating topic. name: system:all, detail: 
Topic 'system:all': 0x563887358350 created
Creating topic. name: endpoint:all, detail: 
Topic 'endpoint:all': 0x563887358840 created
Creating topic. name: cache_pattern:0/endpoint:all, detail: 
Topic 'cache_pattern:0/endpoint:all': 0x563887358ac0 created
[Jul 23 16:09:15]  Asterisk PBX Core Initializing
[Jul 23 16:09:15]  Asterisk Dynamic Loader Starting:
[Jul 23 16:09:15] NOTICE[511]: loader.c:2559 load_modules: 317 modules will be loaded.
[Jul 23 16:09:15] NOTICE[511]: cdr.c:4568 cdr_toggle_runtime_options: CDR simple logging enabled.
[Jul 23 16:09:15]  Bound HTTP server 'http server' to address [::]:8088
[Jul 23 16:09:15]   == TLS/SSL certificate ok
[Jul 23 16:09:15] NOTICE[511]: indications.c:1100 load_indications: Default country for indication tones: us
[Jul 23 16:09:15] NOTICE[511]: indications.c:424 ast_set_indication_country: Setting default indication country to 'us'
[Jul 23 16:09:15] WARNING[511]: config.c:2130 process_text_line: parse error: No category context for line 310 of /etc/asterisk/geolocation.conf
[Jul 23 16:09:15] ERROR[511]: res_sorcery_config.c:334 sorcery_config_internal_load: Contents of config file 'geolocation.conf' are invalid and cannot be parsed
[Jul 23 16:09:15] WARNING[511]: config.c:2130 process_text_line: parse error: No category context for line 310 of /etc/asterisk/geolocation.conf
[Jul 23 16:09:15] ERROR[511]: res_sorcery_config.c:334 sorcery_config_internal_load: Contents of config file 'geolocation.conf' are invalid and cannot be parsed
[Jul 23 16:09:15]     -- Local IPv4 address determined to be: 192.168.0.175
[Jul 23 16:09:15]     -- Local IPv6 address determined to be: fe80::bfb2:1d3a:b6bf:c8b8
[Jul 23 16:09:16]   == 'UDP+IPv4' is an available SIP transport
[Jul 23 16:09:16]   == 'TCP+IPv4' is an available SIP transport
[Jul 23 16:09:16]   == 'TLS+IPv4' is an available SIP transport
[Jul 23 16:09:16]   == 'UDP+IPv6' is an available SIP transport
[Jul 23 16:09:16]   == 'TCP+IPv6' is an available SIP transport
[Jul 23 16:09:16]   == 'TLS+IPv6' is an available SIP transport
[Jul 23 16:09:16]   == libsrtp2 2.5.0 initialized
[Jul 23 16:09:16] WARNING[511]: res_phoneprov.c:1249 get_defaults: Unable to find a valid server address or name.
[Jul 23 16:09:16] ERROR[511]: netsock2.c:96 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
[Jul 23 16:09:16]   == RTP Allocating from port range 10000 -> 20000
[Jul 23 16:09:16] NOTICE[511]: res_smdi.c:1424 load_module: No SMDI interfaces are available to listen on, not starting SMDI listener.
[Jul 23 16:09:16]   == Binding IAX2 to default address 0.0.0.0:4569
[Jul 23 16:09:16]   == 10 helper threads started
[Jul 23 16:09:16]   == IAX Ready and Listening
[Jul 23 16:09:16]     -- Loaded provisioning template 'default'
[Jul 23 16:09:16] ERROR[511]: ari/config.c:312 process_config: No configured users for ARI
[Jul 23 16:09:16] NOTICE[511]: confbridge/conf_config_parser.c:2410 verify_default_profiles: Adding default_menu menu to app_confbridge
[Jul 23 16:09:16] NOTICE[511]: cel_custom.c:92 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Jul 23 16:09:16]  Added CEL CSV mapping for 0 files.
[Jul 23 16:09:16]   == Manager 'admin' logged on from 127.0.0.1
[Jul 23 16:09:17]     -- Security Logging Enabled
[Jul 23 16:09:17]   == Setting global variable 'CONSOLE-AEL' to '"Console/dsp"'
[Jul 23 16:09:17]   == Setting global variable 'OUTBOUND-TRUNK' to '"Zap/g2"'
[Jul 23 16:09:17]   == Setting global variable 'OUTBOUND-TRUNKMSD' to '1'
[Jul 23 16:09:17]     -- Including switch 'DUNDi/e164' in context 'ael-dundi-e164-switch'
[Jul 23 16:09:17]     -- No voicemail provider registered.
[Jul 23 16:09:17]   == DUNDi Ready and Listening on 0.0.0.0:4520
[Jul 23 16:09:17]     -- Registered handler for 'Spandsp' (Spandsp FAX Driver)
[Jul 23 16:09:18] Asterisk Ready.

Calling

type: custom:sipjs-card
server: 192.168.0.175
port: '8089'
button_size: '48'
state_color: true
auto_answer: true
hide_me: true
custom_title: ''
video: false
ringtone: /local/upload/ding_dong.mp3
extensions:
  - person: person.pc
    name: PC
    extension: '100'
    secret: 1224
    icon: mdi:tablet
    entity: binary_sensor.100_registered
    camera: ''
  - person: person.ha_dashboard
    name: Dashboard
    extension: '101'
    secret: 1224
    icon: mdi:tablet
    entity: binary_sensor.102_registered
    camera: ''
ringbacktone: /local/upload/ding_dong.mp3
custom: []
dtmfs: []
buttons: []

AMI
Sensors

projectdimpreza · July 24, 2024, 1:04pm

@jojeda654 are there any alternative way aside from forwarding router ports? Unfortunately my ISP is a dynamic ip and wont allow me to forward ports… Im also accessing my HA outside my network using Cloudflare tunnel. Right now nothing happens when i make a call…

karim-bahgat · August 3, 2024, 12:06pm

Hi, I’m trying to add a doorbell but unfortunately it can’t register.
also when I restart the asterisk add-on everything I have done is erased.
here is my pjsip_custom.conf code

[8001]
type = endpoint
context = default
disallow = all
allow = alaw,ulaw          ; Audio codecs
allow=h264                 ; Video codecs
direct_media_method=invite
dtmf_mode=info
callerid="Doorbell" <8001>
force_rport=no
aors= 8001
auth = auth8001

[8001]
type = aor
max_contacts = 1

[auth8001]
type=auth
auth_type=userpass
password=1234        ; Set your password here
username=8001

This is Asterisk log

'<sip:[email protected]>' failed for '192.168.1.32:5060' (callid: 1141003386) - No matching endpoint found
[Aug  3 15:03:52] NOTICE[637]: res_pjsip/pjsip_distributor.c:673 log_failed_request: Request 'REGISTER' from '<sip:[email protected]>' failed for '192.168.1.32:5060' (callid: 1141003386) - Failed to authenticate

can anyone help me?

Edit

you need to copy pjsip_custom.conf file and extensions.conf file to asterisk/custom file

projectdimpreza · August 13, 2024, 5:50am

@TECHFox is there a symlink for the “sounds” folder? I want to upload a custom pre recorded audio to default path var/lib/asterisk/sounds/en but seems its not possible because its inside the container… is there any workaround for this?
My goal is to play a pre recorded multicast page on all of my extensions… or maybe someone here know a solution?

altrdev · August 24, 2024, 4:39pm

Same problem here!