Hi,
is it possible to send the caller number to a script action?
Something like this maybe:
action:
domain: script
service: turn_on
entity_id: script.ha_sip_forward_call
service_data:
variables:
message: "test"
caller: "{{ caller }}"
Yes, this is available in the webhook callback with {{trigger.json.caller}}. See also GitHub - arnonym/ha-plugins: Home-Assistant SIP Gateway
Thank you for the quick reply.
Yes, that’s available if you trigger an automation with a webhook, but I would like to directly call a script in an action from an incoming call and give the caller number as a variable.
The dirty workaround would be to use a webhook in an automation and call the script from there, but I was hoping to call the script directly from my sip-1-inciming.yaml
I’m afraid it’s not possible to do that in the incoming-yaml file, as this is not parsed by home-assistant. If you don’t want to mix up incoming.yaml and webhooks you can do it like described in Generating dynamic Menus and messages on HA Entity State · Issue #86 · arnonym/ha-plugins · GitHub. That way you also don’t need to access the config directory to change your sip actions, but can do it directly from the home-assistant UI. Personally I don’t feel dirty doing it that way.
And you can even have different menus per caller, as you can use conditions to accept certain calls only.
That’s actually a good idea, but for some reason the jump command isn’t working there. The logs say:
| 20:26:46.660766 [ ] Webhook response 200 b''
| 20:26:46.664727 [1] Scheduled post action: jump
| 20:26:46.664788 [1] Could not find menu id: forward
This is my config, which was working fine in sip-1-incoming.yaml (except for the trigger parts):
service: hassio.addon_stdin
metadata: {}
data:
addon: c7744bff_ha-sip
input:
command: answer
number: "{{ trigger.json.parsed_caller }}"
menu:
message: >-
Welcome message
choices:
"1":
id: weiterleitung
message: >-
Ok, forward to who?
choices:
"1":
id: call_a
message: Ok, you will be connected to A
post_action: jump forward
"2":
id: call_b
message: Ok, you will be connected to B
post_action: jump forward
"999999999999999":
id: forward
message: Ok
action:
domain: script
service: turn_on
entity_id: script.ha_sip_forward_call
service_data:
variables:
caller: "{{ trigger.json.parsed_caller }}"
message: Ok
The menu from the file and menu from stdin are handled exactly the same. At the beginning of every call ha-sip logs the parsed menu. Can you check if the choice is correctly parsed in the logs?
You asked for a stand-alone docker version. Fyi - with todays release it is possible.
You will need an Mqtt Broker like Mosquitto for the communcation. If that is set up - all you need to do is to create your own .env file with your connection details. Then you can do a docker compose up and post your commands as json to hasip/execute! Enjoy! 
Thank you very much, it’s great to see your reply to create a version for using docker. I will use it for whatever purpose I will inform you. have a nice day
Thanks a lot, that’s awesome news!!
Now I have to spent some time to set it up and think of some useful automations 
@arnonym I know I am a little bit late to the party, but I just wanted to say that this add-on is probably the absolute best thing I have ever added to my HA! I have a grandstream UCM IP PBX at home and this really ties everything together really splendidly. I’ve (so far) used the “Announcement Center” in the PBX to upload about 8 .wav files for things like doors being opened, motion detection, etc - and can now trigger them through automations using this add-on. The announcements play in perfect unison across all the IP phones. I’ve only just scratched the surface with what’s possible with this add-on but still wanted to say thanks!
hi, does TTS work for anyone with Piper?
Uhm…what about the error after the 2016.6 update?
Really awesome its working great for me. However I havent been able to get it working with piper TTS - has anyone else managed to do this?
Hi @arnonym , sorry to bother you with this. I’ve finally stumbled across your incredible work and I am a little stuck with a call menu. I’m trying to call into my Home Assistant, but the DTMF tones aren’t working. I have an option to hang-up if no menu option selected, so I know the menu is working. I’m unable to hear the tones when I call the system. I saw a previous post with a similar issue and changing the ‘method’, however this looked like it was for an outbound call. I’m using the HA-SIP service as an operator, that I can call inbound. Hope this makes sense. Thanks again for your amazing work on this.
I got HA-SIP connected to my 3CX portal. Incoming calls trigger the webhook but I would love to see the number on screen. It seems so simple no one mentioned it in the docs 
Even better if I can select the caller ID for two different screens (doorphone and landline)
Any example code available and where to place it?
I left the webhook option in the add-on standard and just trigger on the webhook now.
Can I somehow initiate a call to 2 phones (one internal, and one external) from HA? So make both phones ring, and after they both pick up, they can speak to each other?
Hi @arnonym
I think I manage to connect my doorbell.
11:00:56.039 pjsua_core.c .PJSUA state changed: NULL --> CREATED
11:00:56.040 sip_endpoint.c .Module "mod-pjsua-log" registered
11:00:56.040 sip_endpoint.c .Module "mod-tsx-layer" registered
11:00:56.040 sip_endpoint.c .Module "mod-stateful-util" registered
11:00:56.040 sip_endpoint.c .Module "mod-ua" registered
11:00:56.040 sip_endpoint.c .Module "mod-100rel" registered
11:00:56.040 sip_endpoint.c .Module "mod-pjsua" registered
11:00:56.040 sip_endpoint.c .Module "mod-invite" registered
11:00:56.047 alsa_dev.c ..ALSA driver found 0 devices
11:00:56.047 alsa_dev.c ..ALSA initialized
11:00:56.047 pjlib ..select() I/O Queue created (0x7f99945bba48)
11:00:56.051 sip_endpoint.c .Module "mod-evsub" registered
11:00:56.051 sip_endpoint.c .Module "mod-presence" registered
11:00:56.051 sip_endpoint.c .Module "mod-mwi" registered
11:00:56.051 sip_endpoint.c .Module "mod-refer" registered
11:00:56.051 sip_endpoint.c .Module "mod-pjsua-pres" registered
11:00:56.051 sip_endpoint.c .Module "mod-pjsua-im" registered
11:00:56.051 sip_endpoint.c .Module "mod-pjsua-options" registered
11:00:56.051 pjsua_core.c .No SIP worker threads created
11:00:56.051 pjsua_core.c .pjsua version 2.14.1 for Linux-6.6.54/x86_64 initialized
11:00:56.051 pjsua_core.c .PJSUA state changed: CREATED --> INIT
| 11:00:56.051589 [ ] Supported audio codecs: speex/16000/1, speex/8000/1, speex/32000/1, iLBC/8000/1, GSM/8000/1, PCMU/8000/1, PCMA/8000/1, G722/16000/1, opus/48000/2, L16/44100/2, L16/44100/1
11:00:56.051 pjsua_aud.c Setting null sound device..
11:00:56.051 pjsua_aud.c .Opening null sound device..
11:00:56.052 pjsua_core.c SIP UDP socket reachable at 192.168.1.15:5060
11:00:56.053 udp0x7f99922cefd0 SIP UDP transport started, published address is 192.168.1.15:5060
11:00:56.053 pjsua_core.c PJSUA state changed: INIT --> STARTING
11:00:56.053 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
11:00:56.053 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
11:00:56.053 pjsua_acc.c Adding account: id=sip:[email protected]
11:00:56.053 pjsua_acc.c .Account sip:[email protected] added with id 0
11:00:56.053 pjsua_acc.c .Acc 0: setting registration..
11:00:56.053 pjsua_acc.c ..Contact for acc 0 updated: <sip:[email protected]:5060;ob>;+sip.ice
11:00:56.053 pjsua_core.c ...TX 521 bytes Request msg REGISTER/cseq=63660 (tdta0x7f99920b34a8) to UDP 47.91.88.33:5060:
REGISTER sip:vdpconnect.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5060;rport;branch=z9hG4bKPjsMUzRZ2SMs4RuHkKlO6zN2i3DHaRPzOk
Max-Forwards: 70
From: <sip:[email protected]>;tag=beMg-RDsadSdT5nL001ETPIs9N14Qi.F
To: <sip:[email protected]>
Call-ID: QNuGlyuEED2AobJBePD-gvdiMMuj64hJ
CSeq: 63660 REGISTER
Contact: <sip:[email protected]:5060;ob>;+sip.ice
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
--end msg--
11:00:56.053 pjsua_acc.c ..Acc 0: Registration sent
11:00:56.094 pjsua_core.c .RX 540 bytes Response msg 401/REGISTER/cseq=63660 (rdata0x7f99920ae0d8) from UDP 47.91.88.33:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.15:5060;received=79.131.159.180;rport=57895;branch=z9hG4bKPjsMUzRZ2SMs4RuHkKlO6zN2i3DHaRPzOk
From: <sip:[email protected]>;tag=beMg-RDsadSdT5nL001ETPIs9N14Qi.F
To: <sip:[email protected]>;tag=47f5494f4dd8677feafc1bbd25a7fb92.9417
Call-ID: QNuGlyuEED2AobJBePD-gvdiMMuj64hJ
CSeq: 63660 REGISTER
WWW-Authenticate: Digest realm="vdpconnect.com", nonce="6714b8d60000c9a5760ac251cd8d88d92ec3c52a89659383"
Server: OpenSIPS (2.3.2 (x86_64/linux))
Content-Length: 0
--end msg--
11:00:56.094 pjsua_acc.c ....IP address change detected for account 0 (192.168.1.15:5060 --> 79.131.159.180:57895). Updating registration (using method 4)
11:00:56.094 pjsua_acc.c ....Contact for acc 0 updated: <sip:[email protected]:57895;ob>;+sip.ice
11:00:56.094 sip_auth_client.c ...Digest algorithm is ""
11:00:56.094 pjsua_core.c ....TX 727 bytes Request msg REGISTER/cseq=63661 (tdta0x7f99920b34a8) to UDP 47.91.88.33:5060:
REGISTER sip:vdpconnect.com SIP/2.0
Via: SIP/2.0/UDP 79.131.159.180:57895;rport;branch=z9hG4bKPjUAmqhflxP7S8PJJIBpA6PgIqPKPZTPPU
Max-Forwards: 70
From: <sip:[email protected]>;tag=beMg-RDsadSdT5nL001ETPIs9N14Qi.F
To: <sip:[email protected]>
Call-ID: QNuGlyuEED2AobJBePD-gvdiMMuj64hJ
CSeq: 63661 REGISTER
Contact: <sip:[email protected]:57895;ob>;+sip.ice
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="0e001a2fea54", realm="vdpconnect.com", nonce="6714b8d60000c9a5760ac251cd8d88d92ec3c52a89659383", uri="sip:vdpconnect.com", response="9a6c0d12526678fda2b88ce90bb06772"
Content-Length: 0
--end msg--
11:00:56.140 pjsua_core.c .RX 629 bytes Response msg 200/REGISTER/cseq=63661 (rdata0x7f99920ae038) from UDP 47.91.88.33:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 79.131.159.180:57895;received=79.131.159.180;rport=57895;branch=z9hG4bKPjUAmqhflxP7S8PJJIBpA6PgIqPKPZTPPU
From: <sip:[email protected]>;tag=beMg-RDsadSdT5nL001ETPIs9N14Qi.F
To: <sip:[email protected]>;tag=47f5494f4dd8677feafc1bbd25a7fb92.2207
Call-ID: QNuGlyuEED2AobJBePD-gvdiMMuj64hJ
CSeq: 63661 REGISTER
Contact: <sip:[email protected]:57895;ob>;expires=300;received="sip:79.131.159.180:57895", <sip:[email protected]:49307;transport=udp>;expires=19;received="sip:79.131.159.180:49307"
Server: OpenSIPS (2.3.2 (x86_64/linux))
Content-Length: 0
--end msg--
11:00:56.140 pjsua_acc.c ....SIP outbound status for acc 0 is not active
11:00:56.140 pjsua_acc.c ....sip:[email protected]: registration success, status=200 (OK), will re-register in 300 seconds
11:00:56.140 pjsua_acc.c ....Keep-alive timer started for acc 0, destination:47.91.88.33:5060, interval:13s
| 11:00:56.140991 [1] OnRegState: 200 OK
11:00:57.053 pjsua_aud.c Closing sound device after idle for 1 second(s)
11:00:57.053 pjsua_aud.c .Closing null sound device..
with this configuration
sip_global:
port: 5060
log_level: 5
name_server: ""
cache_dir: /config/audio_cache
sip:
enabled: true
registrar_uri: sip:vdpconnect.com
id_uri: sip: [email protected]
realm: "*"
user_name: xxxxxxx
password: "xxxxxx"
answer_mode: listen
settle_time: 1
incoming_call_file: ""
sip_2:
enabled: false
registrar_uri: sip:fritz.box
id_uri: sip:[email protected]
realm: "*"
user_name: anotheruser
password: secret
answer_mode: listen
settle_time: 1
incoming_call_file: ""
sip_3:
enabled: false
registrar_uri: sip:192.168.178.10
id_uri: ""
realm: "*"
user_name: ""
password: ""
answer_mode: listen
settle_time: 1
incoming_call_file: ""
tts:
platform: google_translate
language: en
webhook:
id: xxxxxx
Failed to save add-on configuration, Missing option 'sip_3' in root in ha-sip (c7744bff_ha-sip). Got {'sip_global': {'port': 5060, 'log_level'
for the time being I haven’t add anything else.
I have never user an event or webhook trigger and nothing I have tried until now seems to working. So I am not sure that everything is ok in my setup.
i tried something like this but it didn’t work
- id: SIP Incoming call
alias: SIP Incoming call
description: ""
trigger:
- platform: webhook
allowed_methods:
- POST
local_only: true
webhook_id: doorbell_ringing_webhook
condition: []
action:
- action: notify.mobile_app_makis_smartphone
data:
message: 'the doorbell is ringing!'
mode: single
I think this is a problem but I am not sure.
SIP/2.0 401 Unauthorized
Can anyone help please?