SIP Doorbell, android tablet and HA integration

greengolfer · February 23, 2021, 11:06am

No. Only for the tablet one is webrtc. The GDS3710 is a “normal” (PJ)SIP extension.

I would enable debug on freepbx side. Connexion to astrieks CLI (asterisk -r) and debug PJSIP dialog.

divieshm · February 23, 2021, 12:07pm

OK question then?

Which extension is being used on the tablet, 1003 or 1004, as 1004 is set to disallow h264/mpeg4, so the video is disabled on 1004, then next point on consideration does 1003 also need webrtc need enabling on 1003, or am I misunderstanding this

greengolfer · February 23, 2021, 12:31pm

If you are using the example I gave:

1003 is a fake extension (virtual) always on so that if the tablet is asleep (and extension unavailable), then the call will not fail
1004 is the tablet and should be the only one with webrtc configuration.

My suggestion, while testing use only 1004. And if your tablet goes asleep when on production mode then use the 1003 trick.
When you are attempting to install, extension 1004 should always be available as you will manually keep the tablet alive.

divieshm · March 3, 2021, 10:34am

Morning Everyone, great progress last night, finally have audio working between HA, freepbx and my Intercom, now going to work on Input Helpers to unlock the door, play the ringing in HA, and node-red, are there any resources you can point me to to learn more about these

greengolfer · March 3, 2021, 1:27pm

In the doordroid card, two input boolean are used (see post #1) and are available in HA:

input_boolean.gds_ringing which is turned on when someone is ringing on the GDS
input_boolean.open_portail which is turned on when you click on the open button on the interface

With these two, you can do pretty much want you want using either automations or nodered.

divieshm · March 3, 2021, 1:55pm

they are in the doordroid card do I need to create them anywhere in HA?

greengolfer · March 3, 2021, 2:52pm

Yes. As “normal” input_boolean.

divieshm · March 3, 2021, 3:13pm

awsome I thought as much - question on the GDS (I’m not using GDS) what is the command to open, on the intercom I’m using it is just “00” how do I initiate the code 00 to unlock, i.e if I call ext sip extension, once answered (say on a sip phone) if type 00 on the phone keypad it opens the door

greengolfer · March 3, 2021, 5:30pm

I don’t know either. My latch is connected to a shelly 1 to open the door and not to the intercom it self.
So, “open_portail” action has nothing to do with my intercom.

divieshm · March 6, 2021, 1:12pm

Any chance you can share a wiring diagram for the shelly 1?

greengolfer · March 6, 2021, 4:54pm

Here you go! It is a shelly 2 but wiring is (almost) the same.
The “Outgoing when turned on” is then connected to a 220V/12V adaptor. When On (for a few seconds) the latch is unlocked. It depends on the way your latch is working. Mine is closed when no power is sent and released when 12V is received.

divieshm · March 6, 2021, 7:11pm

Awesome, that helped, the latch is working, I have started doing the node-red automation to create an action, and currently have it working in node-red, where i manually change the state on the input boolean to off from on the latch kick in and does what it needs to.

One issue I’m having though is the Open the Door button on droidcard isn’t doing anything, I click on the button and its not triggering the state change of the input boolean, but if I manually go into the services and trigger the state change it worked

do I have to do anything in doordroid or in Lovelace to get the “open the door” button to trigger the state change the input bolean

greengolfer · March 6, 2021, 7:17pm

Can you go in the developper tools (the hammer on the side) and check if the input_boolean exists and what is its state?
You may have to force it to off in the configuration:

input_boolean:
    open_portail:
      initial: off
      icon: mdi:door

The javascript code in the card is quite simple:

        openDoorBtn.addEventListener('click', function(opendoor) {
            hass.callService('input_boolean', 'turn_on', { entity_id: 'input_boolean.open_portail' });
        });

So if the boolean exists and is off, when you click it should turn on…

divieshm · March 6, 2021, 7:25pm

the input_boolean definitely exists and is turned off, when clicking it in dev tools it works, but doesn’t from the droidcard

greengolfer · March 6, 2021, 7:37pm

(Re)check the spelling. That is the only explanation I can think of… Otherwise, I don’t know.

sascha005 · June 12, 2021, 6:31pm

Hello, i have also setup this solution for a DIY SIP basend Doorbell. I have followed the whole Howto and have Homeassistant, FreePBX with webrtc running. All works as it should (for example, i can call from a local sip phone to the SIMPL5 Demo connected to my local FreeBBX.) The only thing that does not work ist the lovelace doorpi-card. It shows “registered” on development console but receive not call at all. Did i misunderstood something ? If the SIMPL5 Demo works, should the Doorpi Card works as well, or not ? Dont Know where to start to find the Problem. Can someone help me out with this ? Regards Sascha

greengolfer · June 12, 2021, 6:52pm

I would look at debugging on freepbx. Do you see the Doorbell registered? If yes, if you try to call, what happen on the debug…

sascha005 · June 12, 2021, 7:08pm

Hey there, thank you for your answer. If i load the doorpi card in lovelace i got an “registered” in the browser dev console an the following in asterisk log :

^

REGISTER sip:192.168.179.240 SIP/2.0
Via: SIP/2.0/WSS 33aj8uv7rhf4.invalid;branch=z9hG4bK7298384
Max-Forwards: 69
To: sip:[email protected]
From: sip:[email protected];tag=72ghjb9095
Call-ID: r9c7kncp97kq4ouuq2gvae
CSeq: 1 REGISTER
Contact: sip:[email protected];transport=ws;+sip.ice;reg-id=1;+sip.instance=“urn:uuid:02cd1e43-624a-471b-9e41-c994a803712c”;expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: path,gruu,outbound
User-Agent: JsSIP 3.7.4
Content-Length: 0

<— Transmitting SIP response (469 bytes) to WSS:178.5.65.253:56930 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 33aj8uv7rhf4.invalid;rport=56930;received=178.5.65.253;branch=z9hG4bK7298384
Call-ID: r9c7kncp97kq4ouuq2gvae
From: sip:[email protected];tag=72ghjb9095
To: sip:[email protected];tag=z9hG4bK7298384
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1623524680/ccd98374e3f1f521f4bdc64e8ee20710”,opaque=“3808f2fe4a0d1ee7”,algorithm=md5,qop=“auth”
Server: FPBX-15.0.17.34(17.9.3)
Content-Length: 0

<— Received SIP request (827 bytes) from WSS:178.5.65.253:56930 —>
REGISTER sip:192.168.179.240 SIP/2.0
Via: SIP/2.0/WSS 33aj8uv7rhf4.invalid;branch=z9hG4bK556176
Max-Forwards: 69
To: sip:[email protected]
From: sip:[email protected];tag=72ghjb9095
Call-ID: r9c7kncp97kq4ouuq2gvae
CSeq: 2 REGISTER
Authorization: Digest algorithm=MD5, username=“21”, realm=“asterisk”, nonce=“1623524680/ccd98374e3f1f521f4bdc64e8ee20710”, uri=“sip:192.168.179.240”, response=“4a99e98f486175a4c8dd7437d6759ddb”, opaque=“3808f2fe4a0d1ee7”, qop=auth, cnonce=“0khajpm3nk0t”, nc=00000001
Contact: sip:[email protected];transport=ws;+sip.ice;reg-id=1;+sip.instance=“urn:uuid:02cd1e43-624a-471b-9e41-c994a803712c”;expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: path,gruu,outbound
User-Agent: JsSIP 3.7.4
Content-Length: 0

<— Transmitting SIP response (431 bytes) to WSS:178.5.65.253:56930 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS 33aj8uv7rhf4.invalid;rport=56930;received=178.5.65.253;branch=z9hG4bK556176
Call-ID: r9c7kncp97kq4ouuq2gvae
From: sip:[email protected];tag=72ghjb9095
To: sip:[email protected];tag=z9hG4bK556176
CSeq: 2 REGISTER
Date: Sat, 12 Jun 2021 19:04:40 GMT
Contact: sip:[email protected]:56930;transport=ws;expires=599
Expires: 600
Server: FPBX-15.0.17.34(17.9.3)
Content-Length: 0

<— Transmitting SIP request (463 bytes) to WSS:178.5.65.253:56930 —>
OPTIONS sip:[email protected]:56930;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.168.179.240:8089;rport;branch=z9hG4bKPje6264512-7bbb-4735-84c1-e05828c67467;alias
From: sip:21@7b5a1967c24c;tag=f3f03dd5-3291-4ae5-bc7b-75c7670273a2
To: sip:[email protected]
Contact: sip:21@7b5a1967c24c:5060;transport=ws
Call-ID: 2c191ded-1870-4d19-b7fc-d2fcc7d40084
CSeq: 6503 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-15.0.17.34(17.9.3)
Content-Length: 0

^

sascha005 · June 12, 2021, 7:21pm

Hm, i see the line SIP/2.0 401 Unauthorized, but why. If I, for example, remove the password from doorpi-card for testing purpose, it get’s not registered in the dev console. So I assume that the authentication works so far…

sascha005 · June 12, 2021, 7:28pm

an this is the asterisk log if i try to call the extension of the lovelace card :

root@7b5a1967c24c:/# asterisk -r
Asterisk 17.9.3, Copyright (C) 1999 - 2018, Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Please note that this version of Asterisk no longer receives bug fixes.
Consult the following URL for Asterisk version support status information:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
=========================================================================
Connected to Asterisk 17.9.3 currently running on 7b5a1967c24c (pid = 1253)
<--- Received SIP request (1176 bytes) from UDP:192.168.179.101:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.179.101:5060;branch=z9hG4bK.FR01oO0Vr;rport
From: "100" <sip:[email protected]>;tag=-CyIqFW8U
To: sip:[email protected]
CSeq: 20 INVITE
Call-ID: jlrtrlLnBH
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 521
Contact: <sip:[email protected];transport=udp>;expires=3599;+sip.instance="<urn:uuid:4576de88-4db4-0048-be80-8ecbe2a31c83>"
User-Agent: Linphone Desktop/4.2.5 (Windows 10 Version 2009, Qt 5.14.2) LinphoneCore/4.4.19

v=0
o=100 266 534 IN IP4 192.168.179.101
s=Talk
c=IN IP4 192.168.179.101
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 18 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

<--- Transmitting SIP response (470 bytes) to UDP:192.168.179.101:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.179.101:5060;rport=5060;received=192.168.179.101;branch=z9hG4bK.FR01oO0Vr
Call-ID: jlrtrlLnBH
From: "100" <sip:[email protected]>;tag=-CyIqFW8U
To: <sip:[email protected]>;tag=z9hG4bK.FR01oO0Vr
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1623525935/8ffeb215034f36f3b621b6e5b62f14fc",opaque="512e064e4dc74484",algorithm=md5,qop="auth"
Server: FPBX-15.0.17.34(17.9.3)
Content-Length:  0


<--- Received SIP request (393 bytes) from UDP:192.168.179.101:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.179.101:5060;branch=z9hG4bK.FR01oO0Vr;rport
Call-ID: jlrtrlLnBH
From: "100" <sip:[email protected]>;tag=-CyIqFW8U
To: <sip:[email protected]>;tag=z9hG4bK.FR01oO0Vr
Contact: <sip:[email protected];transport=udp>;expires=3599;+sip.instance="<urn:uuid:4576de88-4db4-0048-be80-8ecbe2a31c83>"
Max-Forwards: 70
CSeq: 20 ACK


<--- Received SIP request (1454 bytes) from UDP:192.168.179.101:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.179.101:5060;branch=z9hG4bK.U4BuBoqZc;rport
From: "100" <sip:[email protected]>;tag=-CyIqFW8U
To: sip:[email protected]
CSeq: 21 INVITE
Call-ID: jlrtrlLnBH
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 521
Contact: <sip:[email protected];transport=udp>;expires=3599;+sip.instance="<urn:uuid:4576de88-4db4-0048-be80-8ecbe2a31c83>"
User-Agent: Linphone Desktop/4.2.5 (Windows 10 Version 2009, Qt 5.14.2) LinphoneCore/4.4.19
Authorization:  Digest realm="asterisk", nonce="1623525935/8ffeb215034f36f3b621b6e5b62f14fc", algorithm=md5, opaque="512e064e4dc74484", username="100",  uri="sip:[email protected]", response="ff2fbfcd00c95f20e1346ec8ca2122d8", cnonce="1bhCkgfAqCVmXq7D", nc=00000001, qop=auth

v=0
o=100 266 534 IN IP4 192.168.179.101
s=Talk
c=IN IP4 192.168.179.101
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 18 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

<--- Transmitting SIP response (296 bytes) to UDP:192.168.179.101:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.179.101:5060;rport=5060;received=192.168.179.101;branch=z9hG4bK.U4BuBoqZc
Call-ID: jlrtrlLnBH
From: "100" <sip:[email protected]>;tag=-CyIqFW8U
To: <sip:[email protected]>
CSeq: 21 INVITE
Server: FPBX-15.0.17.34(17.9.3)
Content-Length:  0


<--- Transmitting SIP response (374 bytes) to UDP:192.168.179.101:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.179.101:5060;rport=5060;received=192.168.179.101;branch=z9hG4bK.U4BuBoqZc
Call-ID: jlrtrlLnBH
From: "100" <sip:[email protected]>;tag=-CyIqFW8U
To: <sip:[email protected]>;tag=0593eb64-c591-4aa9-ac13-ec6ed25813ff
CSeq: 21 INVITE
Server: FPBX-15.0.17.34(17.9.3)
Reason: Q.850;cause=34
Content-Length:  0


<--- Received SIP request (412 bytes) from UDP:192.168.179.101:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.179.101:5060;branch=z9hG4bK.U4BuBoqZc;rport
Call-ID: jlrtrlLnBH
From: "100" <sip:[email protected]>;tag=-CyIqFW8U
To: <sip:[email protected]>;tag=0593eb64-c591-4aa9-ac13-ec6ed25813ff
Contact: <sip:[email protected];transport=udp>;expires=3599;+sip.instance="<urn:uuid:4576de88-4db4-0048-be80-8ecbe2a31c83>"
Max-Forwards: 70
CSeq: 21 ACK