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arnonym / ha-plugins

Home-Assistant SIP Gateway
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Sending DTMF not working, Inconsistent responses #73

Open Pytonballoon810 opened 10 months ago

Pytonballoon810 commented 10 months ago

Using FritzBox 7590 on OS version 7.57, Addon on Version 3.2, HA Core 2023.11.3, Supervisor 2023.11.6, OS 11.2

I have an automation (and script) set up to first, dial a number (script) then wait until a webhook is called, then sending a DTMF job to press #1. Lastly the automation ends the call.

Now the problem is that if I am using the addon to call my mobile phone internally (via the 620) it just works. You may recognize that that is indeed not a problem. And you would be correct. My phone rings, the timeout is set to 15s, if I answer in those 15s I can hear how the caller (homeassistant) is pressing both buttons and then ending the call. The Problem arrises however when I change the internal call number to the 2 (my intercom). The logs don't show any errors (at least I think) but the intercom station is still not able to open the door. (I tested and the exact same prosedure is working when calling internally from my mobile phone via the Fritz!FON app or using a telephone.)

Here is my HA-SIP config:

sip:
  enabled: true
  registrar_uri: sip:fritz.box
  id_uri: sip:homeassistant@fritz.box
  realm: "*"
  user_name: homeassistant
  password: password
  answer_mode: listen
  settle_time: 1
  incoming_call_file: ""

My script for dialing the call:

alias: DialDoorOpenener
sequence:
  - service: hassio.addon_stdin
    data:
      addon: c7744bff_ha-sip
      input:
        command: dial
        number: sip:**2@fritz.box
        ring_timeout: 15
        sip_account: 1
        webhook_to_call:
          call_established: "-huv8wswQGAIDUwLNPy3j14pK"
mode: single
icon: mdi:key-chain

My automation for reacting to the sent webhook, pressing the buttons and ending the call:

alias: React to open door call
description: ""
trigger:
  - platform: webhook
    allowed_methods:
      - POST
      - PUT
    local_only: true
    webhook_id: "-huv8wswQGAIDUwLNPy3j14pK"
    id: webhook
condition: []
action:
  - choose:
      - conditions:
          - condition: trigger
            id:
              - webhook
        sequence:
          - delay:
              hours: 0
              minutes: 0
              seconds: 1
              milliseconds: 0
          - service: hassio.addon_stdin
            data:
              addon: c7744bff_ha-sip
              input:
                command: send_dtmf
                number: sip:**2@fritz.box
                digits: "#1"
                method: sip_info
          - service: hassio.addon_stdin
            data:
              addon: c7744bff_ha-sip
              input:
                command: hangup
                number: sip:**2@fritz.box
                sip_account: 1
    default:
      - service: notify.mobile_app_nothing_phone_1
        data:
          message: Nope
mode: single

Here are the logs after running the first script (all that I could export at least since HA doesnt keep them for long):


Signal=1
Duration=180
--end msg--
00:10:28.725           pjsua_core.c  .RX 441 bytes Response msg 200/INFO/cseq=29815 (rdata0x7fa8aa7088) from UDP 192.168.2.1:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.13:5060;rport=5060;branch=z9hG4bKPjnGra0cUdsXb7YeaHeol94nRjb-W4VAxT
From: <sip:homeassistant@fritz.box>;tag=cOOSp.K-dRWkeMw7FR8eWMp3uei3Un1l
To: <sip:**2@fritz.box>;tag=82576A769B7CB897
Call-ID: KpsWo5BMNAw54BGJT8Mep7MEN6Y0PaXk
CSeq: 29815 INFO
User-Agent: AVM FRITZ!Box 7590 154.07.57 (Sep  2 2023)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Content-Length: 0

--end msg--
00:10:28.725           pjsua_call.c  .....Call 2: DTMF sent successfully with INFO
00:10:28.727           pjsua_core.c  .RX 441 bytes Response msg 200/INFO/cseq=29816 (rdata0x7fa8aa7098) from UDP 192.168.2.1:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.13:5060;rport=5060;branch=z9hG4bKPj63uINkZymrG9czKsx0AO.oFb2fPzbUjw
From: <sip:homeassistant@fritz.box>;tag=cOOSp.K-dRWkeMw7FR8eWMp3uei3Un1l
To: <sip:**2@fritz.box>;tag=82576A769B7CB897
Call-ID: KpsWo5BMNAw54BGJT8Mep7MEN6Y0PaXk
CSeq: 29816 INFO
User-Agent: AVM FRITZ!Box 7590 154.07.57 (Sep  2 2023)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Content-Length: 0

--end msg--
00:10:28.727           pjsua_call.c  .....Call 2: DTMF sent successfully with INFO
| 00:10:28.760978 [ ] Got "hangup" command for sip:**2@fritz.box
| 00:10:28.761258 [1] Hang-up.
00:10:28.761           pjsua_call.c  Call 2 hanging up: code=0..
00:10:28.761          pjsua_media.c  .Call 2: deinitializing media..
00:10:28.761          pjsua_media.c  ..
  [CONFIRMED] To: sip:**2@fritz.box;tag=82576A769B7CB897
    Call time: 00h:00m:03s, 1st res in 86 ms, conn in 851ms
    #0 audio iLBC @8kHz, sendrecv, peer=192.168.2.1:7082
       SRTP status: Not active Crypto-suite: 
       ICE role: Unknown, state: Candidate Gathering, comp_cnt: 2
       RX pt=99, last update:00h:00m:02.733s ago
          total 103pkt 5.1KB (9.2KB +IP hdr) @avg=13.2Kbps/23.8Kbps
          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     :   0.125   1.646   4.125   3.875   0.694
       TX pt=99, ptime=30, last update:00h:00m:01.388s ago
          total 21pkt 1.0KB (1.8KB +IP hdr) @avg=2.7Kbps/4.8Kbps
          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     :   7.500   7.500   7.500   7.500   0.000
       RTT msec      :   0.000   0.000   0.000   0.000   0.000
00:10:28.762          pjsua_media.c  ...Media stream call02:0 is destroyed
00:10:28.762                icetp00  ..Stopping ICE, reason=media stop requested
00:10:28.762       srtp0x7fa7c5e830  ..Destroying SRTP transport
00:10:28.762                icetp00  ..Destroying ICE transport
00:10:28.762          ice_session.c  ..ICE session 0x7fa7eac5d8 destroyed
00:10:28.762                icetp00  ..ICE stream transport 0x7fa7c2d268 destroyed
00:10:28.763                icetp00  ..ICE transport destroyed
00:10:28.763       srtp0x7fa7c5e830  ..SRTP transport destroyed
| 00:10:28.763252 [1] Call disconnected
| 00:10:28.763508 [ ] Calling webhook sip_call_webhook_id with data {'event': 'call_disconnected', 'caller': 'sip:**2@fritz.box', 'parsed_caller': '**2', 'sip_account': 1}
| 00:10:28.784424 [ ] Webhook response 200 b''
| 00:10:28.785529 [ ] Remove from state: sip:**2@fritz.box
00:10:28.785           pjsua_core.c  ....TX 368 bytes Request msg BYE/cseq=29817 (tdta0x7fa7c307e8) to UDP 192.168.2.1:5060:
BYE sip:EAA1987F499C962AB343AA21CCEE480@192.168.2.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.13:5060;rport;branch=z9hG4bKPjW2ySd.WGXf2Yx1WiaLnou4TjgSX.ta9y
Max-Forwards: 70
From: sip:homeassistant@fritz.box;tag=cOOSp.K-dRWkeMw7FR8eWMp3uei3Un1l
To: sip:**2@fritz.box;tag=82576A769B7CB897
Call-ID: KpsWo5BMNAw54BGJT8Mep7MEN6Y0PaXk
CSeq: 29817 BYE
Content-Length:  0

--end msg--
00:10:28.792           pjsua_core.c  .RX 724 bytes Response msg 200/BYE/cseq=29817 (rdata0x7fa8aa70a8) from UDP 192.168.2.1:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.13:5060;rport=5060;branch=z9hG4bKPjW2ySd.WGXf2Yx1WiaLnou4TjgSX.ta9y
From: <sip:homeassistant@fritz.box>;tag=cOOSp.K-dRWkeMw7FR8eWMp3uei3Un1l
To: <sip:**2@fritz.box>;tag=82576A769B7CB897
Call-ID: KpsWo5BMNAw54BGJT8Mep7MEN6Y0PaXk
CSeq: 29817 BYE
X-RTP-Stat: CS=0;PS=56;ES=195;OS=2800;SP=0/0;SO=0;QS=-;PR=52;ER=195;OR=6010;CR=0;SR=0;QR=-;PL=0,0;BL=0;LS=0;RB=0/0;SB=-/-;EN=iLBC-30,G722;DE=iLBC-30,G722;JI=60,31;DL=1,1,1;IP=192.168.2.1:7082,192.168.2.13:4030
X-RTP-Stat-Add: DQ=4;DSS=0;DS=0;PLCS=8704;JS=64
X-SIP-Stat: DRT=0;IR=0
User-Agent: AVM FRITZ!Box 7590 154.07.57 (Sep  2 2023)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Content-Length: 0

--end msg--
00:10:29.763            pjsua_aud.c  Closing sound device after idle for 1 second(s)
00:10:29.763            pjsua_aud.c  .Closing null sound device..
arnonym commented 10 months ago

Hi @Pytonballoon810,

did I understand correctly, that the Intercom does not react to the dialtones? Did you try the other methods of sending DTMF via SIP, namely "in_band" and "rfc2833"?

Pytonballoon810 commented 10 months ago

The intercom is supposed to activate the door opener upon typing #1. It works when typing manually but not with the SIP client. I already tried all three methods. I also think it worked about 6-10 months ago. Then it stopped working without making any changes... No idea why.

arnonym commented 10 months ago

Did you try with a longer delay?

Pytonballoon810 commented 10 months ago

Did you try with a longer delay?

I originally had a 3 sec delay between both typing the digits and ending the call. Still didnt work

Pytonballoon810 commented 9 months ago

Ok so I fixed it now. I put each character in one by one in two seperate stdin calls with a 500ms delay between each digit. Maybe you wanna fix that in your code (or I find the time to fix it myself.) Thanks for the help anyways :)

arnonym commented 9 months ago

Nice you figured that out! ha-sip is only calling the send DTMF function of pjsip, and for rfc2833 and sip_info methods it's not possible to set a delay between tones. You can only set the duration of each tone. For in_band you can control both. It would be interesting if increasing the tone length would also work with your intercom. In the code you would just need to increase the value of DEFAULT_DTMF_ON, if you're able to test that yourself.