SIP client card, as intercom

Weird, I can see it on the repos in HA.

Try hitting the reload button in the upper-right corner.
image

Perfect , i see it now :+)

I donā€™t work with personā€™s in HA, all my tablets/phones are logged in with same user , so sip card is going to use same sip numberā€¦ In pjsip there is s max contact settingā€¦ Will it work also for chansip? So it can register multiple times? Or is now the add-on registering instead of sip card?
Eventually all sip cards on all HA clients will go in ringing state, first one who picks up should have the call, right?

Thank you again for all your great work. Should the Asterisk addon work behind a reverse Proxy. I use NGINX proxy manager. When I start the addon it starts, but after a minute or two it stops.
The log is below;

![image|596x500](upload://vLHdTugObSVgMomrytamLPZNjgB.png)

we dont see the logs

Sorry try again.

[s6-init] making user provided files available at /var/run/s6/etc...exited 0.
[s6-init] ensuring user provided files have correct perms...exited 0.
[fix-attrs.d] applying ownership & permissions fixes...
[fix-attrs.d] done.
[cont-init.d] executing container initialization scripts...
[cont-init.d] done.
[services.d] starting services
[services.d] done.
[17:25:22] INFO: Checking config files...
[17:25:22] INFO: Creating certificate...
Generating a RSA private key
...............................................................................................++++
........................................................................................++++
writing new private key to '/etc/asterisk/keys/asterisk.key'
-----
[17:25:27] INFO: Configuring Asterisk...
[17:25:27] INFO: Starting Asterisk...
Seeding global EID '02:42:ac:1e:21:05' from 'eth0' using 'siocgifhwaddr'
 
[17:25:37] INFO: Name/username Host Dyn Forcerport Comedia ACL Port Status Description 101/0d6b63bbd69b4f74a64e2 (Unspecified) D Auto (No) No 0 Unmonitored 102/andrew (Unspecified) D Auto (No) No 0 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]
 
[17:25:42] INFO: HTTP Server Status: Prefix: Server: Asterisk/18.2.2 Server Enabled and Bound to 0.0.0.0:8088 HTTPS Server Enabled and Bound to 0.0.0.0:8089 Enabled URI's: /httpstatus => Asterisk HTTP General Status /phoneprov/... => Asterisk HTTP Phone Provisioning Tool /ari/... => Asterisk RESTful API /ws => Asterisk HTTP WebSocket Enabled Redirects: None.
 
/run.sh: line 128: SECRET: unbound variable
[cmd] /run.sh exited 1
[cont-finish.d] executing container finish scripts...
[cont-finish.d] done.
[s6-finish] waiting for services.
[s6-finish] sending all processes the TERM signal.

Hi everybody.

I am already trying to get this integration working with the FreePBX solution.

Although I was having issues at not being able to enable DTLS I would now just go on and use the new alpha Asterix add-on instead.

But first, thereā€™s some confusion in my mind about.

  1. I installed the browser-mod, but I get right now 4 device_ids instead of just 1 so which one to use?

  2. what IP to enter in the aor line? My Amcrest AD410 video doorbell IP or what?

for sip card, the ipā€™s you enter, needs to be the PBX itselft , not the sip clientā€¦ i was confused too
i had also indeed issues enabling dtls, it gave me an error when moving from chan to pjsip
so i started creating a pjsip account, did all the settings needed, an last step, i converted too chanā€¦ that worked for meā€¦ but now iā€™m going to move too to the new add-on, not yet tested yet

1 Like

tried running add-n too, but it stops indeed, same log for me
maybe we need to configure something first, lets wait for some extra info from owner

[s6-init] making user provided files available at /var/run/s6/etc...exited 0.
[s6-init] ensuring user provided files have correct perms...exited 0.
[fix-attrs.d] applying ownership & permissions fixes...
[fix-attrs.d] done.
[cont-init.d] executing container initialization scripts...
[cont-init.d] done.
[services.d] starting services
[services.d] done.
[11:35:51] INFO: Checking config files...
[11:35:51] INFO: Creating certificate...
Generating a RSA private key
.................................................................................................................++++
................................................................++++
writing new private key to '/etc/asterisk/keys/asterisk.key'
-----
[11:35:53] INFO: Configuring Asterisk...
[11:35:53] INFO: Starting Asterisk...
Seeding global EID '02:42:ac:1e:21:01' from 'eth0' using 'siocgifhwaddr'
 
[11:36:03] INFO: Name/username Host Dyn Forcerport Comedia ACL Port Status Description 101/fabio (Unspecified) D Auto (No) No 0 Unmonitored 102/vanessa (Unspecified) D Auto (No) No 0 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]
 
[11:36:08] INFO: HTTP Server Status: Prefix: Server: Asterisk/18.2.2 Server Enabled and Bound to 0.0.0.0:8088 HTTPS Server Enabled and Bound to 0.0.0.0:8089 Enabled URI's: /httpstatus => Asterisk HTTP General Status /phoneprov/... => Asterisk HTTP Phone Provisioning Tool /ari/... => Asterisk RESTful API /ws => Asterisk HTTP WebSocket Enabled Redirects: None.
 
/run.sh: line 128: SECRET: unbound variable
[cmd] /run.sh exited 1
[cont-finish.d] executing container finish scripts...
[cont-finish.d] done.
[s6-finish] waiting for services.
[s6-finish] sending all processes the TERM signal.

how would i upgrade to 0.2 asterisk? Whic repo d i have to use ? I am totally confusedā€¦ maybe u can update frst post ?

Hi

add this to your repostorys in your addon section

then you dont need to download manual new updates, it will prompt you for updates

OK addoin inistalled. I need to rad the docs even if not really easyā€¦ maybe yu can insert code instead of screenshots for copy paste?

Now I use this config in my Addon:

auto_add: true
custom_extensions:
  - type: chan_sip
    name: doorbell
    extension: '201'

And how to stup the card? Where to define the password etc? I am new to this all sorry for dumb question

docs will be updated with detailed instructions, wait a bit :slight_smile:
i have issues starting my addon, it stops after a few seconds, think there is something wrong with it, or we need to add some extra info to it

This error:

[s6-init] making user provided files available at /var/run/s6/etc...exited 0.
[s6-init] ensuring user provided files have correct perms...exited 0.
[fix-attrs.d] applying ownership & permissions fixes...
[fix-attrs.d] done.
[cont-init.d] executing container initialization scripts...
[cont-init.d] done.
[services.d] starting services
[services.d] done.
[15:43:02] INFO: Checking config files...
[15:43:02] INFO: Creating certificate...
Generating a RSA private key
................................................++++
......................................++++
writing new private key to '/etc/asterisk/keys/asterisk.key'
-----
[15:43:03] INFO: Configuring Asterisk...
[15:43:03] INFO: Starting Asterisk...
Seeding global EID '02:42:ac:1e:21:0f' from 'eth0' using 'siocgifhwaddr'
 
[15:43:13] INFO: Name/username Host Dyn Forcerport Comedia ACL Port Status Description 101/thorsten_frohlich (Unspecified) D Auto (No) No 0 Unmonitored 102/emilie (Unspecified) D Auto (No) No 0 Unmonitored 103/evan (Unspecified) D Auto (No) No 0 Unmonitored 104/rose (Unspecified) D Auto (No) No 0 Unmonitored 4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 4 offline]
 
[15:43:18] INFO: HTTP Server Status: Prefix: Server: Asterisk/18.2.2 Server Enabled and Bound to 0.0.0.0:8088 HTTPS Server Enabled and Bound to 0.0.0.0:8089 Enabled URI's: /httpstatus => Asterisk HTTP General Status /phoneprov/... => Asterisk HTTP Phone Provisioning Tool /ari/... => Asterisk RESTful API /ws => Asterisk HTTP WebSocket Enabled Redirects: None.
 
/run.sh: line 128: SECRET: unbound variable
[cmd] /run.sh exited 1
[cont-finish.d] executing container finish scripts...
[cont-finish.d] done.
[s6-finish] waiting for services.
[s6-finish] sending all processes the TERM signal.
```
?

indeed, it doesnt start, wait for him :slight_smile:

Something with secret wellā€¦ i wait :slight_smile:

Found it, stupid error. It is now fixed in 0.2.1.

Sorry

I dont think its possible to register a chan_sip extension more then once.
pjsip is not working currently in Asterisk. This is a known bug and is not yet fixed in the main branch.

Screenshot of my Alpine VM for testing:

I wanted to use users, but it is not possible to get the users in a add-on. So I had to use persons.

ah ok, so not possible to run multiple instances with same person that are using the card then?

The custom extensions is for the devices you want to call to. So you dont need a password for those.

If auto_add is set to true, the add-on creates a extension for every person. You can see the extension, secret and user in the entity.