I need an extension without numbers

FreePBX Endpoints
1

Hello!

I have a doorbell camera that does a sip registration with a user like 10101011011 or whatever.

The camera has an intercom button on it.

When the button is pressed, the camera tries to place a phone call to an extension called something like “managementCentre” instead of like “17” or “1372”

However, the freePBX GUI won’t let me create a SIP (or custom) extension containing any non-digit characters.

Can I persuade Freepbx to create and end point that will accept calls sent to “SomeString” or do I need to use some other software?

Thanks!

2

Asterisk can do this. I’m not actually a FreePBX user, but if I understand the customisation mechanisms correctly, you can append the following to /etc/asterisk/extensions_custom.conf, creating it if necessary:

[from-internal](+)
exten => managementCentre,1,Goto(XXXX,1)

where XXXX is an extension number that FreePBX will accept.

4

@FreakWent no need for custom dialplan. For the extension you want the door phone to dial, populate the ‘SIP Alias’ on the advanced tab with the text name you want.

@david55 FYI the context ‘from-internal-custom’ exists for this purpose, no need to (+) from-internal.

@dicko Wat? That won’t work.

5

oops, brain fart.

1 Like
6

Please share logs while dial is executed from doorbell,
most probably it will be failing,
We can create a custom dialplan for it and utilize the dynamic routes to handle via ${CALLERID(num)} or ${CALLERID(all)} then route to specific destination.

7

Well, this is the most help I’ve probably had from any forum post, ever!

Thanks so much everyone!

I first did david55’s suggestion, but it didn’t work, so I reversed it.

Then I added the SIP alias as lgaetz suggested.

Then freepbx complained that extensions.conf had changed, so I did

fwconsole reload

which didn’t help so I did

fwconsole ma refreshsignatures

Now the error has gone.

But now my system is broken!

Even though pjsip.endpoint.conf has this in it:

[17]
type=endpoint
aors=17
auth=17-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=ulaw,alaw,gsm,g726,g722
context=from-internal

and even though I get this at the console:

CLI> pjsip list endpoints

Endpoint: <Endpoint/CID…> <State…> <Channels.>

Endpoint: 10010100000/10010100000 Not in use 0 of inf
Endpoint: 17/17 Not in use 0 of inf
Endpoint: anonymous Unavailable 0 of inf

Now I can’t even dial by number – it’s worse! I used to be able to phone myself, and it would try to connect, but now it can’t even do that…

[2022-05-30 11:17:09] NOTICE[9308]: res_pjsip_session.c:2337 new_invite: Call from ‘17’ (UDP:192.168.0.79:5060) to extension ‘17’ rejected because extension not found in context ‘from-internal’.

Anyway I’ll make another ext. and try that…

I created nineteen :

[2022-05-30 11:29:47] NOTICE[9308]: res_pjsip_session.c:2337 new_invite: Call from ‘19’ (UDP:192.168.0.212:55499) to extension ‘17’ rejected because extension not found in context ‘from-internal’.
[2022-05-30 11:29:54] NOTICE[9308]: res_pjsip_session.c:2337 new_invite: Call from ‘17’ (UDP:192.168.0.79:5060) to extension ‘19’ rejected because extension not found in context ‘from-internal’.

Thanks again!!

8

Not related to your current problem, but you are likely to get other problems, especially for calls to the extension, if you don’t precede this by disallow=all.

If you hadn’t said that you had reversed my suggestion, I would have wondered if you missed out the “(+)”.

9

Thanks! The text I posted what what freepbx automatically generated so I don’t know why it doesn’t work. I think maybe I should somehow “reset” freepbx and start again now :frowning:

10

Silly question but why don’t you reset the camera and change what it want’s to register against? You are doing a lot of changes in FreePBX to accommodate not changing camera settings.

11

@kenn10 the camera has those settings hard-wired and they can’t be changed, I only discovered “ManageCenter” by checking the traffic & logs to begin with. Besides, I can’t even get asterisk to attempt to place a call…

This thing is driving me absolutely bonkers. Freepbx/asterisk seems to be massive overkill for a system with two end points, but here we are. I have spent so many hours getting one thing to dial another thing.

Endpoints:
Endpoint: 10010100000/10010100000 Not in use 0 of inf
Endpoint: 19/19 Not in use 0 of inf

So let’s try dialling:

#asterisk -r

hostname*CLI> originate SIP/19@from-internal extension 10010100000@from-internal

[2022-06-19 12:36:53] VERBOSE[8946] netsock2.c: Using SIP RTP TOS bits 184
[2022-06-19 12:36:53] VERBOSE[8946] netsock2.c: Using SIP RTP CoS mark 5
[2022-06-19 12:36:54] VERBOSE[9086] dial.c: Called 19@from-internal
[2022-06-19 12:37:26] WARNING[5167] chan_sip.c: Retransmission timeout reached on transmission 77032124421583e4532e14a402d2972f@:5160 for seqno 102 (Critical Request) – See SIP Retransmissions - Asterisk Project - Asterisk Project Wiki
Packet timed out after 32000ms with no response

exit

nc -v -u -z -w 3 5160

Ncat: Version 7.50 ( Ncat - Netcat for the 21st Century )
Ncat: Connected to :5160.
Ncat: UDP packet sent successfully
Ncat: 1 bytes sent, 0 bytes received in 2.01 seconds.

If I actually call from the SIP client 19 back to 10010100000 I get, in the log:

[2022-06-19 12:33:00] NOTICE[5137] res_pjsip_session.c: Call from ‘19’ (UDP::44002) to extension ‘10010100000’ rejected because extension not found in context ‘from-internal’.

And of course, calling the other way still gives the evil:

[2022-06-19 12:45:26] NOTICE[5137] res_pjsip_session.c: Call from ‘10010100000’ (UDP::5060) to extension ‘ManageCenter’ rejected because extension not found in context ‘from-internal’.

What command can I type to get asterisk to tell me the extensions that it thinks exist in the from-internal context?

12

It’s working! All I needed was a little more persistence and a little less complaining.

@lgaetz had the right idea. Thanks all!

13

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