Hello i just config the asterisk server but when i'm caling the oder phone it write this in the CLI

Connected to Asterisk 16.28.0~dfsg-0+deb11u3 currently running on tp (pid = 5317)
tp*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
1000/1000                                   D  Yes        Yes            56813    Unmonitored
2000/2000                                   D  Yes        Yes            55916    Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
  == Using SIP RTP CoS mark 5
[Dec 15 15:52:12] NOTICE[5625][C-00000007]: chan_sip.c:26826 handle_request_invite: Call from '1000' ( to extension '2000' rejected because extension not found in context 'internal'.

chan_sip is obsolete.

Asterisk 16.28.0 is unsupported and has known security vulnerabilities.

There are no lines in your log resulting from an outgoing call.

All but one are the result of a manually entered CLI command, and the final one, which is self explanatory, is the result of an incoming call. Asterisk has received a request to call extension 2000 from a peer whose From header contains `“1000” and whose address is 192,168.1.100:56813. The peer definition that matches either 1000 or that IP address, or the general section, if using allowguest,is configured to use the context “internal”. There is no priority 1 line for extension 2000 in extensions.conf, or a file it includes, in the internal context.

Also, neither 1000, nor 2000, have successfully registered, so outgoing calls to either will fail, but with a different message, as Asterisk doesn’t know how to send calls to them.

Note that sip show peers shows devices, not extensions, even though FreePBX conflates the two.

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Hello my friend, I would suggest, that you look into these books:

  1. The Future of Telephony, 2nd Edition by Jim Van Meggelen
  2. Building Telephony Systems with Asterisk by David Gomillion
  3. Switching to VoIP by Ted Wallingford
  4. Asterisk Gateway Interface 1.4 and 1.6 Programming by Nir Simionovich
    There are a couple others that I did not add. It does seem like you want to learn, So This will get you going in the right direction.

so what i must to do ? for fix the problem ?

thx for the book

Define extension 2000 in your extensions.conf, which does what you want it to do (presumably to dial SIP/2000). This is Asterisk 101 stuff and there are many examples available. There will be some in the sample configuration, although they may be unnecessarily complicated, because they are trying illustrate more complex uses.

Debug the signalling between the phones and Asterisk to find out why they are not registering. Start with “sip set debug on”, to see if the registration is arriving but failing, but you may need to look outside of Asterisk, if Asterisk is not receiving attempts. It is possible that you didn’t leave enough time for the phones to register.

Consider whether a telephony toolkit is right for you, or whether you should be using a fully supported commercial product, or, a GUI configured PABX based on Asterisk, like FreePBX. None of these are likely to give easy solutions to the failure to register.

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