SIP trunk cant register

Asterisk Asterisk SIP
1

Hello everyone, I hope you are well

I’m starting out in the world of asterisk, I uploaded my first asterisk in the asterisk-18.14.0 version.

I’m using a SIP trunk, my operator gave me the registration parameters and I configured it according to what was proposed to me.

I will show my sip.conf code

sip.conf

[trunk-01]
username=myuser
type=peer
secret=mypass
Host=myserver.voice.com.br
outboundproxy=myserver.voice.com.br
fromuser=3138501010
fromdomain=mydomain.voice.com.br
insecure=port,invite
disallow=all
allow=alaw
context=interno

register => myuser@mydomain.voice.com.br:mypass:myuser@myserver.voice.com.br/myuser

Output

centos7-template*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
0 SIP registrations.

I would like to understand what could be happening.

I have ping to my logging server, I don’t have a firewall blocking any ports or connections. My registry server is responding to nping port 5060/udp

2

Don’t use chan_sip; it is no longer supported.

Don’t use Asterisk 18; it is two months into its security fixes only, final, year.

I hope you haven’t quoted your real password!

Generally, to debug something like this, you need to enable the full log and turn on protocol logging, using the CLI. For chan_sip, that is “sip set debug on”, for the chan_pjsip, that you should be using, it is “pjsip set logger on”.

1 Like
3

On Thursday 19 December 2024 at 21:30:19, Br3niiN via Asterisk Community
wrote:

I’m starting out in the world of asterisk, I uploaded my first asterisk in
the asterisk-18.14.0 version.

That version is already is “Security fix only” mode - you should really be
starting with version 20 or 22.

https://docs.asterisk.org/About-the-Project/Asterisk-Versions/

I will show my sip.conf code

You definitely should not be using this any more - chan_sip is unsupported,
deprecated in the version you have, and no longer included in current versions
of Asterisk.

Please start with a currently-fully-supported version of Asterisk, and a
supported SIP channel driver (PJSIP) so that we can give better advice on any
problems you run into (many people here no longer have recollection or
familiarity with using chan_sip).

Welcome to Asterisk, though :slight_smile:

Antony.

–
The Magic Words are Squeamish Ossifrage.

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4

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