SIP Invite to SIP Server arriving without user part (to)

[vmc]

I'm trying to make use of a SIP trunk provider, without any SBC in front so have configured a SIP Trunk on SIP Server with the contact details and authentication stuff for the provider.

It seems to register fine, and I"m getting OPTIONS messages, and can make outbound calls.

However inbound calls all arrive without the user part of the invite, eg

INVITE: sip:ipaddress:5060 - not INVITE: sip:dnis@ipaddress:5060

Is this expected and why people are using SBCs?  Can we not get a SIP Trunk to terminate directly into SIP Server?

[gen_den]

Its not a expected. You should get user part in SIP INVITE.
Also  SBC is recommended to use in production environment , it acts as a Firewall.

which provide SIP trunk are you using?

[vmc]

This is for a training lab for myself, thus no SBC.

SIP Trunk provider is a local one in my country - ie so not one of the larger ones like Skype etc.

[gen_den]

Ok for Lab purpose you can connect provider directly to SIP server.
I know the configuration for SIP server integration with Twilio Provider.

May you can post your configuration and SIP messages.

[vmc]

Closed it down for now.  But I started using settings for the trunk someone posted on here for Skype Manager.

Mind if I ask you to please post your trunk configuration (less authentication details)?

I'll compare when I start it up again tomorrow.

I've had to use a . dial plan rule for now but it means I'm locked into a single inbound number.

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[Kubig]

As is mentioned in SIP dep guide :

Table 79: Enabling SIP Authentication
Objective Key Procedures and Actions
1. Configure the inbound
endpoint.
In the endpoint DN, configure the following options:
• authenticate-requests—Set this option to the value INVITE.
Incoming INVITE requests to this DN will result in an
authentication challenge.
• password—Enter the password for realm authentication. The
INVITE sent in response to the challenge must include this
password, otherwise the call will not be authorized.
2. Configure the Trunk. In the Trunk DN, in the AuthClient section, configure the following
options:
• username—Enter a username to be used in the response to the
Digest challenge.
• password—Enter the password to be used in generating the
response.

[vmc]

Got all that

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[vmc]

What if the inbound end point is actually a dial plan first that could then go to one of many route points?

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[Kubig]

Do not see the issue in described scenario..What is not clear for you about SIP authentication?

[vmc]

If I create a dial plan:

. =>12345

The call gets delivered to dn 12345 fine.  I can also make outbound calls fine.  DN 12345 does  it have that authentication configuration on it.  Only the Trunk DN has that.

From what I can tell this means I'm authenticating fine, but somewhere in the path from SIP Trunk provider to SIP Server I'm not getting a fully formed SIP INVITE message.

Routing an INVITE without a dnis in the To section means I'm currently limited to only one external DDI and cannot route to different route points using dial plans.

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[Kubig]

If the incoming INVITE message does not contain user-part in SIP headers to identify the DNIS, you have to consult with SIP provider support as the issue is out of Genesys.