Topic: SIP trunk rings in, but not transferred to extension

[bins]

I am installing the latest release on a new server, to upgrade our current system.  All appears ok:

All appears ok -
trunks setup and registering
extension setup and ringing each other
extensions able to ring out via sip trunk

However, when you call in to the SIP trunk, the line rings, but it is not routing through to an extension.

Looking at the log, the message "chan_sip.c: Call from '<trunk account>' to extension '<trunk account>' rejected because extension not found" is the only related error I can see,
res_config_pgsql.c: Postgresql RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info.
chan_zap.c: Unable to reconfigure channel '1'
chan_zap.c: Reload of chan_zap.so is unsuccessful!

athough I dont think these are related (a card has been probed and set up) .

The router is set up to allow the right traffic through (all works on main system).

Anyone with some thoughts?

[David Harper]

Those chan_zap errors might indicate some issue with the Zaptel driver.

[SARK devs]

This can happen as a result of a missing DDI (DiD) entry.  It all depends how the SIP INVITE is constructed by your carrier.  Look at the SIP invite to see what the URI is.  (Invite somevalue@yourdomain).  You can see it by running a SIP trace in Asterisk or by using Tshark or whatever.

What happens is that the SIP trunk is set up with account number but the invite is referencing the DDI, or vice versa.  The cure is simply to add a PTT_DiD trunk entry for the missing value and then use that to do your routing.

You also have a zap config problem which is not related to the sip call issue.  Suggest you restart your system (reboot) and use PCI Cards panel to reconfig the zap lines.

The Postgress message is benign and can be ignored.

Best

S