Req URI and To header contain trunk name

Has anyone else seen the following and know how to get around it:

INVITE sip:+44XXXXXXXXXX%40192.XXX.XXX.XXX%[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.XXX.XXX.XXX:5060;branch=z9hG4bK36ff634f;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as285aa2fa
To: <sip:+44XXXXXXXXXX%40192.XXX.XXX.XXX%[email protected]>

The URI and To user parts contain the target ip and port, which isnt expected.

I might be wrong, but it appears this is the trunk name.

Which channel driver are you using? How did you configure it?

Not sure what you mean by channel driver.
This is legacy chan_sip on FreePBX15.

chan_sip is the channel driver. Awaiting the configuration.

Outgoing trunk config:
host=192.XXX.XXX.XXX
qualify=yes
type=peer
sendrpid=pai
trustrpid=yes
context=from-trunk

I think we need the full log for the call, as it looks like the strange URI could have come on the inbound leg, or be the result of bad customisation of the dialplan.

Thanks for your assistance David, on reboot of the device (Note: no changes made to config) the issue is no longer persisting. Very odd!

This is at least the fourth time that this bug has bitten. Unfortunately, none of the victims took the effort to reproduce and report it.

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I searched high and low for tickets on this.
If I see the issue again, what logs would the community be interested in?

If there is a bug in chan_sip, I do not think it will be addressed. Better to reconfigure your trunk for PJSIP.

2 Likes

You can still grab a call trace. If you can, please post one that was sure wrong, and then a good one so we can try to compare.

Please post the call traces via pastebin, see instructions: https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs-PartII

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