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Home » Kerio User Forums » Kerio Operator » Call Routing issue with SIP Trunks (Call Routing)
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silars

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After doing a little research on SIP trunks, Caller ID, and Call Routing, I have been unable to resolve my issue.

I have two SIP interfaces (one is an SPA3102 bridging a PSTN line, another is a SIP trunk from Nextiva with 2 lines).

I have set up one of the Nextiva DIDs to be routed to a conference call. The other is routed to a ringing group. The problem is all calls on the SIP trunk are routed to whichever is configured first (conference call or ringing group).

I'm not seeing much leeway for misconfiguration. It is possible there is some protocol issue between Nextiva and Kerio. What fields in the SIP header are you using for the Call Routing?

Furthermore, the "Rewrite Numbers" tab doesn't appear to be functional anymore. Calling numbers don't change regardless of what I configure. Though I'm not sure how much sense it makes to configure rewriting of incoming calling numbers in the outgoing routing rules.





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Vladimir Toncar (Kerio)

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Hi,

Routing is based on the request line, ie. the number right after the 'INVITE' keyword. If you do not figure it out, please send your support information file (from the System Health screen) to us.

Vladimir
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silars

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Solved.

The issue with Nextiva is they always send the first number on the trunk in the Request-Line. The called DID is sent in the "To:" field. Checking the 'Use "To:" instead of INVITE request line' under Advanced settings for the SIP interface appears to have resolved the issue.

Amazing, it works just like it says it does. A good night's sleep does wonders. Must have looked at that setting about 1000 times, but the packet captures just weren't computing in my head last night. First thing this morning...voila!

Thanks for the quick response.
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