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Home » Kerio User Forums » Kerio Operator » 1.2.0 RC 2 routing calls problem
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Vladimir Toncar (Kerio)

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Hi,

Can you send me the support information files from both 1.1.3 and 1.2.0 RC2? You can download them from the Administration GUI, the System Health screen. Please send them to me at vtoncar at kerio dot com.

Vladimir
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fetuk

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sent specified files by e-mail
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Jakub Zeman (Kerio)

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Hi,

can you enable in debug.log 'SIP Protocol' and 'Asterisk' messages, try to call to 5555 again and then send the log please (in 1.2.0 RC2)?

Jakub
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Jakub Zeman (Kerio)

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Hello,

thank you for debug.log. The problem is in SIP packets from "another PBX". Your PBX sends following in SIP packet:

INVITE sip:5000;phonecontext=udp<_at_>10.86.1.199:5060;maddr=10.86.1.199;transport=udp;user=phone SIP/2.0

I think that sip:5000;phonecontext=udp<_at_>10.86.1.199:5060 ... is not correct according to SIP rfc.

Kerio Operator 1.2.0 RC2 is more strict to SIP packet. It means that 1.2.0 RC2 doesn't recognize the right calling number and the call ends in fallback.

You need to fix it in your PBX.

Jakub
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Jakub Zeman (Kerio)

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The correct SIP INVITE request line should look like:
INVITE sip:5000<_at_> 10.86.1.199:5060;phone-context=udp;maddr=10.86.1.199;transpo rt=udp;user=phone SIP/2.0
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fetuk

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Very strange, why is works on r 1.1.3? Not understand ....
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Jakub Zeman (Kerio)

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Asterisk 1.1.3 doean't do proper checking of SIP packets.
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Jakub Zeman (Kerio)

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We will check to codes in 1.1.3 and 1.2.0 and we will try to fix it for you.
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fetuk

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Asterisk 1.1.3 does not allow to make calls on my SIP operator. It is a vicious circle.
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fetuk

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Thank you, I hope everything will be fine
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