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amin1356

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Hi dear Friends,
How is it possible to understand if an external SIP interface (such as external FXO Gateway) is successfully paired with Kerio Operator? is there any status in Kerio Operator for this issue?
Best Regards,
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amin1356

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Vladimir Toncar (Kerio)

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The call routing screen displays an exclamation mark if there is a problem with the registration to a SIP provider or external SIP gateway. If there is no icon next to the interface, the registration should be OK.
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Filip Jenicek (Kerio)

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Hi,

if Operator registers to the provider then there will be a yellow exclamation mark next to the interface. By default, the registration expires every 2 minutes and is renewed periodically.

If the interface is configured not to register, then there is no such information. Basically, Operator uses the remote IP address & port and trusts all packets from that source.

Filip
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amin1356

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Dear Vlademir and Filip,
Thanks for your updates,
But as far as I Know, Kerio Operator only acts as the SIP registration server for internal phones it means SIP interface in call routing does not accept SIP registrations for external FXO gateway devices as SIP Trunk. Therefore there will be no exclamation mark next to the interface to understand the link status.
If the interface is configured not to register, then is there anyway to understand if there is a problem with the link?
Thanks,

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Vladimir Toncar (Kerio)

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Are you asking about a suitable debugging method? We recommend to use the built-in packet sniffer and then analyze the dump with Ethereal/Wireshark.
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amin1356

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Yes Vladimir,I'm asking about a suitable debugging method.
Would you please explain more about how to use the built-in packet sniffer and then analyze the dump with Ethereal/Wireshark?
Thanks,
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Vladimir Toncar (Kerio)

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The packet sniffer is in Operator under Network > Packet Sniffer. You start the sniffer, place a test call, stop the sniffer, and download the dump to your PC.

Now, open the dump in Wireshark/Ethereal and use for example the "sip" filter (just type sip into the filter line and click Apply), or use the tool under Telephony > VoIP calls.

It's good to have some knowledge about SIP and RTP, though.

If you cannot figure it out, you can send the dump my way (vtoncar at kerio dot com).
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