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sincho

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Hi there,

I am trying out to add a SIP provider to the Kerio Operator.

I'll use the provider only for outgoing calls and I do not have a permenant number from the provider.

I am adding a new SIP interface from the Call Routing menu. Entering the server, username&password info to the fields.

Since there is no external number provider from the provider I am leaving it empty.

The interface successfully registers(!), gives no error.

When I try to call a number, after35 secs. of waiting the line drops and it doesn't call.

I've tried my credidentials on Zoiper and it works!

1) How can I exactly know if Operator successfull registers trunk to the SIP?
2) Where should I look to solve this issue?

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Filip Jenicek (Kerio)

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The "register" checkbox is unchecked, thus the interface doesn't register. It probably doesn't matter because registrations are usually needed for inbound calls.

The best way how to debug this is to start a packet dump, make an outbound call and then analyze the dump using wireshark. Look for a SIP INVITE packet sent to the provider.

Some more useful information should also be in the warning log and error log.

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sincho

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Thanks for the swift response.

Yeah I've unchecked the "register" checkbox on purpose. If I check it, it gives a warning that the "interface is not registered or configured. Refresh to see if its temporary..."

When the call is dropped the following lines are from the warning log.

[15/Oct/2015 16:31:05] asterisk[1125]: WARNING[1150]: chan_sip.c:3651 in retrans_pkt: Retransmission timeout reached on transmission 0a9ec7fc3d3f6a5f5f5201515481bc36<_at_>xxx.xxx.xxx.xxx:5060 for seqno 102 (Critical Request) -- See https:/ /wiki.asterisk.... AST/SIP+Retransmissions#012Packet timed out after 32001ms with no response
[15/Oct/2015 16:31:05] asterisk[1125]: WARNING[1150]: chan_sip.c:3680 in retrans_pkt: Hanging up call 0a9ec7fc3d3f6a5f5f5201515481bc36<_at_>xxx.xxx.xxx.xxx:5060 - no reply to our critical packet (see https: /wiki.asterisk /AST/SIP+Retransmissions).

I'vent tried the wireshark yet.

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sincho

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I see such error in the error log when I click the "Registrar".

[16/Oct/2015 11:41:42] Cannot generate sip interface with empty number if registration is enabled


As I said before I do not have a number from the provider!

I've talked with my provider and they don't see a registeration request from our side to their server.

How can I establish to register the trunk without a number?

[Updated on: Fri, 16 October 2015 10:54]

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sincho

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Issue solved. It turned out that there was a catch in the my firewall settings for the connection error. Thanks again:)
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