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blamire

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We are getting the below log infomation when a call is randomly dropped. Any ideas?

[27/Mar/2012 13:31:50] TS3DPBX asterisk[1913]: WARNING[1927]: chan_sip.c:3632 in retrans_pkt: Retransmission timeout reached on transmission 43d230986f19c59f3655dbe73d4fa909<_at_>sip.voipfone.net for seqno 270 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissio ns Packet timed out after 40032ms with no response
[27/Mar/2012 13:32:28] TS3DPBX asterisk[1913]: WARNING[1927]: chan_sip.c:3632 in retrans_pkt: Retransmission timeout reached on transmission fd0f4fa6-8d5deca9-fcd2a7c<_at_>172.16.0.120 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissio ns Packet timed out after 32000ms with no response
[27/Mar/2012 13:32:28] TS3DPBX asterisk[1913]: WARNING[1927]: chan_sip.c:3661 in retrans_pkt: Hanging up call fd0f4fa6-8d5deca9-fcd2a7c<_at_>172.16.0.120 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissio ns).
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Jakub Viták (Kerio)

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Hello, Adam.

That looks like, some - yours/other - firewall is cutting incoming connection.

Are you able to reproduce it? Or is it a problem only with one of your sip providers?

If yes, please, send me to jvitak at kerio dot com your support info (link down in system health screen) and packet dump for that call (in system > network >> packet sniffer, without 4021 port).

Thank you in advance.

Regards, Jakub Viták
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blamire

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Hi

We are not able to reproduce the problem straight away, the calls are dropped randomly. is it possible to leave a constant packet sniff going on and then pick out from the times that the dropped call happens?

Thanks
Adam
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Vladimir Toncar (Kerio)

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Hi Adam,

The packet sniffer is set to use at most half of the space for temporary files (which is an in-memory disk), so it's not possible to capture all traffic for prolonged periods of time. But you could capture only the packets going to/from your SIP provider's IP address and port 5060 only.

Vladimir
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Mars

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I have the same error message and calls are randomly dropped.

I contacted VOIP provider, who said to me to set reregister time of SIP server to 5 minutes.

Can you please tell me how this can be done on Kerio Operator Box 3210?
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Vladimir Toncar (Kerio)

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Hi,

Operator has the registration set to 2 minutes by default but will allow up to 1 hour if the SIP provider asks for it in the signaling. If the provider needs 5 minutes for some reason, it's easy - they only need to send this value.

There could be some other reason for the dropped calls. Do you loose the call while talking? Or do incoming calls fail randomly? What exactly is the behavior?

Vladimir

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Mars

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Hi Vladimir,

Reregister time is not important for SIP provider, 2 minutes or 5 minutes is both OK.

The calls drop sometimes while talking. SIP numbers (call routing) show RED and both incoming and outgoing calls are not possible.

Where should I look to solve this issue? In kerio operator box 3210 (up to date), in internet router, at SIP provider or internet provider?
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Vladimir Toncar (Kerio)

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Could you capture a failed call with Operator's built-in packet sniffer? There could be many reasons and it's best to see the whole communication in such a situation.

Vladimir
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