Retransmission timeout reached on transmission error

I am trying to connect two asterisk servers vi a SIP trunk.

Server 1 has PSTN line and server 2 has python client initiating a call.

what happens here is server1 is receiving 103 INVITE and sending ACK back to server2 but i cannot see this ACK on server2 by SIP debuging.

Here server2 is initiating a call and sending request on server1.

Both the servers are on same subnet reachable with just 1 hop on traceroute.

can someone help me where i am missing config?

Refer attached logs,
server1_PRI_Server_logs.txt (24.7 KB)server2_python_Server_logs.txt (31.8 KB)

Daemon 1 has is the client, but your initial text implies that daemon 2 is the client.

Daemon 1 is sending, not receiving, the cseq 103 INVITE.

Daemon 1 is not receiving the 100 Trying, until after it has given up retrying the INVITE, at which point the call is completely lost.

There are no timestamps on the protocol trace, which probably means it is a screen scrape, rather than taken from the log. As such, I can’t confirm that the repeat INVITEs are being sent with the correct timing.

@david551 can you pls brief more about it.

Do i have to change anything in any of the conf files?

The only thing I can think of that could have been done to configuration files to cause all the INVITEs to be sent before any of the responses were received would be to have changed them in the first place, so as reduce the timeouts to unrealistically low values. I’m not even sure it is possible to set them that low.

Firstly you need a much clearer understanding of your system, as your initial description conflicted with the evidence.

After that, I’d suggest getting the logging from the logs, rather than screen scraping, so that you can see the timestamps. Then you need to work out why it is taking more than the full protocol timeout (which defaults to about 30 seconds) before the 100 Trying is getting through. Unless you have set silly values for the timeouts, something is drastically overloaded. It could be the network, but a better guess would be that you are trying to do this with virtual machines and the daemon2 is being drastically insufficient levels of system resources.

first of the both of these servers are not on virtual machines, both of these servers are physical servers with enough resources to run calls.

my system understanding is very much clear coz i have only 2 asterisk servers.

Daemon 1 with PRI line (IP 172.16.15.151)
Daemon 2 with python client to initiate a call (IP 172.16.14.104)

and then i have created sip trunk in between them with username 3115.

From Daemon 2 initiating a call logs are

calllogsrv2.0_104server.txt (146.9 KB)

From Daemon 1 receiving and forwarding call to PSTN line call logs are

calllogsrv1.0_151server.txt (57.5 KB)

@david551 i have attached logs generated with timestamps please help me to understand…

Thank you…

Just an observation in this case,

suddenly one of the call is getting through in a day without re-transmission error.

Please help me understand why and when would this happen in asterisk as well…

Thank you…

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