Retransmission Timeout Reached for Critical Packet, Configurable?

Asterisk Asterisk SIP
1

Hi all,

We’re getting this one of our border switches,

Retransmission timeout reached on transmission 6fee300f5706a67c172385f3751678db@1.1.1.1:5060 for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response

IP Taken out.

This is due to our SIP trunk provider ignoring any calls above the CPS limit with them (Very bad practice in my eyes, but it is what it is!) We have several other trunks we can route calls down, but the 6400ms is a bit of a while to wait for a call to fail over, Is the 6400ms timeout configurable anywhere?

The box is a FreePBX, but I’ve been unable to find any configuration setting in there, and i’m now looking for one in Asterisk config files instead.

Thanks in advance!

2

The default timeout should be a lot more than that. I think that is the default minimum adaptive timeout, i.e. srtarting at 100ms.

From sip.conf.sample, the relevant options are:

; -------------------------- SIP timers ----------------------------------------------------
; These timers are used primarily in INVITE transactions.
; The default for Timer T1 is 500 ms or the measured run-trip time between
; Asterisk and the device if you have qualify=yes for the device.
;
;t1min=100                      ; Minimum roundtrip time for messages to monitored hosts
                                ; Defaults to 100 ms
;timert1=500                    ; Default T1 timer
                                ; Defaults to 500 ms or the measured round-trip
                                ; time to a peer (qualify=yes).
;timerb=32000                   ; Call setup timer. If a provisional response is not received
                                ; in this amount of time, the call will autocongest
; Defaults to 64*timert1
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