The previous scripts relied on behaviour from grep that seemingly changed, and there were one or two known cases where they didn't manage to track all possible SIP packets. These versions are rewritten in pure awk (which was being used anyway), significantly more comprehensive and should cover all strange corner cases too. Reproducible: Always
Created attachment 310775 [details] find_call_ids.sh Script that locates and dump out call time, a and b numbers, as well as Call-ID values (which can be passed to find_call_sip_trace.sh). This relies on sip debugging being enabled, and logging to a log file from where the script picks it up. Very useful to find problematic calls (ie, client complains call just dropped), now we can find the call id, and from there get a SIP trace and see from which side it got cancelled. This does not yet help with determining bad rtp paths.
Created attachment 310777 [details] find_call_sip_trace.sh Extracts SIP packets for one or more Call-ID values from a SIP debug log. Can be passed multiple IDs, and will output in chronological order. Multiple IDs is useful for cases where for example you need to see the exact ordering (and compare) between two call legs (both on SIP).
Thanks Tony, For the next bump, to replace the existing scripts.
+*asterisk-10.5.0 (06 Jun 2012) +*asterisk-1.8.13.0 (06 Jun 2012) + + 06 Jun 2012; Tony Vroon <chainsaw@gentoo.org> +files/1.8.0/find_call_ids.sh, + +files/1.8.0/find_call_sip_trace.sh, +asterisk-1.8.13.0.ebuild, + +asterisk-10.5.0.ebuild: + Bugfix releases on both the 1.8 & 10 branches, squelches a warning with bind + address set to "any", prevents an overflow on 32-bit systems for + ast_tvdiff_ms calculation and various rerouting/transfer fixes. Updated + helper scripts by Jaco Kroon, closes bug #414585.
