Hello,
I haven't read full thread, but I noticed in the config snippet below that you use $shv(...) - this is single value variable that stores the value in shared memory. The $ci is not replaced in definition of $shv($ci), but is used as literal token '$ci'.
You should use htable module, defining a hash table, say 'overlap' and then use $sht(overlap=>$ci).
Cheers, Daniel
On 04/02/14 17:47, Moritz Graf wrote:
Hi,
nah, that's not what I wanted to hear ;). And it's not really complicated. I hacked it down (my third choice in the initial mail) with $shv(). See (find also attached):
#################################################### ### Hacky section xdbgl("Shared Variable of current ( Call-ID = $ci ) is $shv($(ci)) "); $var(numberLength) = $(var(calledNumber_normalized){s.len}); xdbgl("Length of string ( $var(calledNumber_normalized) )is $var(numberLength) characters."); if ( $shv($(ci)) < $var(numberLength) ) { xdbgl("I detected, that the old value is lower, setting it to the new value."); $shv($(ci)) = $var(numberLength); } # sleeping to wait for potentially new invites sl_send_reply("100", "Trying..."); # So no retransmits happen usleep("500000"); #half a second, this is the timer i was speaking about if( $var(numberLength) < $shv($(ci)) ) { xdbgl("Detected newer INVITE, aborting this transaction with 484 response"); route("address_incomplete"); break; } ######################################################
Do you see any problems with this?
So this is not really smooth, probably someone has a better solution for this??
greetz
Am 04.02.2014 16:16, schrieb Alex Balashov:
I'm not sure this is a task for a proxy like Kamailio, even with its fancy new async relay features. To truly customise how successive INVITEs are dealt with like this, I think it needs to be handled by a UAS that has complete latitude in terms of how it computes state.
On 4 February 2014 10:13:33 GMT-05:00, Moritz Graf moritz.graf@g-fit.de wrote:
Hi, sure. That's why it's called "transaction oriented". On a SIP-Theory basis, the second INVITE opens a new transaction!! For correlation the Call-ID keeps the same! This is how all of the big carriers I know handle OverlapDialing. The problem is: * I am receiving an INVITE. * At this point I don't know if there will be a second INVITE or not. So I have to wait. * If the timer elapses, ok handle this INVITE. * But if there IS a second (third..) INVITE, I have to cancel the first and then handle the second. Sounds like nobody has done this before... any suggetions appriciated. greetz Am 04.02.2014 15:56, schrieb Alex Balashov: Indeed, Section 3.1 in the referenced RFC covers this pretty well. On 4 February 2014 09:47 :38 GMT-05:00, "Olle E. Johansson" <oej@edvina.net> wrote: On 04 Feb 2014, at 15:43, Moritz Graf <moritz.graf@g-fit.de> wrote: Shortly explained what RFC3578 is: In a open numbering plan you never know if the INVITE you received is already complete, or if there are more numbers coming in. One way of accomplishing this is to set up a timer. If the timer elapses you assume the number is complete. If not, you are receiving a new INVITE with one digit more. Now you have to close the old transaction with a "484 - Address Incomplete"-response and start the timer again. (Find the algorithm I want to implement attached) You are not allowed to have two open INVITEs, the second one SHOULD get a 491 response. The client should not send a new INVITE if the ol d INVITE transaction is not complete. I don't th ink overlap dialing in SIP was ever meant to be timer based. Consider that the first INVITE can go to a different proxy than the second. There's no dialog, no route set. /O ------------------------------------------------------------------------ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Sent from my Nexus 10, with all the figments of autocorrect that might imply. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
-- Sent from my Nexus 10, with all the figments of autocorrect that might imply.
Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users