Hello, Please see this discussion http://lists.digium.com/pipermail/asterisk-dev/2015-October/075122.html I guess you're talking about the same problem.
Michael On Tuesday, March 01, 2016 06:26:27 PM Ross Beer wrote: > Hi George, > > We need to store contacts in realtime for our system. However not all > endpoints are registered only about 200, yet asterisk loops through every > endpoint which has been defined. It does this if contacts are in realtime or > not. > > Its almost like pjsip is loading them to check if they need to be qualified > etc. > > Asterisk 1.8 only put things into cache once they were accessed, is this an > option for sourcery? > > Thanks, > > Ross > > From: [email protected] > Date: Tue, 1 Mar 2016 10:42:58 -0700 > To: [email protected] > Subject: Re: [asterisk-dev] Asterisk Segfault After PJSIP Commit 5241 > > > > On Tue, Mar 1, 2016 at 10:29 AM, Ross Beer <[email protected]> wrote: > > > > Hi George, > > I have now got asterisk 13 trunk, however loading is very slow. This is due > to asterisk reading all of the realtime Sorcery peers and marking them all as > 'Unknown'. Is there a way to only cache peers that have tried to register? > > > When you say "Asterisk 13 trunk" you do mean "branch" correct? > Assuming you have contacts coming from realtime, the only was to prevent > them from being qualified is to delete them from the ps_contacts table > before starting Asterisk. You really don't gain anything by using realtime > for contacts anyway. I'd just disable it and let Asterisk use the internal > sqlite3 database to keep track of them. > > So far its taking 20 mins to load!! > > Also asterisk has the following warning: > > taskprocessor.c:803 taskprocessor_push: The > 'subm:ast_device_state_topic-000055d0' task processor queue reached 500 > scheduled tasks. > > > Whoa! This makes me think I might have messed something up in the fix for > contacts not being cached correctly. > Don't use realtime for contacts and see what happens. I'm going to re-test. > Neither were issues in the previous release. > > Thank you for your assistance, > > Ross > > From: [email protected] > Date: Tue, 1 Mar 2016 09:08:37 -0700 > To: [email protected] > Subject: Re: [asterisk-dev] Asterisk Segfault After PJSIP Commit 5241 > > > > On Tue, Mar 1, 2016 at 5:58 AM, Ross Beer <[email protected]> wrote: > > > > Further to my previous email it appears this bug won't be easily resolved by > changing the method: > > pjsip_dlg_create_uas() >> pjsip_dlg_create_uas_and_inc_lock(). > > Asterisk starts ok, allows registrations but no calls progress. > > > You have to pull Asterisk from the 13 branch. This should have been fixed > with review 2236 and I've been running with that patch and pjproject trunk. > > > From: [email protected] > To: [email protected] > Date: Tue, 1 Mar 2016 11:49:55 +0000 > Subject: Re: [asterisk-dev] Asterisk Segfault After PJSIP Commit 5241 > > > > > I've just found an open issue for this > https://issues.asterisk.org/jira/browse/ASTERISK-25751 > > > From: [email protected] > To: [email protected] > Date: Tue, 1 Mar 2016 11:06:09 +0000 > Subject: [asterisk-dev] Asterisk Segfault After PJSIP Commit 5241 > > > > > Hi, > > Since PJSIP Commit 5241 (https://trac.pjsip.org/repos/changeset/5241) > Asterisk crashes when a device registers. > > The commit resolves the following: > > • Crash when endpoint has multiple worker threads and SIP TCP transport is > disconnected during incoming call handling. > • Deprecated pjsip_dlg_create_uas(), replaced by > pjsip_dlg_create_uas_and_inc_lock(). > • Serialized transaction state notifications (of 'terminated' and > 'destroyed') in case of transport error. > > This commit should resolve a previous segfault within Asterisk, however due > to the deprecated method I believe this is causing an additional issue. > > Can this be easily resolved to resolve both segfaults? > > Kind regards, > > Ross > > > > > > >
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