Asterisk Project Security Advisory – AST-2017-003 ProductAsterisk SummaryCrash in PJSIP multi-part body parser Nature of AdvisoryRemote CrashSusceptibilityRemote Unauthenticated Sessions Severity CriticalExploits KnownNo Reported On13 April, 2017 Repor..
On a newly built Asterisk 13.13.1 system, I cant make HEP work with chan_sip (though I could make it work with PJSIP on another instance).Looking either at  or hep.conf, I cant see anything meaning HEPrequires PJSIP.Before diging deeper, may I sim..
I am using Asterisk 12 and PJSIP. Last night I tried to upgrade to Asterisk 13, and it did compile just fine, but PJSIP would not load, and no error was shown on the screen when I did pjsip reload. Do I have to erase some objects before compil..
I saw this issue that was resolved this year: https://issues.asterisk.org/jira/browse/ASTERISK-23173Where they mention that it was not possible to get the status of a qualified contact using PJSIP. Do you know if theres any way of validating the cont..
I use realtime for my Asterisk configuration and are now making the transition to Asterisk 13 and PJSIP.I used alchemy to set up my databases and I can now configure my endpoints.While trying to configure a trunk I can see that there is a database ta..
Ive just discovered PJSIP s support of set_var setting in pjsip.conf. Is this setting also supported in pjsip_wizard.conf ?On a fresh 13.8.2, it doesnt seem but I may have missed somthi..
We currently have a production Asterisk box running 22.214.171.124 which uses MeetMe and chan_sip for conferences. I have been testing the new versions of Asterisk with PJSIP and ConfBridge but have run into an issue which is preventing us from moving forwa..
I am trying to set add a SIP Header to a call before adding it to the Queue.The dial plan sends the call to my macro to perform the work.When I use chan_sip, everything works as expected.When I use PSJIP support, its not adding the SIP header. Look..
In PJSIP configuration, I thought that from_domain parameter in a endpoint permits to have two SIP peers with the same usernames with a different domain.Ive tested at the transport level, I see no changes.Ive also tested with realm parameter in a..
, Im trying to use CHANNEL(aor) and CHANNEL(contact) on PJSIP channel, on asterisk-13.3.2, but they dont return anything. Is this a bug, or did I miss something? Here is my test dialplan: exten => *98,1,Answer same => n,NoOp(Channel=,type= ) same..