Howdy,Im trying to get my feet wet with pjsip using the conversion script mentioned on the Wiki on this page:https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsipIm using the copy of the script thats included with Asterisk 13/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjs..
I have released an updated AppKonference that compiles with Asterisk 13. You can download the latest code from source forge: sourceforge.net/projects/appkonferenceThat said Asteris..
all,what should i do if i want to know how long asterisk server take a time for registration 1 client on server side?especially just for voip server authentication, when we have to registered username and password in sip.conf and extensions.conf fil..
Is there a way to tell in the dialplan, Asterisk AGI, or Asterisk AMI, when a call has been hung up because the SIP rtptimeout has been reached?..