Hello.I continue to transfer chan_sip to pjsip.Friend in chan_sip can has options:deny=0.0.0.0/0.0.0.0permit188.8.131.52pjsip offer to use global ACL without relation to any andpoint. My task is restriction via IP to registering in certain endpoint. Differ..
Hello.Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have match_auth_username=yes and have nothing in pjsip.I have a lot of endpoints and registrations on same SIP server. And its problem in pjsip now. Is not it?I request..
Asterisk, Back in 2009 I built a small Intel Atom based computer runningCentOS 5 for my asterisk system. 5 phones, 2 people 1 POTsline and six or so SIP numbers. So basically no load. Imfeeling like its time to build another machine. Its probablysil..
*friends help me **cant get incoming calls in asterisk**(when i connect **80081 in softphone —every thing is ok**)****INVITE sip:email@example.com:5060 SIP/2.0**Record-Route: **Via: SIP/2.0/UDP 184.108.40.206;branch=z9hG4bKd4fd.b3489837.0**Via: SIP/2.0/..
I am dealing with a FreePBX generated dialplan.I have been following the processing traces attempting to make use of the advice I received here respecting setting a custom ring tone. I have discovered that the context I am using for incoming calls..
HiI plan to host Asterisk instances on AWS/EC2 servers. Requirement is to run asterisk instance with transcoding (g.729 + g.711) and full recording. Number of concurrent calls expected are 500+. 2 instances will be configured for 100% redundancy. He..