Archives : March-2015
Hello.I continue to transfer chan_sip to pjsip.Friend in chan_sip can has options:deny=0.0.0.0/0.0.0.0permit2.168.0.1pjsip 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:80081@10.47.10.10:5060 SIP/2.0**Record-Route: **Via: SIP/2.0/UDP 200.152.125.221;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..
All,I have an Asterisk server v13.1.0 running on EC2 and I am able to connect and register SIP devices and see them on the asterisk CLI. I am also able to place calls, but I am not able to hear any audio on either end after the call is picked up.I ..
Asterisk receives a 180 Ringing with SDP from the called side, then it sends 180 without SDP to the calling side.We would like asterisk to sends to the calling side the same response that was received from the called side. This is Asterisk cert 13..
I have just installed DAHDI 2.10.0.1 on a system running CentOS 5.11(lets not get sidetracked into discussing the version of CentOS – there are reasons for using it in this case).The system has a TE220 card with 2xE1.It has been working fine, but w..
Hello!Just installed asterisk 13.2.0 and see many such messages in log, I see them in console during calls, really something like this: — Executing [6166@kanbaikal:2] Dial(OOH323/kanbaikal-6, SIP/6166@asterisk) in new stack == Using SIP RTP TOS b..