Archives : September-2014
Ill be in Norfolk, VA for xTupleCon in OctoberOn 15 October, there will be two events for WebRTC:14:15 a talk about the xTuple WebRTC extension at xTupleCon- must register for xTupleCon to attend this17:30 a technical / developer workshop at xTup..
Thank you for your reply.After setting pjsip set logger on, the following message is displayed.It seems that the 9002(SIP client) refuse INVITE message. Are SIP methods too many?Thanks, MMEEGGAA——–..
list,I have again come across a router which behaves very badly with my IAX2packets. This time Ive documented it and thought Id share to see if anyone else has seen similar issues.I have two asterisk servers running behind a dlink DI-604 Internet rout..
Im getting daily segfaults when running 40-100 cuncurrent calls in G729 passthrough mode. Any thoughs on why this is happening is most appreciated.#00x0000003cd773356f in __strlen_sse42 () from /lib64/libc.so.6#10x000000000043b352 in update_bridgep..
all, I continue to see the following msg on my Asterisk log:[Sep8 15:34:37] NOTICE[7375]: chan_sip.c:23277 handle_request_invite:Failed to authenticate devic..
Used it with jitsi and linphone softphones, works just OK.Just for testing i did a video-call on the loop-back, great test tool for showing the influence of (limited-) bandwith / latency.Ideal for..
Cant we use pattern matching inside a macro?Because when I am trying to do so call is terminating even for a very simple dummy dialplan.[demo3]exten=>98,1,NoOp()exten=>98,2,Macro(testme)exten=>h,1,NoOp(terminating call);[macro-testme]exten=>s,1,Playback(Digits/2)exten=>s,2,WaitExten(15)exten=>s,3,NoOp()exten=>_X,1,NoOp(${EXTEN})exten=>_X,2,Goto(s,3)E..
We exchange information among call using sipaddheader.Is there a similiar command in IAX?Tha..
I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exac..
We have a plain vanilla installation of AsteriskNOW using Digium D40/50 phones. All transfers are failing from any source to any extension with the message that is not a valid extension. Does anyone have any ideas about where to begin looking for ..