Archives : September-2015
I am having a small problem that is driving me nuts.I can make calls over my Webrtc client without any problems and audio sounds fine. The only problem I have is that when I call an internal SIP extension on my PBX I do not hear the ring while I w..
all,Im required to send a dialplan variable with every AMI event triggered for the duration of the call. For example;… exten => s,n,Set(MyVar=${ODBC_GetSomething(${EXTEN}))…so can I have this Variable MyVar attached in all AMI events for this ca..
I have some problem finding a smart way to add inbound trunks ip authentication. I dont want to set allowguests=yesSome of my providers just list some IP and I add them like:[provider](!)context=fromoutside type=friend insecure=port,invite disallow=..
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Buy an A-grade Duplicate watch without the original..
!I have an Asterisk 1.8.30.0-1 installed on an OpenWRT-Switch. It works correct since July with two VoIP-phones.Now I configured my mobile phone (Samsung Galaxy 2 with Android 4.1.2) to use my Asterisk, so that I can call and receive calls from my mob..
I have a customer who needs to use chanspy to :1. Interrupt a call and whisper to the local extension2. Allow the person on the local extension to reply to the interrupter, but not have the remote party hear it.Narrative:John is on the phone with a custom..
Group,I have a requirement to dialout some external number, once the call is answered the same has to be forwarded to an Internal Queue.Please help me.I have tried calling with two SIP end point forwarding , even that is not working,My dial plan l..
I have a line likeregister => 1yyyyyyy1:xxxxxxxxxxxx@sipconnect.sipgate.de/incomingin sip.conf, and a corresponding stanza (note especially the final setvar):[trunk-sipgate]type=peerqualify=yesinsecure=invitelanguage=dedtmfmode=rfc2833host=sipconnect.sipgate.defromdomain=sipconnect.sipgate.defromuser=1yyyyyyy1defaultuser=1yyyyyyy1secret=xxxxxxxxxxxxcontext=in-trunk-sipgatesession-timers=acceptallow=!all,alaw,ulaw,g726setvar=FOO=BA..