Archives : September-2015
I have a setup where I have polycom phones in an office, behind firewall, going out to a server located elsewhere. I have set nat=force_rport,comedia for my phones.so if I call OUT to my cell I get audio both ways and the call is fine.My issue is..
AllWhat, by definition, goes to the cdr tables dst column ??In our setup, to get outside the user has to dial X before any number. This goes to the dst with the X stripped out.I recently made some changes in a macro and after that the X appeared in t..
Its easy to lose yourself in our massive collection of watches – from Submariners to s..
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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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Hello. Continuea months-longstrugglethat is associatedwith the transfer from chan_sip to res_pjsip. Where are many gates (GSM gate) that do not supportauthentication whensendingMESSAGE. For example, 4goip when relay incoming SMS. Using chan_sip it ..