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Hi, I'd like to implement some way of controlling remote SIP clients while in a call, to execute remote commands. The call topology (think of a PA system) is this: * the caller is in a MeetMe() conference room * the callees are Page()d, then the dynamic conference room is connected to the previous one I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the caller to the

Hi, I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesn't matter whether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone

I'm building a paging system composed of roughly 10 switches in daisy chain, with an embedded box with a speaker and a microphone for each switch. The embedded box runs my software. I need the system to be resilient to any network partition, so that anyone can send announces from any mic to all the reachable clients. I'd need also to page a subset of all the speakers. I'm

hi all, please help... I am calling in the simplest way among two linphone clients connected to one asterisk server... the call ends on one side without any sign of problem, while on the other side it stays connected. I checked the SIP dialogue and at some point the server sends a BYE message to one party I have no timeout set, though the duration of a call is always around 20s. the two

Full_Name: Peter Fortini Version: 2.0.1 OS: Windows 2000 Submission from: (NULL) (65.246.187.164) When the imaginary part of the argument is very large, the complex tangent function returns 0+NaNi. For example, tan(1+1000i)=0+NaNi; it should be 0+1i Easy to fix in complex.c, as the original NaN came from division of si...

I'm trying to improve Opus on an AMD Geode CPU, which has limited SSE support (called 3DNow!), but MMX. Without optimizations I can only encode 16 bit audio @16KHz with complexity up to 2-3 without underruns. I tried compiling with SSE2/4 optimizations, but all I got was a crash with SIGILL, so I looked into optimized code and found that a good starting point was the dot product, so I

Hi, I have the same extension registered with multiple softphones on multiple servers, i.e. 100-local at hostA 100-local at hostB and on both hostA and hostB I have the extension in extension.conf exten => 100,1,Answer() exten => 100,n,Dial(100-local) When from softphone registered as 100-local at hostA I call (100 at hostB) what I see on the softphone on host B is "100-local at

Hi, if I Page more than one extension, then the MeetMe conference stays up even if all the called extensions aren't available or are hung up. Is there a way of keeping track of how many extensions are attached to the conference, and require a number or a particular extension to be present? Thank you