search for: ast_format_slinear

...erisk). >>>>> >>>>>This patch should be applied to indications.c under the main asterisk >>>>>source directory (usually /usr/src/asterisk): >>>>> >>>>>68a69 >>>>> > if (!(chan->nativeformats & AST_FORMAT_SLINEAR)) return 0; >>>>> >>>>>Oh, and finally, here's a shameless plug to a good friend's website >>>>>(which includes a VOIP forum!): http://outcast.ws >>>>> >>>>>Comments? Questions? :) >>>> >>>&gt...

I have read about that on asterisk 1.6, there will be a parameter "o" (Only listen to audio coming from this channel), I have tried, but I still get inbound and outbound audio from the spied channel. Has anyone used this feature? Is it working? Is there any work-around? I will like to only spy the outbound audio from a channel, I dont want to hear the incomming audio of that channel. I

...dannounce.c /* Now place the call to the extention */ snprintf(buf, sizeof(buf), "%d", lot); memset(&oh, 0, sizeof(oh)); oh.parent_channel = chan; oh.vars = ast_variable_new("_PARKEDAT", buf); dchan = __ast_request_and_dial(dialtech, AST_FORMAT_SLINEAR, dialstr,30000, &outstate, chan->cid.cid_num, chan->cid.cid_name, &oh); I assume (I hope not incorrectly) that I have to modify the variable chan->cid.cid_name Could one of the Asterisk gurus point me in the right direction as to how to do this? Thanks in advance Brian --------...

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to

> > Now I have two phones connect to my hardware PBX,and want to Make two calls from within Asterisk and switch them together. I now have the two numbers and the other parameter should how to set. for example: the value of data, type and format ,I set the type "Local" and type AST_FORMAT_SLINEAR but I don't know it is write. and the data is don't know how to set. struct ast_channel *ast_request_and_dial(const char *type, int format, void *data, int timeout, int *reason, const char *cidnum, const char *cidname); -------------- next part -------------- An HTML attachment was scrubbed...

Any ideas on this warning? If I call this number, sometimes I get this error and sometimes the call goes thru fine. Why would it work sometimes? -- Executing Goto("SIP/3044-8d49", "cytel-outgoing|915124512424|1") in new stack -- Goto (cytel-outgoing,915124512424,1) -- Executing SetCIDNum("SIP/3044-8d49", "2814494000") in new stack --

Hello, Users. I loaded module chan_h323.so, chan_vpb.so. I have met a message : "No one is available to answer at this time". I don?t know what I do.. My 'h.323 trace 5' result is : == vpb/1-8: Starting record mode (codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR] -- Executing Dial("vpb/1-8", "h323/192.168.1.107") in new stack 1:21:34.936 ThreadID=0x06f2bbb0 h323ep.cxx(1323) H323 Making call to: 192.168.1.107 1:21:34.937 ThreadID=0x06f2bbb0 rfc2833.cxx(81) RFC2833 Handler created 1:21:34.971...