Ubuntu 13.10 / 64-bit - installing pjproject, pjsip, pjsua i followed this, which seems to be ok, but where is the pjsua binary package? Step 1: sudo add-apt-repository ppa:dennis.guse/sip-tools sudo apt-get update sudo apt-get install libpjsip-samples python-pjsip Step 2: Sample tools are there : /usr/lib/libpjsip-samples$ ./auddemo Found 20 devices: 0: PA [HDA Intel PCH: ALC892 Analog (hw:0,0)] (2/8) 1: PA [HDA Intel PCH: ALC892 Digital (hw:0,1)] (0/2) 2: PA [HDA Intel PCH: ALC892 Alt Analog (hw:0,2)] (2/0) 3: PA [HDA Intel PCH: HDMI 0 (hw:0,3)] (0/2) 4: PA [HDA Intel PCH: HDMI 1 (hw:0,7)] (0/8) 5: PA [USB Device 0x46d:0x825: USB Audio (hw:1,0)] (1/0) 6: PA [USB Device 0x1111:0x2222: USB Audio (hw:2,0)] (1/2) 7: PA [sysdefault] (128/128) 8: PA [front] (0/8) 9: PA [surround40] (0/8) 10: PA [surround41] (0/128) 11: PA [surround50] (0/128) 12: PA [surround51] (0/8) 13: PA [surround71] (0/8) 14: PA [iec958] (0/2) 15: PA [spdif] (0/2) 16: PA [hdmi] (0/2) 17: PA [pulse] (32/32) 18: PA [dmix] (0/2) 19: PA [default] (32/32) Step 3: run $ python call.py 09:17:31.684 os_core_unix.c !pjlib 2.1 for POSIX initialized 09:17:31.684 sip_endpoint.c .Creating endpoint instance... 09:17:31.684 pjlib .select() I/O Queue created (0x298f530) 09:17:31.684 sip_endpoint.c .Module "mod-msg-print" registered 09:17:31.684 sip_transport. .Transport manager created. 09:17:31.684 pjsua_core.c .PJSUA state changed: NULL --> CREATED bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) 09:17:31.700 pjsua_core.c .pjsua version 2.1 for Linux-3.11.0.12/x86_64/glibc-2.17 initialized Listening on 192.168.1.19 port 56313 (-1, -2) My SIP URI is sip:192.168.1.19:56313 Menu: m=make call, h=hangup call, a=answer call, q=quit m Enter destination URI to call: sip:9198 at 192.168.1.12 Making call to sip:9198 at 192.168.1.12 Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1294 Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture, inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1870 Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1994 Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1294 Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture, inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1870 Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1994 Call with sip:9198 at 192.168.1.12 is CALLING last code = 0 () My SIP URI is sip:192.168.1.19:56313 Menu: m=make call, h=hangup call, a=answer call, q=quit Call with sip:9198 at 192.168.1.12 is DISCONNCTD last code = 407 (Proxy Authentication Required) Current call is None My SIP URI is sip:192.168.1.19:56313 Menu: m=make call, h=hangup call, a=answer call, q=quit q Why the sound card capture, playback is still failing? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20131204/e9e738c2/attachment-0001.html>
