I moved to Ubuntu 13.10 64-bit new setup and re-install pjproject but still i do not hear anything and cant send any audio. Can anyone please advise? This is the build and installation i followed: http://trac.pjsip.org/repos/wiki/Python_SIP/Build_Install 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:192.168.1.19:38336 Menu: m=make call, h=hangup call, a=answer call, q=quit 12:09:09.478 tsx0x7f239400b .......Temporary failure in sending Request msg INVITE/cseq=14548 (tdta0x1e27740), will try next server. Err=171060 (Unsupported transport (PJSIP_EUNSUPTRANSPORT)) On Mon, Dec 2, 2013 at 6:43 AM, Nishant Rodrigues <nishantjr at gmail.com>wrote: > Make sure you're linking pjsip with libasound. > On 1 Dec, 2013 10:21 pm, "Shamun Toha Md" <shamun at companysocia.com> wrote: > >> I am using PJSip with Python. Can someone please advise how can i fix >> this please? Or it is PJSip Bug in the core? Was there anyone yet made any >> Python implementation of PJSIP? because this bug was never fixed? Does >> anyone know if there is any working open source implementation of PJSIP >> with Python for Linux? >> >> >> *ISSUE: * >> >> Its failing to get the default audio devices, even i tried with following >> method: >> >> snd_dev = lib.get_snd_dev()print snd_dev ## returns (-1,-2) >> lib.set_snd_dev(0,0) >> >> >> *ERROR/Problem: * >> >> it fails to get microphone and speaker to output >> >> *Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) >> [status=420006]* >> >> *Exception: Object: {Account <sip:192.168.1.16:60791 >> <http://192.168.1.16:60791>>}, operation=make_call(), error=Unable to find >> default audio device (PJMEDIA_EAUD_NODEFDEV)* >> >> >> *CODE: * >> >> >> http://trac.pjsip.org/repos/browser/pjproject/trunk/pjsip-apps/src/python/samples/call.py >> >> *RESULT:* >> >> $ python call_test.py >> >> 14:13:21.938 os_core_unix.c !pjlib 2.1 for POSIX initialized >> >> 14:13:21.945 sip_endpoint.c .Creating endpoint instance... >> >> 14:13:21.948 pjlib .select() I/O Queue created (0x10532f0) >> >> 14:13:21.948 sip_endpoint.c .Module "mod-msg-print" registered >> >> 14:13:21.948 sip_transport. .Transport manager created. >> >> 14:13:21.948 pjsua_core.c .PJSUA state changed: NULL --> CREATED >> >> 14:13:21.966 pjsua_core.c .pjsua version 2.1 for >> Linux-3.5.0.17/x86_64/glibc-2.15 initialized >> >> >> Listening on 192.168.1.16 port 60791 >> >> >> My SIP URI is sip:192.168.1.16:60791 >> >> Menu: m=make call, h=hangup call, a=answer call, q=quit >> >> m >> >> Enter destination URI to call: sip:5000 at 192.168.1.12 >> >> Making call to sip:5000 at 192.168.1.12 >> >> 14:13:41.786 pjsua_aud.c ..Error retrieving default audio device >> parameters: Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) >> [status=420006] >> >> Exception: Object: {Account <sip:192.168.1.16:60791>}, >> operation=make_call(), error=Unable to find default audio device >> (PJMEDIA_EAUD_NODEFDEV) >> >> My SIP URI is sip:192.168.1.16:60791 >> >> Menu: m=make call, h=hangup call, a=answer call, q=quit >> >> >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20131202/72d55df7/attachment-0001.html>
