Hello again, Release tarball didn't work for me, I had to use the trunk from subversion. Is there any essential difference? PJSYSTEST shows now much more devices than the tarball: 11:48:47.755 systest.c Running Audio Device List Audio Device List Found 22 devices 0: ALSA [default:CARD=ALSA] (0/1) 1: ALSA [sysdefault:CARD=ALSA] (0/1) 2: ALSA [dmix:CARD=ALSA,DEV=0] (0/1) 3: ALSA [dmix:CARD=ALSA,DEV=1] (0/1) 4: ALSA [hw:CARD=ALSA,DEV=0] (0/1) 5: ALSA [hw:CARD=ALSA,DEV=1] (0/1) 6: ALSA [plughw:CARD=ALSA,DEV=0] (0/1) 7: ALSA [plughw:CARD=ALSA,DEV=1] (0/1) 8: ALSA [default:CARD=Set] (1/1) 9: ALSA [sysdefault:CARD=Set] (1/1) 10: ALSA [front:CARD=Set,DEV=0] (1/1) 11: ALSA [surround21:CARD=Set,DEV=0] (1/1) 12: ALSA [surround40:CARD=Set,DEV=0] (1/1) 13: ALSA [surround41:CARD=Set,DEV=0] (1/1) 14: ALSA [surround50:CARD=Set,DEV=0] (1/1) 15: ALSA [surround51:CARD=Set,DEV=0] (1/1) 16: ALSA [surround71:CARD=Set,DEV=0] (1/1) 17: ALSA [iec958:CARD=Set,DEV=0] (1/1) 18: ALSA [dmix:CARD=Set,DEV=0] (0/1) 19: ALSA [dsnoop:CARD=Set,DEV=0] (1/0) 20: ALSA [hw:CARD=Set,DEV=0] (1/1) 21: ALSA [plughw:CARD=Set,DEV=0] (1/1) For all other interested people - here is how I get it to work: Prerequisites: 0. Install other packages (certainly incomplete): sudo aptitude install subversion build-essential automake autoconf libtool libasound2-dev libpulse-dev libssl-dev libsamplerate0-dev libcommoncpp2-dev libccrtp-dev libzrtpcpp-dev libdbus-1-dev libdbus-c++-dev libyaml-dev libpcre3-dev libgsm1-dev libspeex-dev libspeexdsp-dev libcelt-dev libasound2 libasound2-dev libasound2-plugins alsa-utils libportaudio2 portaudio19-dev 1. Check out from subversion: svn checkout http://svn.pjsip.org/repos/pjproject/trunk 2. Put options in pjlib/include/pj/config_site.h (see below) 3. in root directory: ./configure && make dep && make clean && make 4. Test PJSUA in pjsip-apps/bin 5. sudo make install 6. Use config_site.h: # undef PJ_OS_HAS_CHECK_STACK # define PJ_OS_HAS_CHECK_STACK 0 # define PJ_LOG_MAX_LEVEL 5 # define PJ_ENABLE_EXTRA_CHECK 0 # define PJ_HAS_ERROR_STRING 0 # undef PJ_IOQUEUE_MAX_HANDLES /* Putting max handles to lower than 32 will make pj_fd_set_t size smaller * than native fdset_t and will trigger assertion on sock_select.c. */ # define PJ_IOQUEUE_MAX_HANDLES 32 # define PJ_CRC32_HAS_TABLES 0 # define PJSIP_MAX_TSX_COUNT 15 # define PJSIP_MAX_DIALOG_COUNT 15 # define PJSIP_UDP_SO_SNDBUF_SIZE 4000 # define PJSIP_UDP_SO_RCVBUF_SIZE 4000 # define PJMEDIA_HAS_ALAW_ULAW_TABLE 1 #define PJMEDIA_RESAMPLE_NONE 1 #define PJMEDIA_HAS_SPEEX_AEC 0 #define PJMEDIA_HAS_VIDEO 0 #define PJMEDIA_CONF_USE_SWITCH_BOARD 1 #define PJ_DEBUG 0 Best Regards, Bernhard Am 13.01.2016 um 22:33 schrieb Bernhard Kaiser: > Hello, > > I have some trouble with the setup of PJSUA on a Raspberry Pi with Raspbian. > > I compiled everything and installed [1], but whenever I try to make a call > via the pjsua application, I get so sound - in both directions. > > I have one cheap USB audio adapter running at my Raspberry. In other > application this adapter is running, but not with pjusa. > > pjmedia-test tells me at the end, everything seems to be okay. > > pjystest give me a lot of errors, as I cannot change the clock-rate, which > is necessary with this device. pjsua gives me the same errors when I leave > the clock-rate unspecified. > > Here the list of my devices: > Audio Device List > Found 6 devices > 0: PA [bcm2835 ALSA: bcm2835 ALSA (hw:0,0)] (0/2) > 1: PA [bcm2835 ALSA: bcm2835 IEC958/HDMI (hw:0,1)] (0/2) > 2: PA [C-Media USB Headphone Set: USB Audio (hw:1,0)] (1/2) > 3: PA [sysdefault] (0/128) > 4: PA [default] (0/128) > 5: PA [dmix] (0/2) > > Please find attached the log entries of a sample call[2] executing with the > following options: > ./pjsua-armv6l-unknown-linux-gnueabihf --app-log-level=6 \ > --log-level=6 --snd-clock-rate=48000 --registrar="sip:192.168.0.15" \ > --realm="*" --username="user" --password="pass" --local-port=5061 \ > --no-vad --clock-rate=48000 > > Does anybody see some irregularities or some mistakes I've made? > > I would really appreciate your help! > > Best Regards, > Bernhard > > [1] Installed upfront: libasound2 libasound2-dev libasound2-plugins > alsa-utils libportaudio2 portaudio19-dev > > [2] I have an asterisk server on localhost (192.168.0.15) and I'm calling > for test purposes a Gigaset IP-telephone directly - without the asterisk. > > > _______________________________________________ > 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/20160115/d52ee0e4/attachment.html>
