Hi, I have problems when running pjsip on arm-linux platform. It seems app sndtest can not open audio capture and playback device. Here is the log: [root at FriendlyARM /mnt]# ./pjsua-arm-unknown-linux-gnu --playback-dev=0 14:03:05.156 os_core_unix.c pjlib 1.0 for POSIX initialized 14:03:05.531 sip_endpoint.c Creating endpoint instance... 14:03:05.535 pjlib select() I/O Queue created (0x152b18) 14:03:05.535 sip_endpoint.s3c2410-uda1341-superlp: audio_set_dsp_speed:44100 prescaler:66 s3c2410-uda1341-superlp: audio_set_dsp_speed:44100 prescaler:66 s3c2410-uda1341-superlp: audio_set_dsp_speed:44100 prescaler:66 c Module "mod-msg-print" registered 14:03:05.536 sip_transport. Transport manager created. 14:03:05.537 sip_endpoint.c Module "mod-pjsua-log" registered 14:03:05.538 sip_endpoint.c Module "mod-tsx-layer" registered 14:03:05.538 sip_endpoint.c Module "mod-stateful-util" registered 14:03:05.538 sip_endpoint.c Module "mod-ua" registered 14:03:05.539 sip_endpoint.c Module "mod-100rel" registered 14:03:05.539 sip_endpoint.c Module "mod-pjsua" registered 14:03:05.540 sip_endpoint.c Module "mod-invite" registered 14:03:05.560 pasound.c PortAudio sound library initialized, status=0 14:03:05.561 pasound.c PortAudio host api count=1 14:03:05.561 pasound.c Sound device count=1 14:03:05.562 pjlib select() I/O Queue created (0x159a1c) 14:03:05.578 sip_endpoint.c Module "mod-evsub" registered 14:03:05.579 sip_endpoint.c Module "mod-presence" registered 14:03:05.580 sip_endpoint.c Module "mod-refer" registered 14:03:05.580 sip_endpoint.c Module "mod-pjsua-pres" registered 14:03:05.581 sip_endpoint.c Module "mod-pjsua-im" registered 14:03:05.582 sip_endpoint.c Module "mod-pjsua-options" registered 14:03:05.583 pjsua_core.c 1 SIP worker threads created 14:03:05.583 pjsua_core.c pjsua version 1.0 for arm-unknown-linux-gnu initialized 14:03:05.673 pjsua_core.c SIP UDP socket reachable at 192.168.1.205:5060 14:03:05.674 udp0x16b688 SIP UDP transport started, published address is 192.168.1.205:5060 14:03:05.675 pjsua_acc.c Account <sip:192.168.1.205:5060> added with id 0 14:03:05.705 tcplis:5060 SIP TCP listener ready for incoming connections at 192.168.1.205:5060 14:03:05.706 pjsua_acc.c Account <sip:192.168.1.205:5060;transport=TCP> added with id 1 14:03:05.735 pjsua_media.c RTP socket reachable at 192.168.1.205:4000 14:03:05.736 pjsua_media.c RTCP socket reachable at 192.168.1.205:4001 14:03:05.766 pjsua_media.c RTP socket reachable at 192.168.1.205:4002 14:03:05.766 pjsua_media.c RTCP socket reachable at 192.168.1.205:4003 14:03:05.796 pjsua_media.c RTP socket reachable at 192.168.1.205:4004 14:03:05.796 pjsua_media.c RTCP socket reachable at 192.168.1.205:4005 14:03:05.826 pjsua_media.c RTP socket reachable at 192.168.1.2s3c2410-uda1341-superlp: audio_set_dsp_speed:44100 prescaler:66 s3c2410-uda1341-superlp: audio_set_dsp_speed:44100 prescaler:66 s3c2410-uda1341-superlp: audio_set_dsp_speed:44100 prescaler:66 s3c2410-uda1341-superlp: audio_set_dsp_speed:16000 prescaler:264 s3c2410-uda1341-superlp: audio_set_dsp_speed:16000 prescaler:264 05:4006 14:03:05.827 pjsua_media.c RTCP socket reachable at 192.168.1.205:4007 14:03:05.828 pjsua_media.c pjsua_set_snd_dev(): attempting to open devices @16000 Hz then the teminal just stopped at this line. I've tested the /dev/dsp by "cat /dev/dsp > a.wav" and "cat a.wav > /dev/dsp" it works fine. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20081031/43aa40fc/attachment.html>
