Hi Guys/Girls, I'm a bit unsure if I should post this question, since I know this would've been asked many times before. But I've done a fair bit of research and haven't been able to solve the issue. I downloaded the latest trunk (2.2.1-svn) and built pjsip for Mac Osx 10.9.4 (build-system.docx). I then ran pjsystest and everything went fine with the audio tests. Then I run pjsua like this "./pjsua --config-file rahul.cfg sip:0046706914265 at sip.actionvoip.com" and my config file looks like this: # config file for actionvoip --id sip:rahulvenkat at sip.actionvoip.com --registrar sip:sip.actionvoip.com --proxy sip:sip.actionvoip.com;lr --realm ?*? --username rahulvenkat --password ********** --no-tcp The log from my terminal is attached in log.txt and I see this: *<sip:192.168.2.131:5060 <http://192.168.2.131:5060>>: does not register * *SIP/2.0 401 Unauthorized* I have tried all types of realm messages but it didn't help. I would really appreciate it if you can help me out here. Cheers, Rahul -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20140719/d68318ce/attachment.html> -------------- next part -------------- 14:26:11.113 os_core_unix.c !pjlib 2.2.1-svn for POSIX initialized 14:26:11.117 sip_endpoint.c .Creating endpoint instance... 14:26:11.117 pjlib .select() I/O Queue created (0x7fc17c814cd8) 14:26:11.117 sip_endpoint.c .Module "mod-msg-print" registered 14:26:11.117 sip_transport. .Transport manager created. 14:26:11.117 pjsua_core.c .PJSUA state changed: NULL --> CREATED 14:26:11.117 sip_endpoint.c .Module "mod-pjsua-log" registered 14:26:11.117 sip_endpoint.c .Module "mod-tsx-layer" registered 14:26:11.117 sip_endpoint.c .Module "mod-stateful-util" registered 14:26:11.117 sip_endpoint.c .Module "mod-ua" registered 14:26:11.117 sip_endpoint.c .Module "mod-100rel" registered 14:26:11.117 sip_endpoint.c .Module "mod-pjsua" registered 14:26:11.118 sip_endpoint.c .Module "mod-invite" registered 14:26:11.143 pa_dev.c ..PortAudio sound library initialized, status=0 14:26:11.143 pa_dev.c ..PortAudio host api count=1 14:26:11.143 pa_dev.c ..Sound device count=4 14:26:11.143 pjlib ..select() I/O Queue created (0x7fc17c828228) 14:26:11.148 sip_endpoint.c .Module "mod-evsub" registered 14:26:11.148 sip_endpoint.c .Module "mod-presence" registered 14:26:11.148 sip_endpoint.c .Module "mod-mwi" registered 14:26:11.148 sip_endpoint.c .Module "mod-refer" registered 14:26:11.148 sip_endpoint.c .Module "mod-pjsua-pres" registered 14:26:11.148 sip_endpoint.c .Module "mod-pjsua-im" registered 14:26:11.148 sip_endpoint.c .Module "mod-pjsua-options" registered 14:26:11.148 pjsua_core.c .1 SIP worker threads created 14:26:11.148 pjsua_core.c .pjsua version 2.2.1-svn for Darwin-13.3/x86_64 initialized 14:26:11.148 pjsua_core.c .PJSUA state changed: CREATED --> INIT 14:26:11.148 sip_endpoint.c Module "mod-default-handler" registered 14:26:11.149 pjsua_core.c SIP UDP socket reachable at 192.168.2.131:5060 14:26:11.149 udp0x7fc17ad02 SIP UDP transport started, published address is 192.168.2.131:5060 14:26:11.149 pjsua_acc.c Adding account: id=<sip:192.168.2.131:5060> 14:26:11.149 pjsua_acc.c .Account <sip:192.168.2.131:5060> added with id 0 14:26:11.149 pjsua_acc.c Modifying accunt 0 14:26:11.149 pjsua_acc.c Acc 0: setting online status to 1.. 14:26:11.149 pjsua_pres.c Adding buddy: sip:0046706914265 at sip.actionvoip.com 14:26:11.149 pjsua_pres.c .Buddy 0 added. 14:26:11.149 pjsua_pres.c .Buddy 0: unsubscribing presence.. 14:26:11.149 pjsua_pres.c ..Buddy 0: updating presence.. 14:26:11.149 pjsua_core.c PJSUA state changed: INIT --> STARTING 14:26:11.149 sip_endpoint.c .Module "mod-unsolicited-mwi" registered 14:26:11.149 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING 14:26:11.149 main.c Ready: Success 14:26:11.149 pjsua_call.c Making call with acc #0 to sip:0046706914265 at sip.actionvoip.com 14:26:11.149 pjsua_aud.c .Set sound device: capture=-1, playback=-2 14:26:11.149 pjsua_app.c ..Turning sound device ON 14:26:11.149 pjsua_aud.c ..Opening sound device PCM at 16000/1/20ms 14:26:11.168 ec0x7fc17ac2cc ...AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=0 ms 14:26:11.169 pjsua_media.c .Call 0: initializing media.. 14:26:11.169 pjsua_media.c ..RTP socket reachable at 192.168.2.131:4000 14:26:11.169 pjsua_media.c ..RTCP socket reachable at 192.168.2.131:4001 14:26:11.169 pjsua_media.c ..Media index 0 selected for audio call 0 14:26:11.207 pjsua_core.c ....TX 1131 bytes Request msg INVITE/cseq=2739 (tdta0x7fc17b08bc00) to UDP 77.72.174.128:5060: INVITE sip:0046706914265 at sip.actionvoip.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.131:5060;rport;branch=z9hG4bKPjaUXxku56AoYRL79qa-jtPzqzjtBjJ1Sf Max-Forwards: 70 From: <sip:192.168.2.131>;tag=yqZrPjjejS1fj2Ccyg7jG5utiFa.ApP. To: sip:0046706914265 at sip.actionvoip.com Contact: <sip:192.168.2.131:5060;ob> Call-ID: v7EfAe-RLOe6WVquTp9VBhcPAeld7w5k CSeq: 2739 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: PJSUA v2.2.1-svn Darwin-13.3/x86_64 Content-Type: application/sdp Content-Length: 476 v=0 o=- 3614761571 3614761571 IN IP4 192.168.2.131 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96 c=IN IP4 192.168.2.131 b=TIAS:64000 a=rtcp:4001 IN IP4 192.168.2.131 a=sendrecv a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 --end msg-- 14:26:11.207 pjsua_app.c .......Call 0 state changed to CALLING >>>> Account list: *[ 0] <sip:192.168.2.131:5060>: does not register Online status: Online Buddy list: [ 1] <?> sip:0046706914265 at sip.actionvoip.com +=============================================================================+ | Call Commands: | Buddy, IM & Presence: | Account: | | | | | | m Make new call | +b Add new buddy .| +a Add new accnt | | M Make multiple calls | -b Delete buddy | -a Delete accnt. | | a Answer call | i Send IM | !a Modify accnt. | | h Hangup call (ha=all) | s Subscribe presence | rr (Re-)register | | H Hold call | u Unsubscribe presence | ru Unregister | | v re-inVite (release hold) | t ToGgle Online status | > Cycle next ac.| | U send UPDATE | T Set online status | < Cycle prev ac.| | ],[ Select next/prev call +--------------------------+-------------------+ | x Xfer call | Media Commands: | Status & Config: | | X Xfer with Replaces | | | | # Send RFC 2833 DTMF | cl List ports | d Dump status | | * Send DTMF with INFO | cc Connect port | dd Dump detailed | | dq Dump curr. call quality | cd Disconnect port | dc Dump config | | | V Adjust audio Volume | f Save config | | S Send arbitrary REQUEST | Cp Codec priorities | | +-----------------------------------------------------------------------------+ | q QUIT L ReLoad sleep MS echo [0|1|txt] n: detect NAT type | +=============================================================================+ You have 1 active call Current call id=0 to sip:0046706914265 at sip.actionvoip.com [CALLING] >>> 14:26:11.236 pjsua_core.c .RX 552 bytes Response msg 401/INVITE/cseq=2739 (rdata0x7fc17b805028) from UDP 77.72.174.128:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.131:5060;rport;branch=z9hG4bKPjaUXxku56AoYRL79qa-jtPzqzjtBjJ1Sf From: <sip:192.168.2.131:5060>;tag=yqZrPjjejS1fj2Ccyg7jG5utiFa.ApP. To: <sip:0046706914265 at sip.actionvoip.com> Contact: sip:0046706914265 at 77.72.174.128:5060 Call-ID: v7EfAe-RLOe6WVquTp9VBhcPAeld7w5k CSeq: 2739 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE WWW-Authenticate: Digest realm="sipdiscount.com",nonce="1073462390",algorithm=MD5 Content-Length: 0 --end msg-- 14:26:11.236 pjsua_core.c ..TX 347 bytes Request msg ACK/cseq=2739 (tdta0x7fc17b81c000) to UDP 77.72.174.128:5060: ACK sip:0046706914265 at sip.actionvoip.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.131:5060;rport;branch=z9hG4bKPjaUXxku56AoYRL79qa-jtPzqzjtBjJ1Sf Max-Forwards: 70 From: <sip:192.168.2.131>;tag=yqZrPjjejS1fj2Ccyg7jG5utiFa.ApP. To: sip:0046706914265 at sip.actionvoip.com Call-ID: v7EfAe-RLOe6WVquTp9VBhcPAeld7w5k CSeq: 2739 ACK Content-Length: 0 --end msg-- 14:26:11.236 sip_auth_clien ....Unable to set auth for tdta0x7fc17b08bc00: can not find credential for sipdiscount.com/Digest 14:26:11.236 pjsua_app.c .....Call 0 is DISCONNECTED [reason=401 (Unauthorized)] 14:26:11.236 pjsua_app_comm ..... [DISCONNCTD] To: sip:0046706914265 at sip.actionvoip.com Call time: 00h:00m:00s, 1st res in 68 ms, conn in 0ms 14:26:11.236 pjsua_media.c .....Call 0: deinitializing media.. 14:26:12.237 pjsua_aud.c !Closing sound device after idle for 1 second(s) 14:26:12.237 pjsua_app.c .Turning sound device OFF 14:26:12.237 pjsua_aud.c .Closing AirPlay sound playback device and Built-in Microph sound capture device -------------- next part -------------- A non-text attachment was scrubbed... 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