Hi Bill, Thanks for the reply. No luck with that either. I get the same error (failed2.txt) - SIP/2.0 401 Unauthorized and then ... 22:50:29.124 sip_auth_clien ....Unable to set auth for tdta0x7fd839040000: can not find credential for sipdiscount.com/Digest 22:50:29.124 pjsua_app.c .....Call 1 is DISCONNECTED [reason=401 (Unauthorized)] 22:50:29.124 pjsua_app_comm ..... [DISCONNCTD] To: sip:0046706914265 at sip.actionvoip.com However, I managed a workaround i.e. used another SIP provider - linphone (instead of actionvoip). Created 2 accounts of linphone (since it allows calls only between registered users). And made a call using pjsua (SIP user1) to my iPhone where I installed the linphone app (SIP user 2). It worked great!! The workaround should hold good for now, but if I'm able to call phone numbers it would be fantastic :) /Rahul On Sat, Jul 19, 2014 at 10:42 PM, Bill Gardner <billg at wavearts.com> wrote: > Try letting the client register first, then make the call, i.e, don't > pass the sip address in the command line. - Bill > > > On 7/19/2014 8:55 AM, Rahul Venkatram wrote: > > 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 > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing listpjsip at lists.pjsip.orghttp://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/20140719/222fecc7/attachment.html> -------------- next part -------------- white-tooth:bin Kingdom$ ./pjsua --config-file rahul.cfg 22:44:47.211 os_core_unix.c !pjlib 2.2.1-svn for POSIX initialized 22:44:47.243 sip_endpoint.c .Creating endpoint instance... 22:44:47.247 pjlib .select() I/O Queue created (0x7fd8388152d8) 22:44:47.248 sip_endpoint.c .Module "mod-msg-print" registered 22:44:47.248 sip_transport. .Transport manager created. 22:44:47.248 pjsua_core.c .PJSUA state changed: NULL --> CREATED 22:44:47.264 sip_endpoint.c .Module "mod-pjsua-log" registered 22:44:47.266 sip_endpoint.c .Module "mod-tsx-layer" registered 22:44:47.266 sip_endpoint.c .Module "mod-stateful-util" registered 22:44:47.266 sip_endpoint.c .Module "mod-ua" registered 22:44:47.266 sip_endpoint.c .Module "mod-100rel" registered 22:44:47.266 sip_endpoint.c .Module "mod-pjsua" registered 22:44:47.267 sip_endpoint.c .Module "mod-invite" registered 22:44:47.312 pa_dev.c ..PortAudio sound library initialized, status=0 22:44:47.312 pa_dev.c ..PortAudio host api count=1 22:44:47.312 pa_dev.c ..Sound device count=4 22:44:47.312 pjlib ..select() I/O Queue created (0x7fd838829828) 22:44:47.331 sip_endpoint.c .Module "mod-evsub" registered 22:44:47.331 sip_endpoint.c .Module "mod-presence" registered 22:44:47.331 sip_endpoint.c .Module "mod-mwi" registered 22:44:47.331 sip_endpoint.c .Module "mod-refer" registered 22:44:47.331 sip_endpoint.c .Module "mod-pjsua-pres" registered 22:44:47.331 sip_endpoint.c .Module "mod-pjsua-im" registered 22:44:47.331 sip_endpoint.c .Module "mod-pjsua-options" registered 22:44:47.331 pjsua_core.c .1 SIP worker threads created 22:44:47.331 pjsua_core.c .pjsua version 2.2.1-svn for Darwin-13.3/x86_64 initialized 22:44:47.331 pjsua_core.c .PJSUA state changed: CREATED --> INIT 22:44:47.332 sip_endpoint.c Module "mod-default-handler" registered 22:44:47.332 pjsua_core.c SIP UDP socket reachable at 192.168.2.131:5060 22:44:47.332 udp0x7fd838514 SIP UDP transport started, published address is 192.168.2.131:5060 22:44:47.332 pjsua_acc.c Adding account: id=<sip:192.168.2.131:5060> 22:44:47.332 pjsua_acc.c .Account <sip:192.168.2.131:5060> added with id 0 22:44:47.332 pjsua_acc.c Modifying accunt 0 22:44:47.332 pjsua_acc.c Acc 0: setting online status to 1.. 22:44:47.333 pjsua_core.c PJSUA state changed: INIT --> STARTING 22:44:47.333 sip_endpoint.c .Module "mod-unsolicited-mwi" registered 22:44:47.333 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING 22:44:47.333 main.c Ready: Success >>>> Account list: *[ 0] <sip:192.168.2.131:5060>: does not register Online status: Online Buddy list: -none- +=============================================================================+ | 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 0 active call >>> m (You currently have 0 calls) Buddy list: -none- Choices: 0 For current dialog. -1 All 0 buddies in buddy list [1 - 0] Select from buddy list URL An URL <Enter> Empty input (or 'q') to cancel Make call: sip:0046706914265 at sip.actionvoip.com 22:50:29.032 pjsua_call.c !Making call with acc #0 to sip:0046706914265 at sip.actionvoip.com 22:50:29.032 pjsua_aud.c .Set sound device: capture=-1, playback=-2 22:50:29.032 pjsua_app.c ..Turning sound device ON 22:50:29.032 pjsua_aud.c ..Opening sound device PCM at 16000/1/20ms 22:50:29.046 ec0x7fd8385157 ...AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=0 ms 22:50:29.047 pjsua_media.c .Call 1: initializing media.. 22:50:29.051 pjsua_media.c ..RTP socket reachable at 192.168.2.131:4000 22:50:29.051 pjsua_media.c ..RTCP socket reachable at 192.168.2.131:4001 22:50:29.051 pjsua_media.c ..Media index 0 selected for audio call 1 22:50:29.090 pjsua_core.c ....TX 1132 bytes Request msg INVITE/cseq=13692 (tdta0x7fd839040000) 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=z9hG4bKPjzrvkMoRGbRD6uS.VS0RQcLF8Z3013Nm2 Max-Forwards: 70 From: <sip:192.168.2.131>;tag=YgLqBSQbXsQhVgA5--2CH-uW.LhkJ9ew To: sip:0046706914265 at sip.actionvoip.com Contact: <sip:192.168.2.131:5060;ob> Call-ID: YSNFFpabFKpzM5vm1xalW3tHsiOZfrVR CSeq: 13692 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=- 3614791829 3614791829 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-- 22:50:29.090 pjsua_app.c .......Call 1 state changed to CALLING >>> 22:50:29.123 pjsua_core.c .RX 553 bytes Response msg 401/INVITE/cseq=13692 (rdata0x7fd839009828) from UDP 77.72.174.128:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.131:5060;rport;branch=z9hG4bKPjzrvkMoRGbRD6uS.VS0RQcLF8Z3013Nm2 From: <sip:192.168.2.131:5060>;tag=YgLqBSQbXsQhVgA5--2CH-uW.LhkJ9ew To: <sip:0046706914265 at sip.actionvoip.com> Contact: sip:0046706914265 at 77.72.174.128:5060 Call-ID: YSNFFpabFKpzM5vm1xalW3tHsiOZfrVR CSeq: 13692 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE WWW-Authenticate: Digest realm="sipdiscount.com",nonce="3965857031",algorithm=MD5 Content-Length: 0 --end msg-- 22:50:29.124 pjsua_core.c ..TX 348 bytes Request msg ACK/cseq=13692 (tdta0x7fd838807000) 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=z9hG4bKPjzrvkMoRGbRD6uS.VS0RQcLF8Z3013Nm2 Max-Forwards: 70 From: <sip:192.168.2.131>;tag=YgLqBSQbXsQhVgA5--2CH-uW.LhkJ9ew To: sip:0046706914265 at sip.actionvoip.com Call-ID: YSNFFpabFKpzM5vm1xalW3tHsiOZfrVR CSeq: 13692 ACK Content-Length: 0 --end msg-- 22:50:29.124 sip_auth_clien ....Unable to set auth for tdta0x7fd839040000: can not find credential for sipdiscount.com/Digest 22:50:29.124 pjsua_app.c .....Call 1 is DISCONNECTED [reason=401 (Unauthorized)] 22:50:29.124 pjsua_app_comm ..... [DISCONNCTD] To: sip:0046706914265 at sip.actionvoip.com Call time: 00h:00m:00s, 1st res in 77 ms, conn in 0ms 22:50:29.124 pjsua_media.c .....Call 1: deinitializing media.. 22:50:30.126 pjsua_aud.c !Closing sound device after idle for 1 second(s) 22:50:30.126 pjsua_app.c .Turning sound device OFF 22:50:30.126 pjsua_aud.c .Closing AirPlay sound playback device and Built-in Microph sound capture device
