same error :( --id sip:rahulvenkat at sip.actionvoip.com --registrar sip:sip.actionvoip.com --proxy sip:sip.actionvoip.com;lr --realm=sipdiscount.com --username rahulvenkat --password ************ On Sun, Jul 20, 2014 at 1:25 AM, Bill Gardner <billg at wavearts.com> wrote: > Try realm=sipdiscount.com > > > Sent from my iPhone > > On Jul 19, 2014, at 5:04 PM, Rahul Venkatram <rahul.venkatram at gmail.com> > wrote: > > 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 >> >> > <failed 2.txt> > > _______________________________________________ > 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/20140720/84c05260/attachment.html> -------------- next part -------------- white-tooth:bin Kingdom$ ./pjsua --config-file rahul.cfg 18:44:55.384 os_core_unix.c !pjlib 2.2.1-svn for POSIX initialized 18:44:55.388 sip_endpoint.c .Creating endpoint instance... 18:44:55.388 pjlib .select() I/O Queue created (0x7fa8aa8130d8) 18:44:55.388 sip_endpoint.c .Module "mod-msg-print" registered 18:44:55.388 sip_transport. .Transport manager created. 18:44:55.388 pjsua_core.c .PJSUA state changed: NULL --> CREATED 18:44:55.388 sip_endpoint.c .Module "mod-pjsua-log" registered 18:44:55.388 sip_endpoint.c .Module "mod-tsx-layer" registered 18:44:55.388 sip_endpoint.c .Module "mod-stateful-util" registered 18:44:55.388 sip_endpoint.c .Module "mod-ua" registered 18:44:55.389 sip_endpoint.c .Module "mod-100rel" registered 18:44:55.389 sip_endpoint.c .Module "mod-pjsua" registered 18:44:55.389 sip_endpoint.c .Module "mod-invite" registered 18:44:55.413 pa_dev.c ..PortAudio sound library initialized, status=0 18:44:55.413 pa_dev.c ..PortAudio host api count=1 18:44:55.413 pa_dev.c ..Sound device count=4 18:44:55.413 pjlib ..select() I/O Queue created (0x7fa8aa013a28) 18:44:55.419 sip_endpoint.c .Module "mod-evsub" registered 18:44:55.419 sip_endpoint.c .Module "mod-presence" registered 18:44:55.419 sip_endpoint.c .Module "mod-mwi" registered 18:44:55.419 sip_endpoint.c .Module "mod-refer" registered 18:44:55.419 sip_endpoint.c .Module "mod-pjsua-pres" registered 18:44:55.419 sip_endpoint.c .Module "mod-pjsua-im" registered 18:44:55.419 sip_endpoint.c .Module "mod-pjsua-options" registered 18:44:55.419 pjsua_core.c .1 SIP worker threads created 18:44:55.420 pjsua_core.c .pjsua version 2.2.1-svn for Darwin-13.3/x86_64 initialized 18:44:55.420 pjsua_core.c .PJSUA state changed: CREATED --> INIT 18:44:55.420 sip_endpoint.c Module "mod-default-handler" registered 18:44:55.420 pjsua_core.c SIP UDP socket reachable at 192.168.2.131:5060 18:44:55.420 udp0x7fa8a962c SIP UDP transport started, published address is 192.168.2.131:5060 18:44:55.420 pjsua_acc.c Adding account: id=<sip:192.168.2.131:5060> 18:44:55.420 pjsua_acc.c .Account <sip:192.168.2.131:5060> added with id 0 18:44:55.420 pjsua_acc.c Modifying accunt 0 18:44:55.420 pjsua_acc.c Acc 0: setting online status to 1.. 18:44:55.421 tcplis:5060 SIP TCP listener ready for incoming connections at 192.168.2.131:5060 18:44:55.421 pjsua_acc.c Adding account: id=<sip:192.168.2.131:5060;transport=TCP> 18:44:55.421 pjsua_acc.c .Account <sip:192.168.2.131:5060;transport=TCP> added with id 1 18:44:55.421 pjsua_acc.c Modifying accunt 1 18:44:55.421 pjsua_acc.c Acc 1: setting online status to 1.. 18:44:55.421 pjsua_acc.c Adding account: id=sip:rahulvenkat at sip.actionvoip.com 18:44:55.421 pjsua_acc.c .Account sip:rahulvenkat at sip.actionvoip.com added with id 2 18:44:55.421 pjsua_acc.c .Acc 2: setting registration.. 18:44:55.421 pjsua_core.c ...TX 607 bytes Request msg REGISTER/cseq=1982 (tdta0x7fa8ab004600) to UDP 77.72.174.128:5060: REGISTER sip:sip.actionvoip.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.131:5060;rport;branch=z9hG4bKPjVoDRo3TSu0dVPxlcKzf0nfYYJX5F6vmi Route: <sip:sip.actionvoip.com;lr> Max-Forwards: 70 From: <sip:rahulvenkat@xxxxxxxxxxxxxxxxxx>;tag=FYOhalxL4wJ6Vnh5eEKmEuLiL.z-UHx6 To: <sip:rahulvenkat at sip.actionvoip.com> Call-ID: kKWz8ZoGxLVibYYwIL.Hdqnt33gEfctJ CSeq: 1982 REGISTER User-Agent: PJSUA v2.2.1-svn Darwin-13.3/x86_64 Contact: <sip:rahulvenkat at 192.168.2.131:5060;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 --end msg-- 18:44:55.421 pjsua_acc.c ..Acc 2: Registration sent 18:44:55.421 pjsua_acc.c Acc 2: setting online status to 1.. 18:44:55.421 pjsua_core.c PJSUA state changed: INIT --> STARTING 18:44:55.421 sip_endpoint.c .Module "mod-unsolicited-mwi" registered 18:44:55.421 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING 18:44:55.421 main.c Ready: Success >>>> Account list: [ 0] <sip:192.168.2.131:5060>: does not register Online status: Online [ 1] <sip:192.168.2.131:5060;transport=TCP>: does not register Online status: Online *[ 2] sip:rahulvenkat at sip.actionvoip.com: 100/In Progress (expires=0) 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 >>> 18:44:55.457 pjsua_core.c .RX 553 bytes Response msg 401/REGISTER/cseq=1982 (rdata0x7fa8aa021a28) from UDP 77.72.174.128:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.131:5060;rport;branch=z9hG4bKPjVoDRo3TSu0dVPxlcKzf0nfYYJX5F6vmi From: <sip:rahulvenkat@xxxxxxxxxxxxxxxxxx>;tag=FYOhalxL4wJ6Vnh5eEKmEuLiL.z-UHx6 To: <sip:rahulvenkat at sip.actionvoip.com> Contact: sip:77.72.174.128:5060 Call-ID: kKWz8ZoGxLVibYYwIL.Hdqnt33gEfctJ CSeq: 1982 REGISTER Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE WWW-Authenticate: Digest realm="sip.actionvoip.com",nonce="1175386843",algorithm=MD5 Content-Length: 0 --end msg-- 18:44:55.457 sip_auth_clien ...Unable to set auth for tdta0x7fa8ab004600: can not find credential for sip.actionvoip.com/Digest 18:44:55.457 pjsua_acc.c ....SIP registration error: No suitable credential (PJSIP_ENOCREDENTIAL) [status=171101] >>>> Account list: [ 0] <sip:192.168.2.131:5060>: does not register Online status: Online [ 1] <sip:192.168.2.131:5060;transport=TCP>: does not register Online status: Online *[ 2] sip:rahulvenkat at sip.actionvoip.com: 401/Unauthorized (expires=-1) 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 18:45:51.640 pjsua_call.c !Making call with acc #2 to sip:0046706914265 at sip.actionvoip.com 18:45:51.640 pjsua_aud.c .Set sound device: capture=-1, playback=-2 18:45:51.640 pjsua_app.c ..Turning sound device ON 18:45:51.640 pjsua_aud.c ..Opening sound device PCM at 16000/1/20ms 18:45:51.663 ec0x7fa8a96397 ...AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=0 ms 18:45:51.663 pjsua_media.c .Call 0: initializing media.. 18:45:51.667 pjsua_media.c ..RTP socket reachable at 192.168.2.131:4000 18:45:51.667 pjsua_media.c ..RTCP socket reachable at 192.168.2.131:4001 18:45:51.667 pjsua_media.c ..Media index 0 selected for audio call 0 18:45:51.667 pjsua_core.c ....TX 1194 bytes Request msg INVITE/cseq=3787 (tdta0x7fa8ab006800) 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=z9hG4bKPjrGZia1baSzFkX-ZDzBL9oGbHMb1iQsny Max-Forwards: 70 From: sip:rahulvenkat@xxxxxxxxxxxxxxxxxx;tag=ryL1PKugLTrvA5-fTJQmusJ742fbtIy3 To: sip:0046706914265 at sip.actionvoip.com Contact: <sip:rahulvenkat at 192.168.2.131:5060;ob> Call-ID: t.fwSGLp.lfxTHu4ERnuDVVKh.skwtRK CSeq: 3787 INVITE Route: <sip:sip.actionvoip.com;lr> 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=- 3614863551 3614863551 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-- 18:45:51.668 pjsua_app.c .......Call 0 state changed to CALLING >>> 18:45:51.699 pjsua_core.c .RX 567 bytes Response msg 401/INVITE/cseq=3787 (rdata0x7fa8aa827228) from UDP 77.72.174.128:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.131:5060;rport;branch=z9hG4bKPjrGZia1baSzFkX-ZDzBL9oGbHMb1iQsny From: <sip:rahulvenkat@xxxxxxxxxxxxxxxxxx>;tag=ryL1PKugLTrvA5-fTJQmusJ742fbtIy3 To: <sip:0046706914265 at sip.actionvoip.com> Contact: sip:0046706914265 at 77.72.174.128:5060 Call-ID: t.fwSGLp.lfxTHu4ERnuDVVKh.skwtRK CSeq: 3787 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE WWW-Authenticate: Digest realm="sip.actionvoip.com",nonce="1175443093",algorithm=MD5 Content-Length: 0 --end msg-- 18:45:51.699 pjsua_core.c ..TX 398 bytes Request msg ACK/cseq=3787 (tdta0x7fa8aa08f600) 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=z9hG4bKPjrGZia1baSzFkX-ZDzBL9oGbHMb1iQsny Max-Forwards: 70 From: sip:rahulvenkat@xxxxxxxxxxxxxxxxxx;tag=ryL1PKugLTrvA5-fTJQmusJ742fbtIy3 To: sip:0046706914265 at sip.actionvoip.com Call-ID: t.fwSGLp.lfxTHu4ERnuDVVKh.skwtRK CSeq: 3787 ACK Route: <sip:sip.actionvoip.com;lr> Content-Length: 0 --end msg-- 18:45:51.699 sip_auth_clien ....Unable to set auth for tdta0x7fa8ab006800: can not find credential for sip.actionvoip.com/Digest 18:45:51.699 pjsua_app.c .....Call 0 is DISCONNECTED [reason=401 (Unauthorized)] 18:45:51.699 pjsua_app_comm ..... [DISCONNCTD] To: sip:0046706914265 at sip.actionvoip.com Call time: 00h:00m:00s, 1st res in 36 ms, conn in 0ms 18:45:51.700 pjsua_media.c .....Call 0: deinitializing media.. 18:45:52.700 pjsua_aud.c !Closing sound device after idle for 1 second(s) 18:45:52.700 pjsua_app.c .Turning sound device OFF 18:45:52.700 pjsua_aud.c .Closing AirPlay sound playback device and Built-in Microph sound capture device
