Hi, yes, the 9002 Linphone answers "400 Bad Request" to Asterisk's INVITE. Does it work in the other direction (9002 calling 9001)? Have you checked your codecs (Linphone is offering PCMA, PCMU and GSM, Asterisk just PCMU)?
Apparently, for debug logging Linphone, you should • open a windows shell prompt • go to c:\Program Files\Linphone • start Linphone like this: bin/linphone --logfile "c:\Temp\logs.txt" So maybe this way you can see some more information. -- marie On 10.09.2014, at 13:00, エムディーシー太郎 <[email protected]> wrote: > Thank you for your reply. > > After setting "pjsip set logger on", > the following message is displayed. > > It seems that the 9002(SIP client) refuse INVITE message. > Are SIP methods too many? > > Thanks, > MMEEGGAA > > -------------------- > <--- Transmitting SIP request (449 bytes) to UDP:192.168.177.180:16060 ---> > OPTIONS sip:[email protected]:16060 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.177.190:5060;rport;branch=z9hG4bKPj3c090235-9385-4cea-9c56-83f2755606d1 > From: > <sip:[email protected]>;tag=ef5b7ffd-6d54-4c46-8100-635d862f699b > To: <sip:[email protected]> > Contact: <sip:[email protected]:5060> > Call-ID: 6294a204-c645-41a8-ac27-e5aaa5f6c08c > CSeq: 44261 OPTIONS > Content-Length: 0 > > > <--- Received SIP response (333 bytes) from UDP:192.168.177.180:16060 ---> > SIP/2.0 200 Ok > Via: SIP/2.0/UDP > 192.168.177.190:5060;rport;branch=z9hG4bKPj3c090235-9385-4cea-9c56-83f2755606d1 > From: > <sip:[email protected]>;tag=ef5b7ffd-6d54-4c46-8100-635d862f699b > To: <sip:[email protected]>;tag=EF1my > Call-ID: 6294a204-c645-41a8-ac27-e5aaa5f6c08c > CSeq: 44261 OPTIONS > > > <--- Transmitting SIP request (443 bytes) to UDP:192.168.177.191:5060 ---> > OPTIONS sip:[email protected] SIP/2.0 > Via: SIP/2.0/UDP > 192.168.177.190:5060;rport;branch=z9hG4bKPj98982748-5a3d-4a4c-9d90-ad4988e8d933 > From: > <sip:[email protected]>;tag=73ae5dfe-d5b7-4c9b-9529-5198492f5c49 > To: <sip:[email protected]> > Contact: <sip:[email protected]:5060> > Call-ID: eb7a8bee-b06f-4a30-a5d9-eb437e104f76 > CSeq: 16803 OPTIONS > Content-Length: 0 > > > <--- Received SIP response (333 bytes) from UDP:192.168.177.191:5060 ---> > SIP/2.0 200 Ok > Via: SIP/2.0/UDP > 192.168.177.190:5060;rport;branch=z9hG4bKPj98982748-5a3d-4a4c-9d90-ad4988e8d933 > From: > <sip:[email protected]>;tag=73ae5dfe-d5b7-4c9b-9529-5198492f5c49 > To: <sip:[email protected]>;tag=hSl7b > Call-ID: eb7a8bee-b06f-4a30-a5d9-eb437e104f76 > CSeq: 16803 OPTIONS > > > <--- Received SIP request (1170 bytes) from UDP:192.168.177.180:16060 ---> > INVITE sip:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.G4FYrVaiY;rport > From: <sip:[email protected]>;tag=~yIpJRFo9 > To: sip:[email protected] > CSeq: 20 INVITE > Call-ID: 2c1KLd1INo > Max-Forwards: 70 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, > INFO > Content-Type: application/sdp > Content-Length: 633 > User-Agent: Linphone/3.6.99 (belle-sip/1.2.4) > Contact: > <sip:[email protected]:16060>;+sip.instance="<urn:uuid:aa35e441-baa3-4e07-af2f-cd1ff2dc99a2>" > > v=0 > o=9001 2189 3894 IN IP4 192.168.177.180 > s=Talk > c=IN IP4 192.168.177.180 > t=0 0 > a=ice-pwd:000030810000373f00004003 > a=ice-ufrag:00004c02 > m=audio 17590 RTP/AVP 0 110 3 8 101 > c=IN IP4 61.117.138.218 > a=rtpmap:110 speex/8000 > a=fmtp:110 vbr=on > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > a=rtcp:21548 > a=candidate:1 1 UDP 2130706431 192.168.177.180 7078 typ host > a=candidate:1 2 UDP 2130706430 192.168.177.180 7079 typ host > a=candidate:2 1 UDP 1694498815 XXX.XXX.XXX.XXX 17590 typ srflx raddr > 192.168.177.180 rport 7078 > a=candidate:2 2 UDP 1694498814 XXX.XXX.XXX.XXX 21548 typ srflx raddr > 192.168.177.180 rport 7079 > > <--- Transmitting SIP response (431 bytes) to UDP:192.168.177.180:16060 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP > 192.168.177.180:16060;rport;received=192.168.177.180;branch=z9hG4bK.G4FYrVaiY > Call-ID: 2c1KLd1INo > From: <sip:[email protected]>;tag=~yIpJRFo9 > To: <sip:[email protected]>;tag=z9hG4bK.G4FYrVaiY > CSeq: 20 INVITE > WWW-Authenticate: Digest > realm="asterisk",nonce="1410336707/cd97e01134333d7d5769e49872f750a4",opaque="58e109d10f49a371",algorithm=md5,qop="auth" > Content-Length: 0 > > > <--- Received SIP request (373 bytes) from UDP:192.168.177.180:16060 ---> > ACK sip:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.G4FYrVaiY;rport > Call-ID: 2c1KLd1INo > From: <sip:[email protected]>;tag=~yIpJRFo9 > To: <sip:[email protected]>;tag=z9hG4bK.G4FYrVaiY > Contact: > <sip:[email protected]:16060>;+sip.instance="<urn:uuid:aa35e441-baa3-4e07-af2f-cd1ff2dc99a2>" > Max-Forwards: 70 > CSeq: 20 ACK > > > <--- Received SIP request (1428 bytes) from UDP:192.168.177.180:16060 ---> > INVITE sip:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.00879CXMH;rport > From: <sip:[email protected]>;tag=~yIpJRFo9 > To: sip:[email protected] > CSeq: 21 INVITE > Call-ID: 2c1KLd1INo > Max-Forwards: 70 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, > INFO > Content-Type: application/sdp > Content-Length: 633 > User-Agent: Linphone/3.6.99 (belle-sip/1.2.4) > Contact: > <sip:[email protected]:16060>;+sip.instance="<urn:uuid:aa35e441-baa3-4e07-af2f-cd1ff2dc99a2>" > Authorization: Digest realm="asterisk", > nonce="1410336707/cd97e01134333d7d5769e49872f750a4", > opaque="58e109d10f49a371", username="9001", uri="sip:[email protected]", > response="a822c66fe1c1d30492beeb08e6daaae5", cnonce="faaa92d5", nc=00000001, > qop=auth > > v=0 > o=9001 2189 3894 IN IP4 192.168.177.180 > s=Talk > c=IN IP4 192.168.177.180 > t=0 0 > a=ice-pwd:000030810000373f00004003 > a=ice-ufrag:00004c02 > m=audio 17590 RTP/AVP 0 110 3 8 101 > c=IN IP4 61.117.138.218 > a=rtpmap:110 speex/8000 > a=fmtp:110 vbr=on > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > a=rtcp:21548 > a=candidate:1 1 UDP 2130706431 192.168.177.180 7078 typ host > a=candidate:1 2 UDP 2130706430 192.168.177.180 7079 typ host > a=candidate:2 1 UDP 1694498815 XXX.XXX.XXX.XXX 17590 typ srflx raddr > 192.168.177.180 rport 7078 > a=candidate:2 2 UDP 1694498814 XXX.XXX.XXX.XXX 21548 typ srflx raddr > 192.168.177.180 rport 7079 > > <--- Transmitting SIP response (256 bytes) to UDP:192.168.177.180:16060 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 192.168.177.180:16060;rport;received=192.168.177.180;branch=z9hG4bK.00879CXMH > Call-ID: 2c1KLd1INo > From: <sip:[email protected]>;tag=~yIpJRFo9 > To: <sip:[email protected]> > CSeq: 21 INVITE > Content-Length: 0 > > > -- Executing [9002@internal:1] Dial("PJSIP/9001-00000006", > "PJSIP/9002,20") in new stack > -- Called PJSIP/9002 > debug > == debug1 (0|0:0/0/0) > == debug2 (2|1:0/0/0) > <--- Transmitting SIP request (910 bytes) to UDP:192.168.177.191:5060 ---> > INVITE sip:[email protected] SIP/2.0 > Via: SIP/2.0/UDP > 192.168.177.190:5060;rport;branch=z9hG4bKPjf90aa758-0538-4145-b2c9-7d945da6f124 > From: <sip:[email protected]>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b > To: <sip:[email protected]> > Contact: <sip:[email protected]:5060> > Call-ID: 8a59fbf2-9cae-48b7-88de-1a2fed4ec7c9 > CSeq: 18942 INVITE > Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, > PRACK, REGISTER, MESSAGE, REFER > Supported: 100rel, timer, replaces, norefersub > Session-Expires: 1800 > Min-SE: 90 > Content-Type: application/sdp > Content-Length: 273 > > v=0 > o=- 278980317 278980317 IN IP4 localhost.localdomain > s=Asterisk > c=IN IP4 192.168.177.190 > t=0 0 > m=audio 10338 RTP/AVP 0 101 > c=IN IP4 192.168.177.190 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=maxptime:150 > a=sendrecv > > <--- Received SIP response (309 bytes) from UDP:192.168.177.191:5060 ---> > SIP/2.0 400 Bad request > Via: SIP/2.0/UDP > 192.168.177.190:5060;rport;branch=z9hG4bKPjf90aa758-0538-4145-b2c9-7d945da6f124 > From: <sip:[email protected]>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b > To: <sip:[email protected]>;tag=bajXh > Call-ID: 8a59fbf2-9cae-48b7-88de-1a2fed4ec7c9 > CSeq: 18942 INVITE > > > <--- Transmitting SIP request (339 bytes) to UDP:192.168.177.191:5060 ---> > ACK sip:[email protected] SIP/2.0 > Via: SIP/2.0/UDP > 192.168.177.190:5060;rport;branch=z9hG4bKPjf90aa758-0538-4145-b2c9-7d945da6f124 > From: <sip:[email protected]>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b > To: <sip:[email protected]>;tag=bajXh > Call-ID: 8a59fbf2-9cae-48b7-88de-1a2fed4ec7c9 > CSeq: 18942 ACK > Content-Length: 0 > > -- PJSIP/9002-00000007 answered PJSIP/9001-00000006 > -- PJSIP/9002-00000007 answered PJSIP/9001-00000006 > > == Everyone is busy/congested at this time (1:0/0/1) > -- Auto fallthrough, channel 'PJSIP/9001-00000006' status is 'CHANUNAVAIL' > <--- Transmitting SIP response (334 bytes) to UDP:192.168.177.180:16060 ---> > SIP/2.0 503 Service Unavailable > Via: SIP/2.0/UDP > 192.168.177.180:16060;rport;received=192.168.177.180;branch=z9hG4bK.00879CXMH > Call-ID: 2c1KLd1INo > From: <sip:[email protected]>;tag=~yIpJRFo9 > To: <sip:[email protected]>;tag=aead10f9-2194-48dd-bf38-0cc78bff561f > CSeq: 21 INVITE > Reason: Q.850;cause=34 > Content-Length: 0 > > > <--- Received SIP response (306 bytes) from UDP:192.168.177.191:5060 ---> > SIP/2.0 400 Bad request > Via: SIP/2.0/UDP > 192.168.177.190:5060;rport;branch=z9hG4bKPjf90aa758-0538-4145-b2c9-7d945da6f124 > From: <sip:[email protected]>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b > To: <sip:[email protected]>;tag=bajXh > Call-ID: 8a59fbf2-9cae-48b7-88de-1a2fed4ec7c9 > CSeq: 18942 ACK > -------------------- > > 2014-09-05 19:24 GMT+09:00 Joshua Colp <[email protected]>: > エムディーシー太郎 wrote: > Hi All, > > I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code > on CentOS7. > --https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject > > <snip> > > > ---------- > 2. dial from 9001 to 9002 > > *CLI> -- Executing [9002@internal:1] Dial("PJSIP/9001-00000000", > "PJSIP/9002,20") in new stack > -- Called PJSIP/9002 > == Everyone is busy/congested at this time (1:0/0/1) > -- Auto fallthrough, channel 'PJSIP/9001-00000000' status is > 'CHANUNAVAIL' > > What is shown if you do "pjsip set logger on" and then try to place the call? > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
