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
