Hi, I'm working on application that should call a phone A and transfer the call to B (with pjsua_call_xfer). the transfer it did well but there is no communication between the two phones (cann't hear neither talk). Note that I deal with an Asterisk Server. Any idea is welcome. thanks 10:49:25.197 os_core_unix.c pjlib 0.5.10.4 for POSIX initialized 10:49:25.197 sip_endpoint.c Creating endpoint instance... 10:49:25.199 pjlib select() I/O Queue created (0x811b66c) 10:49:25.200 sip_endpoint.c Module "mod-msg-print" registered 10:49:25.200 sip_transport. Transport manager created. 10:49:25.200 sip_endpoint.c Module "mod-pjsua-log" registered 10:49:25.200 sip_endpoint.c Module "mod-tsx-layer" registered 10:49:25.200 sip_endpoint.c Module "mod-stateful-util" registered 10:49:25.200 sip_endpoint.c Module "mod-ua" registered 10:49:25.201 sip_endpoint.c Module "mod-pjsua" registered 10:49:25.201 sip_endpoint.c Module "mod-invite" registered 10:49:25.271 pasound.c PortAudio sound library initialized, status=0 10:49:25.271 pasound.c PortAudio host api count=1 10:49:25.271 pasound.c Sound device count=1 10:49:25.272 pjlib select() I/O Queue created (0x8153f24) 10:49:25.273 sip_endpoint.c Module "mod-evsub" registered 10:49:25.274 sip_endpoint.c Module "mod-presence" registered 10:49:25.274 sip_endpoint.c Module "mod-refer" registered 10:49:25.274 sip_endpoint.c Module "mod-pjsua-pres" registered 10:49:25.274 sip_endpoint.c Module "mod-pjsua-im" registered 10:49:25.274 sip_endpoint.c Module "mod-pjsua-options" registered 10:49:25.274 pjsua_core.c 1 SIP worker threads created 10:49:25.274 pjsua_core.c pjsua version 0.5.10.4 for i686-pc-linux-gnu initialized 10:49:25.275 pjsua_core.c SIP UDP socket reachable at 192.168.1.213:50060 10:49:25.275 udp0x81734cc SIP UDP transport started, published address is 192.168.1.213:50060 10:49:25.276 pjsua_acc.c Account SIP:131 at localhost added with id 0 10:49:25.276 pjsua_core.c TX 350 bytes Request msg REGISTER/cseq=64678 (tdta0x81750f4) to UDP 127.0.0.1:5060: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000367458b6bc623 Max-Forwards: 70 From: ;tag=25e0000267458b6bc623 To: Call-ID: 25e0000167458b6bc623 CSeq: 64678 REGISTER Contact: Expires: 55 Content-Length: 0 --end msg-- 10:49:25.276 pjsua_acc.c Registration sent 10:49:25.276 sip_endpoint.c Module "mod-handle-requests" registered 10:49:25.276 pjsua_call.c Making call with acc #0 to SIP:131 at 192.168.1.10 10:49:25.277 pjsua_core.c TX 900 bytes Request msg INVITE/cseq=1714636915 (tdta0x817833c) to UDP 192.168.1.10:5060: INVITE sip:131 at 192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000667458b6bc623 Max-Forwards: 70 From: sip:131@localhost;tag=25e0000467458b6bc623 To: sip:131 at 192.168.1.10 Contact: Call-ID: 25e0000567458b6bc623 CSeq: 1714636915 INVITE Allow: INVITE, ACK, BYE, CANCEL, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS Supported: replaces, norefersub Content-Type: application/sdp Content-Length: 403 v=0 o=- 3404648965 3404648965 IN IP4 0.0.0.0 s=pjmedia c=IN IP4 0.0.0.0 t=0 0 m=audio 0 RTP/AVP 103 102 104 117 3 0 8 101 a=rtcp:0 IN IP4 0.0.0.0 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:117 iLBC/8000 a=fmtp:117 mode=20 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- 10:49:25.277 makeCall -- Call 0 state=CALLING 10:49:25.278 pjsua_core.c RX 514 bytes Response msg 407/INVITE/cseq=1714636915 (rdata0x81738dc) from UDP 192.168.1.10:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.213:50060;branch=z9hG4bKPj25e0000667458b6bc623;received=192.168.1.213;rport=50060 From: sip:131@localhost;tag=25e0000467458b6bc623 To: sip:131 at 192.168.1.10;tag=as785e783b Call-ID: 25e0000567458b6bc623 CSeq: 1714636915 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="63da021a" Content-Length: 0 --end msg-- 10:49:25.278 pjsua_core.c TX 299 bytes Request msg ACK/cseq=1714636915 (tdta0x817aaf4) to UDP 192.168.1.10:5060: ACK sip:131 at 192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000667458b6bc623 Max-Forwards: 70 From: sip:131@localhost;tag=25e0000467458b6bc623 To: sip:131 at 192.168.1.10;tag=as785e783b Call-ID: 25e0000567458b6bc623 CSeq: 1714636915 ACK Content-Length: 0 --end msg-- 10:49:25.278 pjsua_core.c TX 1068 bytes Request msg INVITE/cseq=1714636916 (tdta0x817833c) to UDP 192.168.1.10:5060: INVITE sip:131 at 192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000767458b6bc623 Max-Forwards: 70 From: sip:131@localhost;tag=25e0000467458b6bc623 To: sip:131 at 192.168.1.10 Contact: Call-ID: 25e0000567458b6bc623 CSeq: 1714636916 INVITE Allow: INVITE, ACK, BYE, CANCEL, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS Supported: replaces, norefersub Proxy-Authorization: Digest username="131", realm="asterisk", nonce="63da021a", uri="sip:131 at 192.168.1.10", response="a91966d158535db8463ac523fe353e99", algorithm=md5 Content-Type: application/sdp Content-Length: 403 v=0 o=- 3404648965 3404648965 IN IP4 0.0.0.0 s=pjmedia c=IN IP4 0.0.0.0 t=0 0 m=audio 0 RTP/AVP 103 102 104 117 3 0 8 101 a=rtcp:0 IN IP4 0.0.0.0 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:117 iLBC/8000 a=fmtp:117 mode=20 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- 10:49:25.279 makeCall -- Call 0 state=CALLING 10:49:25.280 pjsua_core.c RX 431 bytes Response msg 100/INVITE/cseq=1714636916 (rdata0x81738dc) from UDP 192.168.1.10:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.213:50060;branch=z9hG4bKPj25e0000767458b6bc623;received=192.168.1.213;rport=50060 From: sip:131@localhost;tag=25e0000467458b6bc623 To: sip:131 at 192.168.1.10 Call-ID: 25e0000567458b6bc623 CSeq: 1714636916 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --end msg-- 10:49:25.432 pjsua_core.c RX 447 bytes Response msg 180/INVITE/cseq=1714636916 (rdata0x81738dc) from UDP 192.168.1.10:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.213:50060;branch=z9hG4bKPj25e0000767458b6bc623;received=192.168.1.213;rport=50060 From: sip:131@localhost;tag=25e0000467458b6bc623 To: sip:131 at 192.168.1.10;tag=as2d7b4585 Call-ID: 25e0000567458b6bc623 CSeq: 1714636916 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --end msg-- 10:49:25.432 makeCall -- Call 0 state=EARLY 10:49:25.779 pjsua_core.c TX 350 bytes Request msg REGISTER/cseq=64678 (tdta0x81750f4) to UDP 127.0.0.1:5060: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000367458b6bc623 Max-Forwards: 70 From: ;tag=25e0000267458b6bc623 To: Call-ID: 25e0000167458b6bc623 CSeq: 64678 REGISTER Contact: Expires: 55 Content-Length: 0 --end msg-- 10:49:26.779 pjsua_core.c TX 350 bytes Request msg REGISTER/cseq=64678 (tdta0x81750f4) to UDP 127.0.0.1:5060: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000367458b6bc623 Max-Forwards: 70 From: ;tag=25e0000267458b6bc623 To: Call-ID: 25e0000167458b6bc623 CSeq: 64678 REGISTER Contact: Expires: 55 Content-Length: 0 --end msg-- 10:49:26.931 pjsua_core.c RX 762 bytes Response msg 200/INVITE/cseq=1714636916 (rdata0x81738dc) from UDP 192.168.1.10:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.213:50060;branch=z9hG4bKPj25e0000767458b6bc623;received=192.168.1.213;rport=50060 From: sip:131@localhost;tag=25e0000467458b6bc623 To: sip:131 at 192.168.1.10;tag=as2d7b4585 Call-ID: 25e0000567458b6bc623 CSeq: 1714636916 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 28818 28818 IN IP4 192.168.1.10 s=session c=IN IP4 192.168.1.10 t=0 0 m=audio 12538 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --end msg-- 10:49:26.931 makeCall -- Call 0 state=CONNECTING 10:49:26.931 pjsua_call.c SDP negotiation has failed: No active media stream after negotiation (PJMEDIA_SDPNEG_ENOMEDIA) [status=220048] 10:49:26.931 pjsua_core.c TX 299 bytes Request msg ACK/cseq=1714636916 (tdta0x817dac4) to UDP 192.168.1.10:5060: ACK sip:131 at 192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000867458b6bc623 Max-Forwards: 70 From: sip:131@localhost;tag=25e0000467458b6bc623 To: sip:131 at 192.168.1.10;tag=as2d7b4585 Call-ID: 25e0000567458b6bc623 CSeq: 1714636916 ACK Content-Length: 0 --end msg-- 10:49:26.931 makeCall -- Call 0 state=CONFIRMED 10:49:26.932 pjsua_core.c TX 524 bytes Request msg REFER/cseq=1714636917 (tdta0x817ded4) to UDP 192.168.1.10:5060: REFER sip:131 at 192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000967458b6bc623 Max-Forwards: 70 From: sip:131@localhost;tag=25e0000467458b6bc623 To: sip:131 at 192.168.1.10;tag=as2d7b4585 Contact: Call-ID: 25e0000567458b6bc623 CSeq: 1714636917 REFER Event: refer Expires: 300 Accept: message/sipfrag;version=2.0 Allow-Events: presence, refer Refer-To: SIP:5147729433 at 192.168.1.10 Referred-By: SIP:131 at localhost Content-Length: 0 --end msg-- 10:49:26.932 evsub0x817da74 Subscription state changed NULL --> SENT 10:49:26.932 pjsua_core.c RX 447 bytes Response msg 202/REFER/cseq=1714636917 (rdata0x81738dc) from UDP 192.168.1.10:5060: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.1.213:50060;branch=z9hG4bKPj25e0000967458b6bc623;received=192.168.1.213;rport=50060 From: sip:131@localhost;tag=25e0000467458b6bc623 To: sip:131 at 192.168.1.10;tag=as2d7b4585 Call-ID: 25e0000567458b6bc623 CSeq: 1714636917 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --end msg-- 10:49:26.932 evsub0x817da74 Subscription state changed SENT --> ACCEPTED 10:49:26.933 makeCall Call 0: transfer status=100 (Accepted) 10:49:26.933 pjsua_core.c RX 571 bytes Request msg NOTIFY/cseq=102 (rdata0x81738dc) from UDP 192.168.1.10:5060: NOTIFY sip:131 at 192.168.1.213:50060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK03d9d68f;rport From: sip:131@192.168.1.10;tag=as2d7b4585 To: sip:131 at localhost;tag=25e0000467458b6bc623 Contact: Call-ID: 25e0000567458b6bc623 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=1714636917 Subscription-state: active Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 21 SIP/2.0 183 Ringing --end msg-- 10:49:26.933 pjsua_core.c TX 447 bytes Response msg 200/NOTIFY/cseq=102 (tdta0x817bafc) to UDP 192.168.1.10:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.10:5060;rport=5060;received=192.168.1.10;branch=z9hG4bK03d9d68f Call-ID: 25e0000567458b6bc623 From: ;tag=as2d7b4585 To: ;tag=25e0000467458b6bc623 CSeq: 102 NOTIFY Contact: Allow: INVITE, ACK, BYE, CANCEL, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS Supported: replaces, norefersub Content-Length: 0 --end msg-- 10:49:26.933 evsub0x817da74 Subscription state changed ACCEPTED --> ACTIVE 10:49:26.933 makeCall Call 0: transfer status=183 ( Ringing) 10:49:26.933 pjsua_core.c RX 588 bytes Request msg NOTIFY/cseq=103 (rdata0x81738dc) from UDP 192.168.1.10:5060: NOTIFY sip:131 at 192.168.1.213:50060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1c6b1e6e;rport From: sip:131@192.168.1.10;tag=as2d7b4585 To: sip:131 at localhost;tag=25e0000467458b6bc623 Contact: Call-ID: 25e0000567458b6bc623 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=1714636917 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 16 SIP/2.0 200 Ok --end msg-- 10:49:26.933 pjsua_core.c TX 447 bytes Response msg 200/NOTIFY/cseq=103 (tdta0x8181e3c) to UDP 192.168.1.10:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.10:5060;rport=5060;received=192.168.1.10;branch=z9hG4bK1c6b1e6e Call-ID: 25e0000567458b6bc623 From: ;tag=as2d7b4585 To: ;tag=25e0000467458b6bc623 CSeq: 103 NOTIFY Contact: Allow: INVITE, ACK, BYE, CANCEL, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS Supported: replaces, norefersub Content-Length: 0 --end msg-- 10:49:26.933 evsub0x817da74 Subscription state changed ACTIVE --> TERMINATED 10:49:26.933 pjsua_call.c Xfer client subscription terminated 10:49:26.933 makeCall Call 0: transfer status=200 ( Ok) [final] 10:49:26.933 makeCall Call 0: call transfered successfully, disconnecting call 10:49:26.934 pjsua_core.c TX 299 bytes Request msg BYE/cseq=1714636918 (tdta0x8183dec) to UDP 192.168.1.10:5060: BYE sip:131 at 192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.213:50060;rport;branch=z9hG4bKPj25e0000a67458b6bc623 Max-Forwards: 70 From: sip:131@localhost;tag=25e0000467458b6bc623 To: sip:131 at 192.168.1.10;tag=as2d7b4585 Call-ID: 25e0000567458b6bc623 CSeq: 1714636918 BYE Content-Length: 0 --end msg-- 10:49:26.934 sip_regc.c Unable to send request, regc has another transaction pending 10:49:26.934 pjsua_acc.c Unable to create/send REGISTER: Object is busy (PJSIP_EBUSY) [status=171001] 10:49:26.934 pjsua_acc.c Account id 0 deleted 10:49:26.940 pjsua_core.c RX 439 bytes Response msg 200/BYE/cseq=1714636918 (rdata0x81738dc) from UDP 192.168.1.10:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.213:50060;branch=z9hG4bKPj25e0000a67458b6bc623;received=192.168.1.213;rport=50060 From: sip:131@localhost;tag=25e0000467458b6bc623 To: sip:131 at 192.168.1.10;tag=as2d7b4585 Call-ID: 25e0000567458b6bc623 CSeq: 1714636918 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 _________________________________________________________________ D?couvrez Windows Live Spaces et cr?ez votre site Web perso en 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