Hi, there was a suspicious failure when we receive 200/OK for INVITE: > 10:49:26.931 pjsua_call.c SDP negotiation has failed: No active media stream after negotiation (PJMEDIA_SDPNEG_ENOMEDIA) [status=220048] I'm not sure why the SDP was rejected, but one thing for sure, you're using a pretty old pjsip, so maybe new version will work better. Can you try with using pjsua application from the 0.8 version? cheers, -benny boujbel zied wrote: > 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
