Hi, I've already tried to upload it to here, so for now i will just copy the log. --end msg-- 19:52:35.236 pjsua_app.c Call 0 is DISCONNECTED [reason=200 (Normal call clearing)] 19:52:35.272 pjsua_media.c Media session for call 0 is destroyed 19:52:36.271 pjsua_media.c Closing sound device after idle for 1 seconds 19:52:36.271 pjsua_media.c Closing iPhone Sound Device sound playback device and iPhone Sound Device sound capture device 19:52:59.091 pjsua_core.c RX 823 bytes Request msg INVITE/cseq=225308893 (rdata0x84d064) from UDP 80.179.0.4:5060: INVITE sip:722511874 at 192.168.1.59:5060 SIP/2.0 Via: SIP/2.0/UDP 80.179.0.4:5060;branch=z9hG4bKgqo3mo00d0b10koup7c0.1 From: "a 1"<sip:1867@xxxxxxxxxxx <sip%3A1867 at 072.012.net> ;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- To: "a 1"<sip:722511874 at 072.012.net <sip%3A722511874 at 072.012.net>> Call-ID: SD1m0gd01-cbf46a742831cc7b6fa4564187179755-06al420 CSeq: 225308893 INVITE Contact: <sip:1867 at 80.179.0.4:5060;transport=udp> Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY Accept: multipart/mixed,application/media_control+xml,application/sdp Supported: Max-Forwards: 4 Content-Type: application/sdp Content-Length: 227 v=0 o=BroadWorks 162350108 1 IN IP4 212.199.138.8 s=- c=IN IP4 212.199.138.8 t=0 0 m=audio 21758 RTP/AVP 18 101 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv --end msg-- 19:52:59.092 pjsua_media.c Media index 0 selected for call 1 19:52:59.094 pjsua_core.c TX 344 bytes Response msg 100/INVITE/cseq=225308893 (tdta0x869c00) to UDP 80.179.0.4:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 80.179.0.4:5060 ;received=80.179.0.4;branch=z9hG4bKgqo3mo00d0b10koup7c0.1 Call-ID: SD1m0gd01-cbf46a742831cc7b6fa4564187179755-06al420 From: "a 1" <sip:1867@xxxxxxxxxxx <sip%3A1867 at 072.012.net> ;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- To: "a 1" <sip:722511874 at 072.012.net <sip%3A722511874 at 072.012.net>> CSeq: 225308893 INVITE Content-Length: 0 --end msg-- 19:52:59.104 pjsua_media.c Opening sound device PCM at 16000/1/20ms 19:52:59.235 ec0x24d100 Echo suppressor created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=140 ms 19:52:59.459 conference.c Port 2 (ring) transmitting to port 0 (iPhone Sound Device) 19:52:59.468 pjsua_app.c Incoming call for account 2! From: "a 1" <sip:1867@xxxxxxxxxxx <sip%3A1867 at 072.012.net>;user=phone> To: "a 1" <sip:722511874 at 072.012.net <sip%3A722511874 at 072.012.net>> Press a to answer or h to reject call 2010-03-01 19:52:59.498 MySIPhoneV2[1527:207] Could not load the "siax.png" image referenced from a nib in the bundle with identifier "com.zemingo.MySIPhoneV2" 19:53:00.564 strm0x871b74 VAD temporarily disabled 19:53:00.565 strm0x871b74 Encoder stream started 19:53:00.565 strm0x871b74 Decoder stream started 19:53:00.565 pjsua_media.c Media updates, stream #0: G729 (sendrecv) 19:53:00.566 conference.c Port 2 (ring) stop transmitting to port 0 (iPhone Sound Device) 19:53:00.566 conference.c Port 3 (sip:1867 at 072.012.net<sip%3A1867 at 072.012.net>;user=phone) transmitting to port 0 (iPhone Sound Device) 19:53:00.566 conference.c Port 0 (iPhone Sound Device) transmitting to port 3 (sip:1867 at 072.012.net <sip%3A1867 at 072.012.net>;user=phone) 19:53:00.566 pjsua_app.c Media for call 1 is active 19:53:00.566 pjsua_core.c TX 869 bytes Response msg 200/INVITE/cseq=225308893 (tdta0x869c00) to UDP 80.179.0.4:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 80.179.0.4:5060 ;received=80.179.0.4;branch=z9hG4bKgqo3mo00d0b10koup7c0.1 Call-ID: SD1m0gd01-cbf46a742831cc7b6fa4564187179755-06al420 From: "a 1" <sip:1867@xxxxxxxxxxx <sip%3A1867 at 072.012.net> ;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- To: "a 1" <sip:722511874 at 072.012.net <sip%3A722511874 at 072.012.net> >;tag=xbC9.pmCU9.-vEm7FhtxB25YcK.RmCye CSeq: 225308893 INVITE Contact: <sip:722511874 at 192.168.1.59:5060> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 274 v=0 o=- 3476454779 3476454780 IN IP4 192.168.1.59 s=pjmedia c=IN IP4 192.168.1.59 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 18 101 a=rtcp:4003 IN IP4 192.168.1.59 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- 19:53:00.569 pjsua_app.c Call 1 state changed to CONNECTING 19:53:01.096 pjsua_core.c TX 869 bytes Response msg 200/INVITE/cseq=225308893 (tdta0x869c00) to UDP 80.179.0.4:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 80.179.0.4:5060 ;received=80.179.0.4;branch=z9hG4bKgqo3mo00d0b10koup7c0.1 Call-ID: SD1m0gd01-cbf46a742831cc7b6fa4564187179755-06al420 From: "a 1" <sip:1867@xxxxxxxxxxx <sip%3A1867 at 072.012.net> ;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- To: "a 1" <sip:722511874 at 072.012.net <sip%3A722511874 at 072.012.net> >;tag=xbC9.pmCU9.-vEm7FhtxB25YcK.RmCye CSeq: 225308893 INVITE Contact: <sip:722511874 at 192.168.1.59:5060> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 274 v=0 o=- 3476454779 3476454780 IN IP4 192.168.1.59 s=pjmedia c=IN IP4 192.168.1.59 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 18 101 a=rtcp:4003 IN IP4 192.168.1.59 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- 19:53:02.111 pjsua_core.c TX 869 bytes Response msg 200/INVITE/cseq=225308893 (tdta0x869c00) to UDP 80.179.0.4:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 80.179.0.4:5060 ;received=80.179.0.4;branch=z9hG4bKgqo3mo00d0b10koup7c0.1 Call-ID: SD1m0gd01-cbf46a742831cc7b6fa4564187179755-06al420 From: "a 1" <sip:1867@xxxxxxxxxxx <sip%3A1867 at 072.012.net> ;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- To: "a 1" <sip:722511874 at 072.012.net <sip%3A722511874 at 072.012.net> >;tag=xbC9.pmCU9.-vEm7FhtxB25YcK.RmCye CSeq: 225308893 INVITE Contact: <sip:722511874 at 192.168.1.59:5060> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 274 v=0 o=- 3476454779 3476454780 IN IP4 192.168.1.59 s=pjmedia c=IN IP4 192.168.1.59 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 18 101 a=rtcp:4003 IN IP4 192.168.1.59 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- 19:53:02.211 pjsua_core.c RX 450 bytes Request msg ACK/cseq=225308893 (rdata0x84d064) from UDP 80.179.0.4:5060: ACK sip:722511874 at 192.168.1.59:5060 SIP/2.0 Via: SIP/2.0/UDP 80.179.0.4:5060;branch=z9hG4bK7d5nfa1060s0nmg3r3k0.1 From: "a 1" <sip:1867@xxxxxxxxxxx <sip%3A1867 at 072.012.net> ;user=phone>;tag=SD1m0gd01-1080909488-1267465970104- To: "a 1" <sip:722511874 at 072.012.net <sip%3A722511874 at 072.012.net> >;tag=xbC9.pmCU9.-vEm7FhtxB25YcK.RmCye Call-ID: SD1m0gd01-cbf46a742831cc7b6fa4564187179755-06al420 CSeq: 225308893 ACK Contact: <sip:1867 at 80.179.0.4:5060;transport=udp> Max-Forwards: 4 Content-Length: 0 --end msg-- 19:53:02.214 pjsua_app.c Call 1 state changed to CONFIRMED Regards, Maya -------------- next part -------------- An HTML attachment was scrubbed... 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