[G1 Android Development] dhkr123:
Found a bug. Don't know if its AOSP or something broken though.

Integrated SIP seems to work very well when signalling is on UDP, but get this when trying to signal on TCP:

(personal information removed...)

(observe the crash about halfway down...)

I/ActivityManager( 135): Starting: Intent { act=android.intent.action.CALL_PRIVILEGED dat=tel:xxx-xxx-xxxx flg=0x10000000 cmp=com.android.phone/.PrivilegedOutgoingCallBroadcaster } from pid 322
D/PhoneUtils( 286): checkAndCopyPhoneProviderExtras: some or all extras are missing.
D/PhoneUtils( 286): checkAndCopyPhoneProviderExtras: some or all extras are missing.
I/ActivityManager( 135): Starting: Intent { act=android.phone.extra.NEW_CALL_INTENT dat=tel:xxx-xxx-xxxx flg=0x10000000 cmp=com.android.phone/.SipCallOptionHandler (has extras) } from pid 286
V/SipCallOptionHandler( 286): Call option is SIP_ALWAYS
I/ActivityManager( 135): Displayed com.android.phone/.SipCallOptionHandler: +420ms (total +643ms)
D/dalvikvm( 286): GetFieldID: unable to find field Ljava/util/HashMap;.loadFactor:F
D/dalvikvm( 286): GetFieldID: unable to find field Ljava/util/HashMap;.loadFactor:F
D/dalvikvm( 286): GetFieldID: unable to find field Ljava/util/HashMap;.loadFactor:F
D/dalvikvm( 286): GetFieldID: unable to find field Ljava/util/HashMap;.loadFactor:F
V/SipCallOptionHandler( 286): primary SIP URI is sip:MYPBXESORGACCOUNT@pbxes.org:5060
I/ActivityManager( 135): Starting: Intent { act=android.intent.action.CALL dat=tel:xxx-xxx-xxxx flg=0x10000000 cmp=com.android.phone/.InCallScreen (has extras) } from pid 286
D/dalvikvm( 135): GC_CONCURRENT freed 707K, 47% free 4278K/8007K, external 1500K/2012K, paused 20ms+24ms
V/SipPhone( 286): canDial(): serviceState = 0
V/SipPhone( 286): canDial(): disableCall = false
V/SipPhone( 286): canDial(): ringingCall: IDLE
V/SipPhone( 286): canDial(): foregndCall: IDLE
V/SipPhone( 286): canDial(): backgndCall: IDLE
V/Call ( 286): +***+ call state changed: IDLE --> DIALING: IDLE:com.android.internal.telephony.sip.SipPhone$SipCall@407e3308: on phone Handler{40812ac8} 1
D/SipPhone( 286): ^^^ new phone state: OFFHOOK
I/AudioService( 135): AudioFocus requestAudioFocus() from AudioFocus_For_Phone_Ring_And_Calls
D/SipHelper( 286): send INVITE: INVITE sip:MYREALPHONENUMBERWASHERE@pbxes.org SIP/2.0
D/SipHelper( 286): Call-ID: 84e24779505aa6608ce9be97b509ce57@192.168.1.103
D/SipHelper( 286): CSeq: 6901 INVITE
D/SipHelper( 286): From: "MYREALNAMEWASHERE" <sip:MYPBXESORGACCOUNT@pbxes.org:5060>;tag=2667853903
D/SipHelper( 286): To: <sip:MYREALPHONENUMBERWASHERE@pbxes.org>
D/SipHelper( 286): Via: SIP/2.0/TCP 192.168.1.103:60201;branch=z9hG4bK06d9652645c4a5359cb6504c2cd5cbc9383037;rport
D/SipHelper( 286): Max-Forwards: 70
D/SipHelper( 286): Contact: "MYREALNAMEWASHERE" <sip:MYPBXESORGACCOUNT@192.168.1.103:60201;transport=tcp>
D/SipHelper( 286): Content-Type: application/sdp
D/SipHelper( 286): Content-Length: 299
D/SipHelper( 286):
D/SipHelper( 286): v=0
D/SipHelper( 286): o=- 1295392691601 1295392691606 IN IP4 192.168.1.103
D/SipHelper( 286): s=-
D/SipHelper( 286): c=IN IP4 192.168.1.103
D/SipHelper( 286): t=0 0
D/SipHelper( 286): m=audio 59270 RTP/AVP 96 97 3 0 8 127
D/SipHelper( 286): a=rtpmap:96 GSM-EFR/8000
D/SipHelper( 286): a=rtpmap:97 AMR/8000
D/SipHelper( 286): a=rtpmap:3 GSM/8000
D/SipHelper( 286): a=rtpmap:0 PCMU/8000
D/SipHelper( 286): a=rtpmap:8 PCMA/8000
D/SipHelper( 286): a=rtpmap:127 telephone-event/8000
D/SipHelper( 286): a=fmtp:127 0-15
I/power ( 135): *** set_screen_state 1
D/SipSession( 286): +++ add a session with key: '84e24779505aa6608ce9be97b509ce57@192.168.1.103'
D/SipSession( 286): 84e24779505aa6608ce9be97b509ce57@192.168.1.103: @4062ea68:OUTGOING_CALL
D/SipAudioCall( 286): calling... android.net.sip.SipSession@4062ead8
D/SipSession( 286): session key from event: 84e24779505aa6608ce9be97b509ce57@192.168.1.103
D/SipSession( 286): active sessions:
D/SipSession( 286): ...84e24779505aa6608ce9be97b509ce57@192.168.1.103: @4062ea68:OUTGOING_CALL
D/SipSession( 286): ~~~~~ @4062ea68:OUTGOING_CALL: OUTGOING_CALL: processing SIP/2.0 407 Proxy Authentication Required
D/SipSession( 286): Via: SIP/2.0/TCP 192.168.1.103:60201;branch=z9hG4bK06d9652645c4a5359cb6504c2cd5cbc9383037
D/SipSession( 286): From: "MYREALNAMEWASHERE" <sip:MYPBXESORGACCOUNT@pbxes.org:5060>;tag=2667853903
D/SipSession( 286): To: <sip:MYREALPHONENUMBERWASHERE@pbxes.org>;tag=as7cf9275b
D/SipSession( 286): Call-ID: 84e24779505aa6608ce9be97b509ce57@192.168.1.103
D/SipSession( 286): CSeq: 6901 INVITE
D/SipSession( 286): User-Agent: PBX
D/SipSession( 286): Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
D/SipSession( 286): Contact: <sip:MYREALPHONENUMBERWASHERE@188.40.65.170:5060>
D/SipSession( 286): Proxy-Authenticate: Digest realm="pbxes.org",nonce="0635de64099b464638bc83241f0303eb58f91151"
D/SipSession( 286): Content-Length: 0
D/SipSession( 286):
D/SipHelper( 286): send request with challenge response: INVITE sip:MYREALPHONENUMBERWASHERE@pbxes.org:5060;maddr=pbxes.org SIP/2.0
D/SipHelper( 286): Call-ID: 84e24779505aa6608ce9be97b509ce57@192.168.1.103
D/SipHelper( 286): CSeq: 6902 INVITE
D/SipHelper( 286): From: "MYREALNAMEWASHERE" <sip:MYPBXESORGACCOUNT@pbxes.org:5060>;tag=2667853903
D/SipHelper( 286): To: <sip:MYREALPHONENUMBERWASHERE@pbxes.org>
D/SipHelper( 286): Via: SIP/2.0/TCP 192.168.1.103:60201;branch=z9hG4bKd636f7a1531c0a18e342877c448d57a4383037;rport
D/SipHelper( 286): Max-Forwards: 70
D/SipHelper( 286): Contact: "MYREALNAMEWASHERE" <sip:MYPBXESORGACCOUNT@192.168.1.103:60201;transport=tcp>
D/SipHelper( 286): Content-Type: application/sdp
D/SipHelper( 286): Proxy-Authorization: Digest username="MYPBXESORGACCOUNT",realm="pbxes.org",nonce="0635de64099b464638bc83241f0303eb58f91151",uri="sip:MYREALPHONENUMBERWASHERE@pbxes.org:5060;maddr=pbxes.org",response="1136d3d04769ab42af41e4aee25306c9"
D/SipHelper( 286): Content-Length: 299
D/SipHelper( 286):
D/SipHelper( 286): v=0
D/SipHelper( 286): o=- 1295392691601 1295392691606 IN IP4 192.168.1.103
D/SipHelper( 286): s=-
D/SipHelper( 286): c=IN IP4 192.168.1.103
D/SipHelper( 286): t=0 0
D/SipHelper( 286): m=audio 59270 RTP/AVP 96 97 3 0 8 127
D/SipHelper( 286): a=rtpmap:96 GSM-EFR/8000
D/SipHelper( 286): a=rtpmap:97 AMR/8000
D/SipHelper( 286): a=rtpmap:3 GSM/8000
D/SipHelper( 286): a=rtpmap:0 PCMU/8000
D/SipHelper( 286): a=rtpmap:8 PCMA/8000
D/SipHelper( 286): a=rtpmap:127 telephone-event/8000
D/SipHelper( 286): a=fmtp:127 0-15
D/SipSession( 286): authentication retry count=1
D/SipSession( 286): remove session @4062ea68:OUTGOING_CALL @key '84e24779505aa6608ce9be97b509ce57@192.168.1.103'
D/SipSession( 286): +++ add a session with key: '84e24779505aa6608ce9be97b509ce57@192.168.1.103'
D/SipSession( 286): 84e24779505aa6608ce9be97b509ce57@192.168.1.103: @4062ea68:OUTGOING_CALL
D/SipSession( 286): new state after: OUTGOING_CALL
D/SipSession( 286): session key from event: 84e24779505aa6608ce9be97b509ce57@192.168.1.103
D/SipSession( 286): active sessions:
D/SipSession( 286): ...84e24779505aa6608ce9be97b509ce57@192.168.1.103: @4062ea68:OUTGOING_CALL
D/SipSession( 286): not the current transaction; current=req=INVITE,6902,s=CALLING,ds=null,, target=req=INVITE,6901,s=TERMINATED,ds=TERMINATED,
D/dalvikvm( 286): GC_CONCURRENT freed 1477K, 51% free 4449K/9031K, external 2927K/3611K, paused 19ms+13ms
D/SipSession( 286): session key from event: 84e24779505aa6608ce9be97b509ce57@192.168.1.103
D/SipSession( 286): active sessions:
D/SipSession( 286): ...84e24779505aa6608ce9be97b509ce57@192.168.1.103: @4062ea68:OUTGOING_CALL
D/SipSession( 286): ~~~~~ @4062ea68:OUTGOING_CALL: OUTGOING_CALL: processing SIP/2.0 100 Trying
D/SipSession( 286): Via: SIP/2.0/TCP 192.168.1.103:60201;branch=z9hG4bKd636f7a1531c0a18e342877c448d57a4383037
D/SipSession( 286): From: "MYREALNAMEWASHERE" <sip:MYPBXESORGACCOUNT@pbxes.org:5060>;tag=2667853903
D/SipSession( 286): To: <sip:MYREALPHONENUMBERWASHERE@pbxes.org>
D/SipSession( 286): Call-ID: 84e24779505aa6608ce9be97b509ce57@192.168.1.103
D/SipSession( 286): CSeq: 6902 INVITE
D/SipSession( 286): User-Agent: PBX
D/SipSession( 286): Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
D/SipSession( 286): Contact: <sip:MYREALPHONENUMBERWASHERE@188.40.65.170:5060>
D/SipSession( 286): Content-Length: 0
D/SipSession( 286):
D/SipSession( 286): new state after: OUTGOING_CALL
D/SipSession( 286): session key from event: 84e24779505aa6608ce9be97b509ce57@192.168.1.103
D/SipSession( 286): active sessions:
D/SipSession( 286): ...84e24779505aa6608ce9be97b509ce57@192.168.1.103: @4062ea68:OUTGOING_CALL
D/SipSession( 286): ~~~~~ @4062ea68:OUTGOING_CALL: OUTGOING_CALL: processing SIP/2.0 183 Session Progress
D/SipSession( 286): Via: SIP/2.0/TCP 192.168.1.103:60201;branch=z9hG4bKd636f7a1531c0a18e342877c448d57a4383037
D/SipSession( 286): From: "MYREALNAMEWASHERE" <sip:MYPBXESORGACCOUNT@pbxes.org:5060>;tag=2667853903
D/SipSession( 286): To: <sip:MYREALPHONENUMBERWASHERE@pbxes.org>;tag=as2111e100
D/SipSession( 286): Call-ID: 84e24779505aa6608ce9be97b509ce57@192.168.1.103
D/SipSession( 286): CSeq: 6902 INVITE
D/SipSession( 286): User-Agent: PBX
D/SipSession( 286): Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
D/SipSession( 286): Contact: <sip:MYREALPHONENUMBERWASHERE@188.40.65.170:5060>
D/SipSession( 286): Content-Type: application/sdp
D/SipSession( 286): Content-Length: 236
D/SipSession( 286):
D/SipSession( 286): v=0
D/SipSession( 286): o=root 5761 5761 IN IP4 76.191.104.53
D/SipSession( 286): s=session
D/SipSession( 286): c=IN IP4 76.191.104.53
D/SipSession( 286): t=0 0
D/SipSession( 286): m=audio 38342 RTP/AVP 0 8 3 127
D/SipSession( 286): a=rtpmap:0 PCMU/8000
D/SipSession( 286): a=rtpmap:8 PCMA/8000
D/SipSession( 286): a=rtpmap:3 GSM/8000
D/SipSession( 286): a=rtpmap:127 telephone-event/8000
D/SipSession( 286): a=fmtp:127 0-16
D/SipSession( 286): new state after: OUTGOING_CALL_RING_BACK
D/SipAudioCall( 286): sip call ringing back: android.net.sip.SipSession@4062ead8
V/Call ( 286): +***+ call state changed: DIALING --> ALERTING: DIALING:com.android.internal.telephony.sip.SipPhone$SipCall@407e3308: on phone Handler{40812ac8} 1
V/SipPhone( 286): +***+ connection state changed: sip:MYREALPHONENUMBERWASHERE@pbxes.org: ALERTING on phone Handler{40812ac8}
I/AudioHardwareMSM72XX( 101): Routing audio to Handset
D/HTC Acoustic( 101): msm72xx_enable_audpp: 0x0000
W/SipSession( 286): peer did not respond rport
W/SipSession( 286): peer did not respond rport
D/dalvikvm( 322): GC_EXPLICIT freed 829K, 50% free 3605K/7111K, external 2826K/3450K, paused 152ms
D/SipSession( 286): session key from event: 84e24779505aa6608ce9be97b509ce57@192.168.1.103
D/SipSession( 286): active sessions:
D/SipSession( 286): ...84e24779505aa6608ce9be97b509ce57@192.168.1.103: @4062ea68:OUTGOING_CALL_RING_BACK
D/SipSession( 286): not the current dialog; current=gov.nist.javax.sip.stack.SIPDialog@407dd260, terminated=gov.nist.javax.sip.stack.SIPDialog@40770418
D/SipSession( 286): session key from event: 84e24779505aa6608ce9be97b509ce57@192.168.1.103
D/SipSession( 286): active sessions:
D/SipSession( 286): ...84e24779505aa6608ce9be97b509ce57@192.168.1.103: @4062ea68:OUTGOING_CALL_RING_BACK
D/SipSession( 286): ~~~~~ @4062ea68:OUTGOING_CALL_RING_BACK: OUTGOING_CALL_RING_BACK: processing SIP/2.0 200 OK
D/SipSession( 286): Via: SIP/2.0/TCP 192.168.1.103:60201;branch=z9hG4bKd636f7a1531c0a18e342877c448d57a4383037
D/SipSession( 286): From: "MYREALNAMEWASHERE" <sip:MYPBXESORGACCOUNT@pbxes.org:5060>;tag=2667853903
D/SipSession( 286): To: <sip:MYREALPHONENUMBERWASHERE@pbxes.org>;tag=as2111e100
D/SipSession( 286): Call-ID: 84e24779505aa6608ce9be97b509ce57@192.168.1.103
D/SipSession( 286): CSeq: 6902 INVITE
D/SipSession( 286): User-Agent: PBX
D/SipSession( 286): Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
D/SipSession( 286): Contact: <sip:MYREALPHONENUMBERWASHERE@188.40.65.170:5060>
D/SipSession( 286): Content-Type: application/sdp
D/SipSession( 286): Content-Length: 236
D/SipSession( 286):
D/SipSession( 286): v=0
D/SipSession( 286): o=root 5761 5762 IN IP4 76.191.104.53
D/SipSession( 286): s=session
D/SipSession( 286): c=IN IP4 76.191.104.53
D/SipSession( 286): t=0 0
D/SipSession( 286): m=audio 38342 RTP/AVP 0 8 3 127
D/SipSession( 286): a=rtpmap:0 PCMU/8000
D/SipSession( 286): a=rtpmap:8 PCMA/8000
D/SipSession( 286): a=rtpmap:3 GSM/8000
D/SipSession( 286): a=rtpmap:127 telephone-event/8000
D/SipSession( 286): a=fmtp:127 0-16
W/System.err( 286): Unexpected internal error FIXME!! Cannot create ACK - no ListeningPoint for transport towards next hop found:UDP
W/System.err( 286): javax.sip.SipException: Cannot create ACK - no ListeningPoint for transport towards next hop found:UDP
W/System.err( 286):	at gov.nist.javax.sip.stack.SIPDialog.createAck(SIPDialog.java:2443)
W/System.err( 286):	at com.android.server.sip.SipHelper.sendInviteAck(SipHelper.java:389)
W/System.err( 286):	at com.android.server.sip.SipSessionGroup$SipSessionImpl.outgoingCall(SipSessionGroup.java:1021)
W/System.err( 286):	at com.android.server.sip.SipSessionGroup$SipSessionImpl.process(SipSessionGroup.java:648)
W/System.err( 286):	at com.android.server.sip.SipSessionGroup.process(SipSessionGroup.java:324)
W/System.err( 286):	at com.android.server.sip.SipSessionGroup.processResponse(SipSessionGroup.java:301)
W/System.err( 286):	at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:291)
W/System.err( 286):	at gov.nist.javax.sip.EventScanner.run(EventScanner.java:511)
W/System.err( 286):	at java.lang.Thread.run(Thread.java:1019)
W/SipSession( 286): event process error: gov.nist.javax.sip.ResponseEventExt[source=gov.nist.javax.sip.SipProviderImpl@4065e660]
W/SipSession( 286): java.lang.RuntimeException: Unexpected internal error FIXME!! Cannot create ACK - no ListeningPoint for transport towards next hop found:UDP
W/SipSession( 286):	at gov.nist.core.InternalErrorHandler.handleException(InternalErrorHandler.java:45)
W/SipSession( 286):	at gov.nist.javax.sip.stack.SIPDialog.createAck(SIPDialog.java:2495)
W/SipSession( 286):	at com.android.server.sip.SipHelper.sendInviteAck(SipHelper.java:389)
W/SipSession( 286):	at com.android.server.sip.SipSessionGroup$SipSessionImpl.outgoingCall(SipSessionGroup.java:1021)
W/SipSession( 286):	at com.android.server.sip.SipSessionGroup$SipSessionImpl.process(SipSessionGroup.java:648)
W/SipSession( 286):	at com.android.server.sip.SipSessionGroup.process(SipSessionGroup.java:324)
W/SipSession( 286):	at com.android.server.sip.SipSessionGroup.processResponse(SipSessionGroup.java:301)
W/SipSession( 286):	at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:291)
W/SipSession( 286):	at gov.nist.javax.sip.EventScanner.run(EventScanner.java:511)
W/SipSession( 286):	at java.lang.Thread.run(Thread.java:1019)
W/SipSession( 286): Caused by: javax.sip.SipException: Cannot create ACK - no ListeningPoint for transport towards next hop found:UDP
W/SipSession( 286):	at gov.nist.javax.sip.stack.SIPDialog.createAck(SIPDialog.java:2443)
W/SipSession( 286):	... 8 more
D/SipSession( 286): remove session @4062ea68:OUTGOING_CALL_RING_BACK @key '84e24779505aa6608ce9be97b509ce57@192.168.1.103'
D/SipAudioCall( 286): sip session error: CLIENT_ERROR: javax.sip.SipException: Cannot create ACK - no ListeningPoint for transport towards next hop found:UDP
W/SipPhone( 286): error: CLIENT_ERROR: javax.sip.SipException: Cannot create ACK - no ListeningPoint for transport towards next hop found:UDP
D/SipPhone( 286): SIP error: ERROR_UNSPECIFIED
D/SipPhone( 286): --- connection ended: sip:MYREALPHONENUMBERWASHERE@pbxes.org: 0, cause: ERROR_UNSPECIFIED, on phone Handler{40812ac8}
D/SipAudioCall( 286): stop audiocall
D/Call ( 286): ---check connections: 1
D/Call ( 286): state=DISCONNECTED: incoming: false state: DISCONNECTED post dial state: NOT_STARTED
V/Call ( 286): +***+ call state changed: ALERTING --> DISCONNECTED: ALERTING:com.android.internal.telephony.sip.SipPhone$SipCall@407e3308: on phone Handler{40812ac8} 1
D/SipPhone( 286): ^^^ new phone state: IDLE
I/AudioService( 135): AudioFocus abandonAudioFocus() from AudioFocus_For_Phone_Ring_And_Calls
D/SipAudioCall( 286): stop audiocall
I/PowerManagerService( 135): clearUserActivity for 10000ms from now
W/SipSession( 286): peer did not respond rport
D/SipSession( 286): ~~~~~ @4086a398:READY_TO_CALL: READY_TO_CALL: processing SIP/2.0 200 OK
D/SipSession( 286): Via: SIP/2.0/TCP 192.168.1.103:60201;branch=z9hG4bKd636f7a1531c0a18e342877c448d57a4383037
D/SipSession( 286): From: "MYREALNAMEWASHERE" <sip:MYPBXESORGACCOUNT@pbxes.org:5060>;tag=2667853903
D/SipSession( 286): To: <sip:MYREALPHONENUMBERWASHERE@pbxes.org>;tag=as2111e100
D/SipSession( 286): Call-ID: 84e24779505aa6608ce9be97b509ce57@192.168.1.103
D/SipSession( 286): CSeq: 6902 INVITE
D/SipSession( 286): User-Agent: PBX
D/SipSession( 286): Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
D/SipSession( 286): Contact: <sip:MYREALPHONENUMBERWASHERE@188.40.65.170:5060>
D/SipSession( 286): Content-Type: application/sdp
D/SipSession( 286): Content-Length: 236
D/SipSession( 286):
D/SipSession( 286): v=0
D/SipSession( 286): o=root 5761 5762 IN IP4 76.191.104.53
D/SipSession( 286): s=session
D/SipSession( 286): c=IN IP4 76.191.104.53
D/SipSession( 286): t=0 0
D/SipSession( 286): m=audio 38342 RTP/AVP 0 8 3 127
D/SipSession( 286): a=rtpmap:0 PCMU/8000
D/SipSession( 286): a=rtpmap:8 PCMA/8000
D/SipSession( 286): a=rtpmap:3 GSM/8000
D/SipSession( 286): a=rtpmap:127 telephone-event/8000
D/SipSession( 286): a=fmtp:127 0-16
W/SipSession( 286): peer did not respond rport
V/Call ( 286): +***+ call state changed: DISCONNECTED --> IDLE: DISCONNECTED:com.android.internal.telephony.sip.SipPhone$SipCall@407e3308: on phone Handler{40812ac8} 0
I/ActivityManager( 135): Starting: Intent { act=android.intent.action.VIEW typ=vnd.android.cursor.dir/calls flg=0x10000 cmp=com.android.contacts/.DialtactsActivity } from pid 286
I/ActivityManager( 135): moveTaskToBack: 9
D/dalvikvm( 135): GC_CONCURRENT freed 772K, 47% free 4300K/8007K, external 1500K/2012K, paused 19ms+34ms
I/ActivityManager( 135): Displayed com.android.contacts/.DialtactsActivity: +968ms
D/SipSession( 286): ~~~~~ @4086a398:READY_TO_CALL: READY_TO_CALL: processing SIP/2.0 200 OK
D/SipSession( 286): Via: SIP/2.0/TCP 192.168.1.103:60201;branch=z9hG4bKd636f7a1531c0a18e342877c448d57a4383037
D/SipSession( 286): From: "MYREALNAMEWASHERE" <sip:MYPBXESORGACCOUNT@pbxes.org:5060>;tag=2667853903
D/SipSession( 286): To: <sip:MYREALPHONENUMBERWASHERE@pbxes.org>;tag=as2111e100
D/SipSession( 286): Call-ID: 84e24779505aa6608ce9be97b509ce57@192.168.1.103
D/SipSession( 286): CSeq: 6902 INVITE
D/SipSession( 286): User-Agent: PBX
D/SipSession( 286): Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
D/SipSession( 286): Contact: <sip:MYREALPHONENUMBERWASHERE@188.40.65.170:5060>
D/SipSession( 286): Content-Type: application/sdp
D/SipSession( 286): Content-Length: 236
D/SipSession( 286):
D/SipSession( 286): v=0
D/SipSession( 286): o=root 5761 5762 IN IP4 76.191.104.53
D/SipSession( 286): s=session
D/SipSession( 286): c=IN IP4 76.191.104.53
D/SipSession( 286): t=0 0
D/SipSession( 286): m=audio 38342 RTP/AVP 0 8 3 127
D/SipSession( 286): a=rtpmap:0 PCMU/8000
D/SipSession( 286): a=rtpmap:8 PCMA/8000
D/SipSession( 286): a=rtpmap:3 GSM/8000
D/SipSession( 286): a=rtpmap:127 telephone-event/8000
D/SipSession( 286): a=fmtp:127 0-16
D/SipSession( 286): ~~~~~ @4086a398:READY_TO_CALL: READY_TO_CALL: processing SIP/2.0 200 OK
D/SipSession( 286): Via: SIP/2.0/TCP 192.168.1.103:60201;branch=z9hG4bKd636f7a1531c0a18e342877c448d57a4383037
D/SipSession( 286): From: "MYREALNAMEWASHERE" <sip:MYPBXESORGACCOUNT@pbxes.org:5060>;tag=2667853903
D/SipSession( 286): To: <sip:MYREALPHONENUMBERWASHERE@pbxes.org>;tag=as2111e100
D/SipSession( 286): Call-ID: 84e24779505aa6608ce9be97b509ce57@192.168.1.103
D/SipSession( 286): CSeq: 6902 INVITE
D/SipSession( 286): User-Agent: PBX
D/SipSession( 286): Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
D/SipSession( 286): Contact: <sip:MYREALPHONENUMBERWASHERE@188.40.65.170:5060>
D/SipSession( 286): Content-Type: application/sdp
D/SipSession( 286): Content-Length: 236
D/SipSession( 286):
D/SipSession( 286): v=0
D/SipSession( 286): o=root 5761 5762 IN IP4 76.191.104.53
D/SipSession( 286): s=session
D/SipSession( 286): c=IN IP4 76.191.104.53
D/SipSession( 286): t=0 0
D/SipSession( 286): m=audio 38342 RTP/AVP 0 8 3 127
D/SipSession( 286): a=rtpmap:0 PCMU/8000
D/SipSession( 286): a=rtpmap:8 PCMA/8000
D/SipSession( 286): a=rtpmap:3 GSM/8000
D/SipSession( 286): a=rtpmap:127 telephone-event/8000
D/SipSession( 286): a=fmtp:127 0-16
D/dalvikvm( 286): GC_CONCURRENT freed 1590K, 52% free 4419K/9031K, external 2927K/3611K, paused 10ms+10ms