Here is the log : 07-27 22:07:46.670: VERBOSE/SipAudioCall(15522): onCallEstablished()v=0 07-27 22:07:46.670: VERBOSE/SipAudioCall(15522): o=cp10 131179726711 131179726712 IN IP4 172.18.24.29 07-27 22:07:46.670: VERBOSE/SipAudioCall(15522): s=SIP Call 07-27 22:07:46.670: VERBOSE/SipAudioCall(15522): c=IN IP4 212.27.52.130 07-27 22:07:46.670: VERBOSE/SipAudioCall(15522): t=0 0 07-27 22:07:46.670: VERBOSE/SipAudioCall(15522): m=audio 37588 RTP/AVP 8 07-27 22:07:46.670: VERBOSE/SipAudioCall(15522): b=AS:75 07-27 22:07:46.670: VERBOSE/SipAudioCall(15522): a=rtpmap:8 PCMA/ 8000/1 07-27 22:07:46.670: VERBOSE/SipAudioCall(15522): a=ptime:30 07-27 22:07:46.670: VERBOSE/SipAudioCall(15522): a=sendrecv 07-27 22:07:46.670: DEBUG/SipAudioCall(15522): stop audiocall 07-27 22:07:46.670: VERBOSE/SipAudioCall(15522): acquire wifi high perf lock 07-27 22:07:46.670: DEBUG/AudioGroup(15522): stream[85] is configured as PCMA 8kHz 20ms mode 0 07-27 22:07:46.670: DEBUG/AudioGroup(15522): stream[90] is configured as RAW 8kHz 32ms mode 0 07-27 22:07:46.670: DEBUG/AudioGroup(15522): stream[90] joins group[89] 07-27 22:07:46.670: DEBUG/AudioGroup(15522): stream[85] joins group[89] 07-27 22:07:46.670: DEBUG/AudioGroup(15522): group[89] switches from mode 0 to 2 07-27 22:07:46.670: DEBUG/AudioGroup(15522): reported frame count: output 743, input 320 07-27 22:07:46.670: DEBUG/AudioGroup(15522): adjusted frame count: output 743, input 512 07-27 22:07:46.680: DEBUG/AudioGroup(15522): latency: output 184, input 64 07-27 22:07:46.730: DEBUG/SipSession(220): session key from event: 26fdbc8fd021583b836219087c4926af@192.168.0.50 07-27 22:07:46.730: DEBUG/SipSession(220): active sessions: 07-27 22:07:46.730: DEBUG/SipSession(220): ... 26fdbc8fd021583b836219087c4926af@192.168.0.50: @4177a5f0:IN_CALL 07-27 22:07:46.730: DEBUG/SipSession(220): transaction terminated: req=INVITE,3141,s=TERMINATED,ds=CONFIRMED, 07-27 22:07:46.730: DEBUG/SipSession(220): Transaction terminated; do nothing 07-27 22:07:49.310: DEBUG/dalvikvm(205): GC_FOR_ALLOC freed 394K, 14% free 10284K/11847K, paused 42ms 07-27 22:07:49.310: INFO/dalvikvm-heap(205): Grow heap (frag case) to 10.604MB for 513744-byte allocation 07-27 22:07:49.350: DEBUG/dalvikvm(205): GC_FOR_ALLOC freed 44K, 14% free 10741K/12359K, paused 34ms 07-27 22:07:49.510: DEBUG/dalvikvm(205): GC_FOR_ALLOC freed 2479K, 32% free 8478K/12359K, paused 50ms 07-27 22:07:49.510: INFO/dalvikvm-heap(205): Grow heap (frag case) to 9.218MB for 908816-byte allocation 07-27 22:07:49.580: DEBUG/dalvikvm(205): GC_CONCURRENT freed 89K, 25% free 9277K/12359K, paused 2ms+4ms 07-27 22:07:49.840: DEBUG/PhoneWindow(221): couldn't save which view has focus because the focused view android.widget.TextView@40872250 has no id. 07-27 22:07:50.030: DEBUG/dalvikvm(15522): GC_CONCURRENT freed 3502K, 30% free 9915K/14023K, paused 3ms+6ms 07-27 22:07:50.780: DEBUG/dalvikvm(220): GC_FOR_ALLOC freed 1566K, 10% free 21310K/23623K, paused 200ms 07-27 22:07:54.210: DEBUG/SipSession(220): session key from event: 26fdbc8fd021583b836219087c4926af@192.168.0.50 07-27 22:07:54.210: DEBUG/SipSession(220): active sessions: 07-27 22:07:54.210: DEBUG/SipSession(220): ... 26fdbc8fd021583b836219087c4926af@192.168.0.50: @4177a5f0:IN_CALL 07-27 22:07:54.210: DEBUG/SipSession(220): not the current dialog; current=gov.nist.javax.sip.stack.SIPDialog@416a3f70, terminated=gov.nist.javax.sip.stack.SIPDialog@40efc128 07-27 22:07:55.330: VERBOSE/SipAudioCall(15522): send DTMF: 2 07-27 22:08:01.900: WARN/ProcessStats(141): Skipping unknown process pid 15920 07-27 22:08:02.650: DEBUG/dalvikvm(220): GC_FOR_ALLOC freed 761K, 10% free 21292K/23623K, paused 181ms 07-27 22:08:05.060: DEBUG/dalvikvm(282): GC_EXPLICIT freed 343K, 11% free 6931K/7751K, paused 5ms+2ms
On 16 juil, 13:44, Robert Auger <bobyg...@gmail.com> wrote: > One more information : if I call : > > (TelephonyManager) > getActivity().getSystemService(Context.TELEPHONY_SERVICE).getPhoneType() > > in the "onCallEstablished" callback of the "SipAudioCall.Listener" > included in the "makeAudioCall" (at this time I can perfectly listen > to my audio messages), the answer is : > > 0 : value of "TelephonyManager.PHONE_TYPE_NONE" > > I was expecting : > > 3 : value of "TelephonyManager.PHONE_TYPE_SIP" > > The SIP API is maybe not fully integrated on Android 3.0 ? > > On 5 juil, 21:37, Robert Auger <bobyg...@gmail.com> wrote: > > > > > > > > > Hello, > > > Does the « sendDtmf » method from « SipAudioCall » class really work > > on Android 3.0 / MotorolaXoomWiFi ? > > > I am developping a SIP activated application for Android 3.0 tablets > > and testing it on MotorolaXoomWiFi(no 3G nor 4G) > > > I am able to : > > - create a « SipManager » with « SipManager.newInstance() » > > - use « manageurSip.makeAudioCall() » to retrieve my voicemail in my > > SIP provider account > > - in the « onCallEstablished » callback, I can use « startAudio() » > > and « setSpeakerMode(true) », to hear messages > > > But when I try to use « sendDtmf(int) » to save or delete my messages, > > nothing happens. > > > If I try to use an already developped SIP application "CSIPSimple", I > > am also unable to send DTMF tones. > > > Should I wait for Android 3.1 to use this feature ? > > > Thank you in advance. -- You received this message because you are subscribed to the Google Groups "Android Developers" group. To post to this group, send email to android-developers@googlegroups.com To unsubscribe from this group, send email to android-developers+unsubscr...@googlegroups.com For more options, visit this group at http://groups.google.com/group/android-developers?hl=en
