[android-developers] Re: sendDtmf on Xoom WiFI / Android 3.0

2011-07-30 Thread Robert Auger
I think we miss the methods getAudioGroup() and getAudioStream()
in the SipAudioCall class

On 28 juil, 21:04, Robert Auger bobyg...@gmail.com wrote:
 Here is the log I get when I am registering :

 07-28 20:55:57.980: DEBUG/dalvikvm(3853): GC_CONCURRENT freed 460K, 8%
 free 6788K/7367K, paused 2ms+2ms
 07-28 20:55:58.340: DEBUG/SipSession(217): +++  add a session with
 key:  'a0aaa95da53e5c8b00d1dbc183bc1...@192.168.0.50'
 07-28 20:55:58.340: DEBUG/SipSession(217):
 a0aaa95da53e5c8b00d1dbc183bc1...@192.168.0.50: @408aa5b0:DEREGISTERING
 07-28 20:55:58.370: DEBUG/dalvikvm(217): GC_CONCURRENT freed 351K, 6%
 free 6826K/7239K, paused 2ms+2ms
 07-28 20:55:58.380: DEBUG/SipSession(217): session key from event:
 a0aaa95da53e5c8b00d1dbc183bc1...@192.168.0.50
 07-28 20:55:58.380: DEBUG/SipSession(217): active sessions:
 07-28 20:55:58.380: DEBUG/
 SipSession(217):  ...a0aaa95da53e5c8b00d1dbc183bc1...@192.168.0.50:
 @408aa5b0:DEREGISTERING
 07-28 20:55:58.380: DEBUG/SipSession(217):  ~
 @408aa5b0:DEREGISTERING: DEREGISTERING: processing SIP/2.0 100 Trying
 07-28 20:55:58.380: DEBUG/SipSession(217): Call-ID:
 a0aaa95da53e5c8b00d1dbc183bc1...@192.168.0.50
 07-28 20:55:58.380: DEBUG/SipSession(217): CSeq: 7833 REGISTER
 07-28 20:55:58.380: DEBUG/SipSession(217): From: Freebox
 sip:xxx...@freephonie.net;tag=1576196663
 07-28 20:55:58.380: DEBUG/SipSession(217): To: Freebox
 sip:xxx...@freephonie.net
 07-28 20:55:58.380: DEBUG/SipSession(217): Via: SIP/2.0/UDP
 192.168.0.50:47118;received=88.189.161.218;rport=47118;branch=z9hG4bKc032fa 
 a67c86615d912bd26401267d58343539
 07-28 20:55:58.380: DEBUG/SipSession(217): Content-Length: 0
 07-28 20:55:58.380: DEBUG/SipSession(217):
 07-28 20:55:58.390: DEBUG/SipSession(217): session key from event:
 a0aaa95da53e5c8b00d1dbc183bc1...@192.168.0.50
 07-28 20:55:58.390: DEBUG/SipSession(217): active sessions:
 07-28 20:55:58.390: DEBUG/
 SipSession(217):  ...a0aaa95da53e5c8b00d1dbc183bc1...@192.168.0.50:
 @408aa5b0:DEREGISTERING
 07-28 20:55:58.390: DEBUG/SipSession(217):  ~
 @408aa5b0:DEREGISTERING: DEREGISTERING: processing SIP/2.0 401
 Unauthorized
 07-28 20:55:58.390: DEBUG/SipSession(217): Call-ID:
 a0aaa95da53e5c8b00d1dbc183bc1...@192.168.0.50
 07-28 20:55:58.390: DEBUG/SipSession(217): CSeq: 7833 REGISTER
 07-28 20:55:58.390: DEBUG/SipSession(217): From: Freebox
 sip:xxx...@freephonie.net;tag=1576196663
 07-28 20:55:58.390: DEBUG/SipSession(217): To: Freebox
 sip:xxx...@freephonie.net;tag=00-08080-0a4df414-3689cf434
 07-28 20:55:58.390: DEBUG/SipSession(217): Via: SIP/2.0/UDP
 192.168.0.50:47118;received=88.189.161.218;rport=47118;branch=z9hG4bKc032fa 
 a67c86615d912bd26401267d58343539
 07-28 20:55:58.390: DEBUG/SipSession(217): WWW-Authenticate: Digest
 realm=freephonie.net,nonce=0a4df35f3d555d9e1855864a78e12cfd,opaque=0a4 
 cb0a41fcfc7b,stale=false,algorithm=MD5
 07-28 20:55:58.390: DEBUG/SipSession(217): Server: Cirpack/v4.42q
 (gw_sip)
 07-28 20:55:58.390: DEBUG/SipSession(217): Content-Length: 0
 07-28 20:55:58.390: DEBUG/SipSession(217):
 07-28 20:55:58.410: DEBUG/SipHelper(217): send request with challenge
 response: REGISTER sip:freephonie.net:5060 SIP/2.0
 07-28 20:55:58.410: DEBUG/SipHelper(217): Call-ID:
 a0aaa95da53e5c8b00d1dbc183bc1...@192.168.0.50
 07-28 20:55:58.410: DEBUG/SipHelper(217): CSeq: 7834 REGISTER
 07-28 20:55:58.410: DEBUG/SipHelper(217): From: Freebox
 sip:xxx...@freephonie.net;tag=1576196663
 07-28 20:55:58.410: DEBUG/SipHelper(217): To: Freebox
 sip:xxx...@freephonie.net
 07-28 20:55:58.410: DEBUG/SipHelper(217): Via: SIP/2.0/UDP
 192.168.0.50:47118;branch=z9hG4bKf35572d19841940b7d72bd2d9ddf8feb343539;rpo rt
 07-28 20:55:58.410: DEBUG/SipHelper(217): Max-Forwards: 70
 07-28 20:55:58.410: DEBUG/SipHelper(217): User-Agent: SIPAUA/0.1.001
 07-28 20:55:58.410: DEBUG/SipHelper(217): Contact: *
 07-28 20:55:58.410: DEBUG/SipHelper(217): Expires: 0
 07-28 20:55:58.410: DEBUG/SipHelper(217): Authorization: Digest
 username=XX,realm=freephonie.net,nonce=0a4df35f3d555d9e1855864 
 a78e12cfd,uri=sip:freephonie.net:
 5060,response=6194c6c5737201ffa281cdf59f7ce074,algorithm=MD5,opaque=0a4 
 cb0a41fcfc7b
 07-28 20:55:58.410: DEBUG/SipHelper(217): Content-Length: 0
 07-28 20:55:58.410: DEBUG/SipHelper(217):
 07-28 20:55:58.420: DEBUG/SipSession(217):    authentication retry
 count=1
 07-28 20:55:58.420: DEBUG/SipSession(217): new state after:
 DEREGISTERING
 07-28 20:55:58.450: DEBUG/dalvikvm(217): GC_CONCURRENT freed 476K, 8%
 free 6892K/7431K, paused 2ms+3ms
 07-28 20:55:58.460: DEBUG/SipSession(217): session key from event:
 a0aaa95da53e5c8b00d1dbc183bc1...@192.168.0.50
 07-28 20:55:58.460: DEBUG/SipSession(217): active sessions:
 07-28 20:55:58.460: DEBUG/
 SipSession(217):  ...a0aaa95da53e5c8b00d1dbc183bc1...@192.168.0.50:
 @408aa5b0:DEREGISTERING
 07-28 20:55:58.460: DEBUG/SipSession(217):  ~
 @408aa5b0:DEREGISTERING: DEREGISTERING: processing SIP/2.0 100 Trying
 07-28 20:55:58.460: DEBUG/SipSession(217): 

[android-developers] Re: sendDtmf on Xoom WiFI / Android 3.0

2011-07-28 Thread Robert Auger
Here is the log I get when I am registering :

07-28 20:55:57.980: DEBUG/dalvikvm(3853): GC_CONCURRENT freed 460K, 8%
free 6788K/7367K, paused 2ms+2ms
07-28 20:55:58.340: DEBUG/SipSession(217): +++  add a session with
key:  'a0aaa95da53e5c8b00d1dbc183bc1a2c@192.168.0.50'
07-28 20:55:58.340: DEBUG/SipSession(217):
a0aaa95da53e5c8b00d1dbc183bc1a2c@192.168.0.50: @408aa5b0:DEREGISTERING
07-28 20:55:58.370: DEBUG/dalvikvm(217): GC_CONCURRENT freed 351K, 6%
free 6826K/7239K, paused 2ms+2ms
07-28 20:55:58.380: DEBUG/SipSession(217): session key from event:
a0aaa95da53e5c8b00d1dbc183bc1a2c@192.168.0.50
07-28 20:55:58.380: DEBUG/SipSession(217): active sessions:
07-28 20:55:58.380: DEBUG/
SipSession(217):  ...a0aaa95da53e5c8b00d1dbc183bc1a2c@192.168.0.50:
@408aa5b0:DEREGISTERING
07-28 20:55:58.380: DEBUG/SipSession(217):  ~
@408aa5b0:DEREGISTERING: DEREGISTERING: processing SIP/2.0 100 Trying
07-28 20:55:58.380: DEBUG/SipSession(217): Call-ID:
a0aaa95da53e5c8b00d1dbc183bc1a2c@192.168.0.50
07-28 20:55:58.380: DEBUG/SipSession(217): CSeq: 7833 REGISTER
07-28 20:55:58.380: DEBUG/SipSession(217): From: Freebox
sip:xxx...@freephonie.net;tag=1576196663
07-28 20:55:58.380: DEBUG/SipSession(217): To: Freebox
sip:xxx...@freephonie.net
07-28 20:55:58.380: DEBUG/SipSession(217): Via: SIP/2.0/UDP
192.168.0.50:47118;received=88.189.161.218;rport=47118;branch=z9hG4bKc032faa67c86615d912bd26401267d58343539
07-28 20:55:58.380: DEBUG/SipSession(217): Content-Length: 0
07-28 20:55:58.380: DEBUG/SipSession(217):
07-28 20:55:58.390: DEBUG/SipSession(217): session key from event:
a0aaa95da53e5c8b00d1dbc183bc1a2c@192.168.0.50
07-28 20:55:58.390: DEBUG/SipSession(217): active sessions:
07-28 20:55:58.390: DEBUG/
SipSession(217):  ...a0aaa95da53e5c8b00d1dbc183bc1a2c@192.168.0.50:
@408aa5b0:DEREGISTERING
07-28 20:55:58.390: DEBUG/SipSession(217):  ~
@408aa5b0:DEREGISTERING: DEREGISTERING: processing SIP/2.0 401
Unauthorized
07-28 20:55:58.390: DEBUG/SipSession(217): Call-ID:
a0aaa95da53e5c8b00d1dbc183bc1a2c@192.168.0.50
07-28 20:55:58.390: DEBUG/SipSession(217): CSeq: 7833 REGISTER
07-28 20:55:58.390: DEBUG/SipSession(217): From: Freebox
sip:xxx...@freephonie.net;tag=1576196663
07-28 20:55:58.390: DEBUG/SipSession(217): To: Freebox
sip:xxx...@freephonie.net;tag=00-08080-0a4df414-3689cf434
07-28 20:55:58.390: DEBUG/SipSession(217): Via: SIP/2.0/UDP
192.168.0.50:47118;received=88.189.161.218;rport=47118;branch=z9hG4bKc032faa67c86615d912bd26401267d58343539
07-28 20:55:58.390: DEBUG/SipSession(217): WWW-Authenticate: Digest
realm=freephonie.net,nonce=0a4df35f3d555d9e1855864a78e12cfd,opaque=0a4cb0a41fcfc7b,stale=false,algorithm=MD5
07-28 20:55:58.390: DEBUG/SipSession(217): Server: Cirpack/v4.42q
(gw_sip)
07-28 20:55:58.390: DEBUG/SipSession(217): Content-Length: 0
07-28 20:55:58.390: DEBUG/SipSession(217):
07-28 20:55:58.410: DEBUG/SipHelper(217): send request with challenge
response: REGISTER sip:freephonie.net:5060 SIP/2.0
07-28 20:55:58.410: DEBUG/SipHelper(217): Call-ID:
a0aaa95da53e5c8b00d1dbc183bc1a2c@192.168.0.50
07-28 20:55:58.410: DEBUG/SipHelper(217): CSeq: 7834 REGISTER
07-28 20:55:58.410: DEBUG/SipHelper(217): From: Freebox
sip:xxx...@freephonie.net;tag=1576196663
07-28 20:55:58.410: DEBUG/SipHelper(217): To: Freebox
sip:xxx...@freephonie.net
07-28 20:55:58.410: DEBUG/SipHelper(217): Via: SIP/2.0/UDP
192.168.0.50:47118;branch=z9hG4bKf35572d19841940b7d72bd2d9ddf8feb343539;rport
07-28 20:55:58.410: DEBUG/SipHelper(217): Max-Forwards: 70
07-28 20:55:58.410: DEBUG/SipHelper(217): User-Agent: SIPAUA/0.1.001
07-28 20:55:58.410: DEBUG/SipHelper(217): Contact: *
07-28 20:55:58.410: DEBUG/SipHelper(217): Expires: 0
07-28 20:55:58.410: DEBUG/SipHelper(217): Authorization: Digest
username=XX,realm=freephonie.net,nonce=0a4df35f3d555d9e1855864a78e12cfd,uri=sip:freephonie.net:
5060,response=6194c6c5737201ffa281cdf59f7ce074,algorithm=MD5,opaque=0a4cb0a41fcfc7b
07-28 20:55:58.410: DEBUG/SipHelper(217): Content-Length: 0
07-28 20:55:58.410: DEBUG/SipHelper(217):
07-28 20:55:58.420: DEBUG/SipSession(217):authentication retry
count=1
07-28 20:55:58.420: DEBUG/SipSession(217): new state after:
DEREGISTERING
07-28 20:55:58.450: DEBUG/dalvikvm(217): GC_CONCURRENT freed 476K, 8%
free 6892K/7431K, paused 2ms+3ms
07-28 20:55:58.460: DEBUG/SipSession(217): session key from event:
a0aaa95da53e5c8b00d1dbc183bc1a2c@192.168.0.50
07-28 20:55:58.460: DEBUG/SipSession(217): active sessions:
07-28 20:55:58.460: DEBUG/
SipSession(217):  ...a0aaa95da53e5c8b00d1dbc183bc1a2c@192.168.0.50:
@408aa5b0:DEREGISTERING
07-28 20:55:58.460: DEBUG/SipSession(217):  ~
@408aa5b0:DEREGISTERING: DEREGISTERING: processing SIP/2.0 100 Trying
07-28 20:55:58.460: DEBUG/SipSession(217): Call-ID:
a0aaa95da53e5c8b00d1dbc183bc1a2c@192.168.0.50
07-28 20:55:58.460: DEBUG/SipSession(217): CSeq: 7834 REGISTER
07-28 20:55:58.460: DEBUG/SipSession(217): From: Freebox
sip:xxx...@freephonie.net;tag=1576196663
07-28 20:55:58.460: 

[android-developers] Re: sendDtmf on Xoom WiFI / Android 3.0

2011-07-27 Thread Robert Auger
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,
  

[android-developers] Re: sendDtmf on Xoom WiFI / Android 3.0

2011-07-16 Thread Robert Auger
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.

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