I was able to solve this by simply putting the full address in, so for example if trying to dial extension 900, instead of just putting 900, I had to put in "900@asterisk_server_address" to get it to work. Hope this helps!
On May 18, 4:07 pm, Shaun Clark <[email protected]> wrote: > When I try to fire off aSIPcall I get: > > 05-18 16:01:50.303: DEBUG/dalvikvm(1179): GC_EXPLICIT freed 11K, 6% > free 6240K/6595K, paused 2ms+1ms > 05-18 16:01:55.273: DEBUG/BatteryService(162): level:88 scale:100 > status:2 health:2 present:true voltage: 8325 temperature: 357 > technology: Li-ion AC powered:true USB powered:false icon:17302715 > invalid charger:0 > 05-18 16:01:55.283: DEBUG/PowerUI(267): closing low battery warning: > level=88 > 05-18 16:01:55.283: DEBUG/WifiService(162): ACTION_BATTERY_CHANGED > pluggedType: 1 > 05-18 16:01:55.393: DEBUG/dalvikvm(291): GC_CONCURRENT freed 546K, 10% > free 7569K/8327K, paused 2ms+3ms > 05-18 16:02:05.053: DEBUG/dalvikvm(1189): GC_EXPLICIT freed 13K, 6% > free 6236K/6595K, paused 5ms+1ms > 05-18 16:02:10.303: DEBUG/dalvikvm(1198): GC_EXPLICIT freed 19K, 6% > free 6247K/6595K, paused 1ms+2ms > 05-18 16:02:15.413: DEBUG/dalvikvm(291): GC_CONCURRENT freed 394K, 10% > free 7570K/8327K, paused 3ms+3ms > 05-18 16:02:21.253: DEBUG/dalvikvm(2361): GC_EXPLICIT freed 184K, 9% > free 7235K/7943K, paused 6ms+2ms > 05-18 16:02:25.513: INFO/dalvikvm(3308): threadid=1: recursive native > library load attempt (/system/lib/librtp_jni.so) > 05-18 16:02:25.573: DEBUG/SipHelper(291): send INVITE: > INVITEsip:null@902SIP/2.0 > 05-18 16:02:25.573: DEBUG/SipHelper(291): Call-ID: > [email protected] > 05-18 16:02:25.573: DEBUG/SipHelper(291): CSeq: 5416 INVITE > 05-18 16:02:25.573: DEBUG/SipHelper(291): From: <sip: > 901@asteriskserver>;tag=978053780 > 05-18 16:02:25.573: DEBUG/SipHelper(291): To: <sip:null@902> > 05-18 16:02:25.573: DEBUG/SipHelper(291): Via:SIP/2.0/UDP > 10.3.10.150:45767;branch=z9hG4bK67929aa9f0283695cebd7e8de5038ee6323239;rpor t > 05-18 16:02:25.573: DEBUG/SipHelper(291): Max-Forwards: 70 > 05-18 16:02:25.573: DEBUG/SipHelper(291): Contact: <sip: > [email protected]:45767;transport=udp> > 05-18 16:02:25.573: DEBUG/SipHelper(291): Content-Type: application/ > sdp > 05-18 16:02:25.573: DEBUG/SipHelper(291): Content-Length: 295 > 05-18 16:02:25.573: DEBUG/SipHelper(291): > 05-18 16:02:25.573: DEBUG/SipHelper(291): v=0 > 05-18 16:02:25.573: DEBUG/SipHelper(291): o=- 1305759745477 > 1305759745526 IN IP4 10.3.10.150 > 05-18 16:02:25.573: DEBUG/SipHelper(291): s=- > 05-18 16:02:25.573: DEBUG/SipHelper(291): c=IN IP4 10.3.10.150 > 05-18 16:02:25.573: DEBUG/SipHelper(291): t=0 0 > 05-18 16:02:25.573: DEBUG/SipHelper(291): m=audio 27272 RTP/AVP 96 97 > 3 0 8 127 > 05-18 16:02:25.573: DEBUG/SipHelper(291): a=rtpmap:96 GSM-EFR/8000 > 05-18 16:02:25.573: DEBUG/SipHelper(291): a=rtpmap:97 AMR/8000 > 05-18 16:02:25.573: DEBUG/SipHelper(291): a=rtpmap:3 GSM/8000 > 05-18 16:02:25.573: DEBUG/SipHelper(291): a=rtpmap:0 PCMU/8000 > 05-18 16:02:25.573: DEBUG/SipHelper(291): a=rtpmap:8 PCMA/8000 > 05-18 16:02:25.573: DEBUG/SipHelper(291): a=rtpmap:127 telephone-event/ > 8000 > 05-18 16:02:25.573: DEBUG/SipHelper(291): a=fmtp:127 0-15 > 05-18 16:02:25.593: WARN/SipSession(291): command error: > com.android.server.sip.SipSessionGroup > $MakeCallCommand[source=android.net.sip.SipProfile@408d8710] > 05-18 16:02:25.593: WARN/SipSession(291): javax.sip.SipException: IO > Error sending request > 05-18 16:02:25.593: WARN/SipSession(291): at > gov.nist.javax.sip.stack.SIPClientTransaction.sendRequest(SIPClientTransact > ion.java: > 971) > 05-18 16:02:25.593: WARN/SipSession(291): at > com.android.server.sip.SipHelper.sendInvite(SipHelper.java:272) > 05-18 16:02:25.593: WARN/SipSession(291): at > com.android.server.sip.SipSessionGroup > $SipSessionImpl.readyForCall(SipSessionGroup.java:939) > 05-18 16:02:25.593: WARN/SipSession(291): at > com.android.server.sip.SipSessionGroup > $SipSessionImpl.process(SipSessionGroup.java:639) > 05-18 16:02:25.593: WARN/SipSession(291): at > com.android.server.sip.SipSessionGroup > $SipSessionImpl.processCommand(SipSessionGroup.java:591) > 05-18 16:02:25.593: WARN/SipSession(291): at > com.android.server.sip.SipSessionGroup$SipSessionImpl.access > $1500(SipSessionGroup.java:385) > 05-18 16:02:25.593: WARN/SipSession(291): at > com.android.server.sip.SipSessionGroup$SipSessionImpl > $1.run(SipSessionGroup.java:514) > 05-18 16:02:25.593: WARN/SipSession(291): at > java.lang.Thread.run(Thread.java:1020) > 05-18 16:02:25.593: WARN/SipSession(291): Caused by: > java.net.SocketException: Invalid argument > 05-18 16:02:25.593: WARN/SipSession(291): at > org.apache.harmony.luni.platform.OSNetworkSystem.send(Native Method) > 05-18 16:02:25.593: WARN/SipSession(291): at > dalvik.system.BlockGuard$WrappedNetworkSystem.send(BlockGuard.java: > 320) > 05-18 16:02:25.593: WARN/SipSession(291): at > org.apache.harmony.luni.net.PlainDatagramSocketImpl.send(PlainDatagramSocke > tImpl.java: > 187) > 05-18 16:02:25.593: WARN/SipSession(291): at > java.net.DatagramSocket.send(DatagramSocket.java:435) > 05-18 16:02:25.593: WARN/SipSession(291): at > gov.nist.javax.sip.stack.UDPMessageChannel.sendMessage(UDPMessageChannel.ja > va: > 724) > 05-18 16:02:25.593: WARN/SipSession(291): at > gov.nist.javax.sip.stack.MessageChannel.sendMessage(MessageChannel.java: > 250) > 05-18 16:02:25.593: WARN/SipSession(291): at > gov.nist.javax.sip.stack.SIPTransaction.sendMessage(SIPTransaction.java: > 734) > 05-18 16:02:25.593: WARN/SipSession(291): at > gov.nist.javax.sip.stack.SIPClientTransaction.sendMessage(SIPClientTransact > ion.java: > 476) > 05-18 16:02:25.593: WARN/SipSession(291): at > gov.nist.javax.sip.stack.SIPClientTransaction.sendRequest(SIPClientTransact > ion.java: > 967) > 05-18 16:02:25.593: WARN/SipSession(291): ... 7 more > 05-18 16:02:25.593: DEBUG/SipAudioCall(3308):sipsession error: > SOCKET_ERROR: java.net.SocketException: Invalid argument > 05-18 16:02:25.593: DEBUG/SipAudioCall(3308): stop audiocall > 05-18 16:02:25.663: WARN/Resources(278): Converting to boolean: > TypedValue{t=0x3/d=0x15 "true" a=-1} > 05-18 16:02:25.663: WARN/Resources(278): Converting to boolean: > TypedValue{t=0x3/d=0x15 "true" a=-1} > 05-18 16:02:25.833: DEBUG/dalvikvm(291): GC_CONCURRENT freed 390K, 10% > free 7565K/8327K, paused 3ms+3ms > 05-18 16:02:30.923: DEBUG/dalvikvm(296): GC_EXPLICIT freed 99K, 9% > free 6433K/7047K, paused 2ms+2ms > 05-18 16:02:35.333: DEBUG/SipSession(291): ~~~~~ > @408b1ea0:READY_TO_CALL: READY_TO_CALL: processing OPTIONSsip: > [email protected]:45767;transport=udpSIP/2.0 > 05-18 16:02:35.333: DEBUG/SipSession(291): Via:SIP/2.0/UDP > asteriskserver: > 5060;branch=z9hG4bK4acd3e5f;rport=5060;received=asteriskserver > 05-18 16:02:35.333: DEBUG/SipSession(291): Max-Forwards: 70 > 05-18 16:02:35.333: DEBUG/SipSession(291): From: "Unknown" > <sip:Unknown@asteriskserver>;tag=as116a5a19 > 05-18 16:02:35.333: DEBUG/SipSession(291): To: <sip: > [email protected]:45767;transport=udp> > 05-18 16:02:35.333: DEBUG/SipSession(291): Contact: > <sip:Unknown@asteriskserver> > 05-18 16:02:35.333: DEBUG/SipSession(291): Call-ID: > 486385d11685bbe5631b63d83a067b59@asteriskserver > 05-18 16:02:35.333: DEBUG/SipSession(291): CSeq: 102 OPTIONS > 05-18 16:02:35.333: DEBUG/SipSession(291): User-Agent: > FPBX-2.6.0(1.6.1.11) > 05-18 16:02:35.333: DEBUG/SipSession(291): Date: Wed, 18 May 2011 > 23:02:23 GMT > 05-18 16:02:35.333: DEBUG/SipSession(291): Allow: > INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO > 05-18 16:02:35.333: DEBUG/SipSession(291): Supported: replaces,timer > 05-18 16:02:35.333: DEBUG/SipSession(291): Content-Length: 0 > 05-18 16:02:35.333: DEBUG/SipSession(291): > 05-18 16:02:35.333: DEBUG/SipHelper(291): send response:SIP/2.0 200 > OK > 05-18 16:02:35.333: DEBUG/SipHelper(291): Via:SIP/2.0/UDP > asteriskserver: > 5060;branch=z9hG4bK4acd3e5f;rport=5060;received=asteriskserver > 05-18 16:02:35.333: DEBUG/SipHelper(291): From: "Unknown" > <sip:Unknown@asteriskserver>;tag=as116a5a19 > 05-18 16:02:35.333: DEBUG/SipHelper(291): To: <sip: > [email protected]:45767;transport=udp> > 05-18 16:02:35.333: DEBUG/SipHelper(291): Call-ID: > 486385d11685bbe5631b63d83a067b59@asteriskserver > 05-18 16:02:35.333: DEBUG/SipHelper(291): CSeq: 102 OPTIONS > 05-18 16:02:35.333: DEBUG/SipHelper(291): Content-Length: 0 > 05-18 16:02:35.333: DEBUG/SipHelper(291): > 05-18 16:02:35.333: DEBUG/SipSession(291): new state after: > READY_TO_CALL > > 05-18 16:03:35.443: DEBUG/SipSession(291): ~~~~~ > @408b1ea0:READY_TO_CALL: READY_TO_CALL: processing OPTIONSsip: > [email protected]:45767;transport=udpSIP/2.0 > 05-18 16:03:35.443: DEBUG/SipSession(291): Via:SIP/2.0/UDP > asteriskserver: > 5060;branch=z9hG4bK6728da15;rport=5060;received=asteriskserver > 05-18 16:03:35.443: DEBUG/SipSession(291): Max-Forwards: 70 > 05-18 16:03:35.443: DEBUG/SipSession(291): From: "Unknown" > <sip:Unknown@asteriskserver>;tag=as51f381f9 > 05-18 16:03:35.443: DEBUG/SipSession(291): To: <sip: > [email protected]:45767;transport=udp> > 05-18 16:03:35.443: DEBUG/SipSession(291): Contact: > <sip:Unknown@asteriskserver> > 05-18 16:03:35.443: DEBUG/SipSession(291): Call-ID: > [email protected] > 05-18 16:03:35.443: DEBUG/SipSession(291): CSeq: 102 OPTIONS > 05-18 16:03:35.443: DEBUG/SipSession(291): User-Agent: > FPBX-2.6.0(1.6.1.11) > 05-18 16:03:35.443: DEBUG/SipSession(291): Date: Wed, 18 May 2011 > 23:03:23 GMT > 05-18 16:03:35.443: DEBUG/SipSession(291): Allow: > INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO > 05-18 16:03:35.443: DEBUG/SipSession(291): Supported: replaces,timer > 05-18 16:03:35.443: DEBUG/SipSession(291): Content-Length: 0 > 05-18 16:03:35.443: DEBUG/SipSession(291): > 05-18 16:03:35.443: DEBUG/SipHelper(291): send response:SIP/2.0 200 > OK > 05-18 16:03:35.443: DEBUG/SipHelper(291): Via:SIP/2.0/UDP > asteriskserver: > 5060;branch=z9hG4bK6728da15;rport=5060;received=asteriskserver > 05-18 16:03:35.443: DEBUG/SipHelper(291): From: "Unknown" > <sip:Unknown@asteriskserver>;tag=as51f381f9 > 05-18 16:03:35.443: DEBUG/SipHelper(291): To: <sip: > [email protected]:45767;transport=udp> > 05-18 16:03:35.443: DEBUG/SipHelper(291): Call-ID: > 75a3fa453447e5a328a64de01ba8086f@asteriskserver > 05-18 16:03:35.443: DEBUG/SipHelper(291): CSeq: 102 OPTIONS > 05-18 16:03:35.443: DEBUG/SipHelper(291): Content-Length: 0 > 05-18 16:03:35.443: DEBUG/SipHelper(291): > 05-18 16:03:35.453: DEBUG/SipSession(291): new state after: > READY_TO_CALL > > Any ideas why this doesn't work? I can see it register on the asterisk > server fine, but nothing happens on the server when the call attempt > is made. The otherSIPclient like SIPDroid can make it through with > the same username/password/server combo. Thanks for the help! > > Shaun -- You received this message because you are subscribed to the Google Groups "Android Developers" group. To post to this group, send email to [email protected] To unsubscribe from this group, send email to [email protected] For more options, visit this group at http://groups.google.com/group/android-developers?hl=en
