When I try to fire off a SIP call 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: INVITE
sip:null@902 SIP/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;rport
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(SIPClientTransaction.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(PlainDatagramSocketImpl.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.java:
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(SIPClientTransaction.java:
476)
05-18 16:02:25.593: WARN/SipSession(291):     at
gov.nist.javax.sip.stack.SIPClientTransaction.sendRequest(SIPClientTransaction.java:
967)
05-18 16:02:25.593: WARN/SipSession(291):     ... 7 more
05-18 16:02:25.593: DEBUG/SipAudioCall(3308): sip session 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 OPTIONS sip:
[email protected]:45767;transport=udp SIP/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 OPTIONS sip:
[email protected]:45767;transport=udp SIP/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 other SIP client like SIPDroid can make it through with
the same username/password/server combo. Thanks for the help!

Shaun

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