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Weird issue with Spark phone plugin

Hi,

I’m having a very strange issue…

I’ve got the latest version of Spark running 3 systems with identical OS’s - all Arch Linux 64 bit. All machines are up to date, the only thing seperating them is hardware and the packages installed, though each has sound and video working.

Spark works perfectly on all boxes in IM (chat) mode, however it only manages to call out on one client.

My setup is as such: Openfire (latest) running on FreeBSD 8.3 64bit with SIP connection to FreePBX running Asterisk version 1.8.18.

When I dial out one 2 of the 3 clients nothing happens and nothing is logged anywhere to be seen, however the working client is fine.

I’m running OpenJDK version 7 and the latest version from the Arch stable repos.

I’m wondering if anyone can help me?

Unfortunately I haven’t found a way to debug Spark, and the Openfire and FreePBX boxes have nothing in their logs either for me to display here that might be helpful.

The odd thing is that initially when fired up Spark manages to recieve a call on the 2 malfunctioning clients, but if rejected and dialed again; the system just responds with busy.

How can this work on one system but not two others? The settings are the same for each machine or similar, just different IP’s.

I have tried with different SIP users and the same SIP user across machines but the only one working is my initial client - the one I setup first.

Can anybody make sense of this?

Thanks

I have some output now as I wasn’t able to hear anything using the working box either:


FORMATE NEU: [speex/rtp, Unknown Sample Rate, ALAW/rtp, Unknown Sample Rate, ULAW/rtp, Unknown Sample Rate, gsm/rtp, Unknown Sample Rate, ilbc/rtp, Unknown Sample Rate, g723/rtp, Unknown Sample Rate]
TOASTER_REJECT_BUTTON not found.
Created RTP session at 5236
RTP Transmission Stopped.
Jun 3, 2013 9:58:20 PM net.sf.fmj.media.protocol.javasound.DataSource connect
WARNING: javax.sound.sampled.LineUnavailableException: line with format PCM_SIGNED 44100.0 Hz, 32 bit, stereo, 8 bytes/frame, little-endian not supported.
javax.sound.sampled.LineUnavailableException: line with format PCM_SIGNED 44100.0 Hz, 32 bit, stereo, 8 bytes/frame, little-endian not supported.
    at com.sun.media.sound.DirectAudioDevice$DirectDL.implOpen(Unknown Source)
    at com.sun.media.sound.AbstractDataLine.open(Unknown Source)
    at net.sf.fmj.media.protocol.javasound.DataSource.connect(DataSource.java:141)
    at javax.media.Manager.createDataSource(Manager.java:664)
    at org.jivesoftware.spark.phone.PhoneManager.getDataSource(PhoneManager.java:317)
    at net.java.sipmack.media.AudioChannel.createProcessor(AudioChannel.java:212)
    at net.java.sipmack.media.AudioChannel.start(AudioChannel.java:110)
    at net.java.sipmack.media.AudioMediaSession.startTrasmit(AudioMediaSession.java:16 2)
    at net.java.sipmack.softphone.SoftPhoneManager.callStateChanged(SoftPhoneManager.j ava:689)
    at net.java.sipmack.sip.Call.fireCallStatusChangedEvent(Call.java:329)
    at net.java.sipmack.sip.Call.setState(Call.java:195)
    at net.java.sipmack.sip.CallProcessing.processInviteOK(CallProcessing.java:202)
    at net.java.sipmack.sip.SipManager.processResponse(SipManager.java:1533)
    at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:290)
    at gov.nist.javax.sip.EventScanner.run(EventScanner.java:492)
    at java.lang.Thread.run(Unknown Source)
java.io.IOException: javax.sound.sampled.LineUnavailableException: line with format PCM_SIGNED 44100.0 Hz, 32 bit, stereo, 8 bytes/frame, little-endian not supported.
    at net.sf.fmj.media.protocol.javasound.DataSource.connect(DataSource.java:150)
    at javax.media.Manager.createDataSource(Manager.java:664)
    at org.jivesoftware.spark.phone.PhoneManager.getDataSource(PhoneManager.java:317)
    at net.java.sipmack.media.AudioChannel.createProcessor(AudioChannel.java:212)
    at net.java.sipmack.media.AudioChannel.start(AudioChannel.java:110)
    at net.java.sipmack.media.AudioMediaSession.startTrasmit(AudioMediaSession.java:16 2)
    at net.java.sipmack.softphone.SoftPhoneManager.callStateChanged(SoftPhoneManager.j ava:689)
    at net.java.sipmack.sip.Call.fireCallStatusChangedEvent(Call.java:329)
    at net.java.sipmack.sip.Call.setState(Call.java:195)
    at net.java.sipmack.sip.CallProcessing.processInviteOK(CallProcessing.java:202)
    at net.java.sipmack.sip.SipManager.processResponse(SipManager.java:1533)
    at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:290)
    at gov.nist.javax.sip.EventScanner.run(EventScanner.java:492)
    at java.lang.Thread.run(Unknown Source)
javax.media.NoProcessorException
    at javax.media.Manager.createProcessor(Manager.java:488)
    at javax.media.Manager.createProcessor(Manager.java:512)
    at net.java.sipmack.media.AudioChannel.createProcessor(AudioChannel.java:212)
    at net.java.sipmack.media.AudioChannel.start(AudioChannel.java:110)
    at net.java.sipmack.media.AudioMediaSession.startTrasmit(AudioMediaSession.java:16 2)
    at net.java.sipmack.softphone.SoftPhoneManager.callStateChanged(SoftPhoneManager.j ava:689)
    at net.java.sipmack.sip.Call.fireCallStatusChangedEvent(Call.java:329)
    at net.java.sipmack.sip.Call.setState(Call.java:195)
    at net.java.sipmack.sip.CallProcessing.processInviteOK(CallProcessing.java:202)
    at net.java.sipmack.sip.SipManager.processResponse(SipManager.java:1533)
    at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:290)
    at gov.nist.javax.sip.EventScanner.run(EventScanner.java:492)
    at java.lang.Thread.run(Unknown Source)

It’s almost like a sound issue, though I have ALSA runnig fine on all boxes and in Spark/Media Settings JavaSound comes up fine too.