[sip-comm] only receive audio when using sip-communicator with trixbox (asterisk)


#1

I only receive audio when I connect between 2 PC's using sip communicator and trixbox asterisk. Does anyone have any ideas?

···

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#2

Hi,

Which build of sip-communicator are you using ?
Any NAT between PSs, what codec are you using?
Have you seen for any exceptions in the logs or in the console ?

damencho

···

On 9/11/07, FRANKOFF.S@forces.gc.ca <FRANKOFF.S@forces.gc.ca> wrote:

I only receive audio when I connect between 2 PC's using sip communicator and trixbox asterisk. Does anyone have any ideas?

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#3

Further elaboration of the problem:

- there is no NAT there are 2 PC's running the client connected through a switch to a server running asterisk
- I am using the latest build (nightly snapshot)
- If I call PC "B" from PC "A" there are no issues and everything works fine
- If I call PC "A" from PC "B" the console shows a "payload change error" and I can only recive audio
- The codecs selected are GSM or ULAW
- The PC's are configured the same that is not the issue

why does this work one way and not the other?

···

-----Original Message-----

From: Damian Minkov [mailto:damencho@damencho.com]

Sent: Tuesday, 11 September, 2007 10:20
To: users@sip-communicator.dev.java.net
Subject: Re: [sip-comm] only receive audio when using sip-communicator
with trixbox (asterisk)

Hi,

Which build of sip-communicator are you using ?
Any NAT between PSs, what codec are you using?
Have you seen for any exceptions in the logs or in the console ?

damencho

On 9/11/07, FRANKOFF.S@forces.gc.ca <FRANKOFF.S@forces.gc.ca> wrote:

I only receive audio when I connect between 2 PC's using sip communicator and trixbox asterisk. Does anyone have any ideas?

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#4

Hi,

On both PCs you are using sip-communicator ?
Can you send me the log. Have you access to the asterisk box ?
Is the sip-communicator account in * configured with canreinvite=no ?

damencho

···

FRANKOFF.S@forces.gc.ca wrote:

Further elaboration of the problem:

- there is no NAT there are 2 PC's running the client connected through a switch to a server running asterisk
- I am using the latest build (nightly snapshot)
- If I call PC "B" from PC "A" there are no issues and everything works fine
- If I call PC "A" from PC "B" the console shows a "payload change error" and I can only recive audio
- The codecs selected are GSM or ULAW
- The PC's are configured the same that is not the issue

why does this work one way and not the other?

-----Original Message-----
From: Damian Minkov [mailto:damencho@damencho.com]
Sent: Tuesday, 11 September, 2007 10:20
To: users@sip-communicator.dev.java.net
Subject: Re: [sip-comm] only receive audio when using sip-communicator
with trixbox (asterisk)

Hi,

Which build of sip-communicator are you using ?
Any NAT between PSs, what codec are you using?
Have you seen for any exceptions in the logs or in the console ?

damencho

On 9/11/07, FRANKOFF.S@forces.gc.ca <FRANKOFF.S@forces.gc.ca> wrote:
  

I only receive audio when I connect between 2 PC's using sip communicator and trixbox asterisk. Does anyone have any ideas?

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#5

I am using the JAIN sip-applet phone on both pc's. If you are not familiar with this it is the same as the SIP part of the sip communicator (hence me posting on this forum). In answer to your first thought yes I do get the same results when using the actual sip communicator.

I have access to the asterisk box it is currently dedicated to this testing alone and yes "canreinvite=no" for all accounts

The following is a print out of the callee from the console (the caller printout is after):

···

-------------------------------------------------------------------------------------------------------------------------------------------------------------
Init UDP Transmitter
Threshold enabled : true
buf length : 0
minimum Threshold : 2147483647
  - Waiting for RTP data to arrive...
  - Open RTP session for: Address: 192.168.1.10 localPort: 1590 destPort : 13074 Time To Live: 60
- number of capture devices: 2
    - name of the capture device: DirectSoundCapture
         - format accepted by this AUDIO device: LINEAR, 48000.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 48000.0 Hz, 8-bit, Stereo, Unsigned
         - format accepted by this AUDIO device: LINEAR, 48000.0 Hz, 8-bit, Mono, Unsigned
         - format accepted by this AUDIO device: LINEAR, 44100.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 44100.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 44100.0 Hz, 8-bit, Stereo, Unsigned
         - format accepted by this AUDIO device: LINEAR, 44100.0 Hz, 8-bit, Mono, Unsigned
         - format accepted by this AUDIO device: LINEAR, 32000.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 32000.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 32000.0 Hz, 8-bit, Stereo, Unsigned
         - format accepted by this AUDIO device: LINEAR, 32000.0 Hz, 8-bit, Mono, Unsigned
         - format accepted by this AUDIO device: LINEAR, 22050.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 22050.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 22050.0 Hz, 8-bit, Stereo, Unsigned
         - format accepted by this AUDIO device: LINEAR, 22050.0 Hz, 8-bit, Mono, Unsigned
         - format accepted by this AUDIO device: LINEAR, 16000.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 16000.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 16000.0 Hz, 8-bit, Stereo, Unsigned
         - format accepted by this AUDIO device: LINEAR, 16000.0 Hz, 8-bit, Mono, Unsigned
         - format accepted by this AUDIO device: LINEAR, 11025.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 11025.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 11025.0 Hz, 8-bit, Stereo, Unsigned
         - format accepted by this AUDIO device: LINEAR, 11025.0 Hz, 8-bit, Mono, Unsigned
         - format accepted by this AUDIO device: LINEAR, 8000.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 8000.0 Hz, 8-bit, Stereo, Unsigned
         - format accepted by this AUDIO device: LINEAR, 8000.0 Hz, 8-bit, Mono, Unsigned
    - name of the capture device: JavaSound audio capture
         - format accepted by this AUDIO device: LINEAR, 44100.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 44100.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 22050.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 22050.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 11025.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 11025.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 8000.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed
Track 0 is set to transmit as:
  gsm/rtp, 8000.0 Hz, Mono, FrameSize=264 bits
Created RTP session: 192.168.1.10 dest 13074
Media Transmission started!!!
Get chat session: 202@192.168.1.10
callee 202@192.168.1.10:chatFrame gov.nist.applet.phone.ua.gui.ChatFrame[frame0,0,0,750x430,invalid,layout=java.awt.BorderLayout,title=In conversation with 202@192.168.1.10,resizable,normal,defaultCloseOperation=DISPOSE_ON_CLOSE,rootPane=javax.swing.JRootPane[,4,23,742x403,invalid,layout=javax.swing.JRootPane$RootLayout,alignmentX=0.0,alignmentY=0.0,border=,flags=16777673,maximumSize=,minimumSize=,preferredSize=],rootPaneCheckingEnabled=true]:status In a call
  - Received new RTP stream: gsm/rtp, 8000.0 Hz, Mono
      The sender of this stream had yet to be identified.
1
format list : 1
mpegControl::::null
mpegControl::::null
mpegControl::::null
mpegControl::::null
mpegControl::::null
Class for gain contolcom.sun.media.renderer.audio.DirectAudioRenderer$MCA@126d3df
Class for componentnull
-------------------------------------------------------------------------------------------------------------------------------------------------------------

Print out from the caller console:

-------------------------------------------------------------------------------------------------------------------------------------------------------------
Track 0 is set to transmit as:
  gsm/rtp, 8000.0 Hz, Mono, FrameSize=264 bits
  - Received new RTP stream: ULAW/rtp, 8000.0 Hz, 8-bit, Mono
      The sender of this stream had yet to be identified.
1
format list : 1
mpegControl::::null
Created RTP session: 192.168.1.10 dest 14554
Media Transmission started!!!
Get chat session: 201@192.168.1.251
callee 201@192.168.1.251:chatFrame gov.nist.applet.phone.ua.gui.ChatFrame[frame0,0,0,750x430,invalid,layout=java.awt.BorderLayout,title=In conversation with 201@192.168.1.251,resizable,normal,defaultCloseOperation=DISPOSE_ON_CLOSE,rootPane=javax.swing.JRootPane[,4,23,742x403,layout=javax.swing.JRootPane$RootLayout,alignmentX=0.0,alignmentY=0.0,border=,flags=16777673,maximumSize=,minimumSize=,preferredSize=],rootPaneCheckingEnabled=true]:status In a call
mpegControl::::null
mpegControl::::null
mpegControl::::null
mpegControl::::null
Class for gain contolcom.sun.media.renderer.audio.DirectAudioRenderer$MCA@44f787
Class for componentnull
  - Received an RTP PayloadChangeEvent.
Sorry, cannot handle payload change.
-------------------------------------------------------------------------------------------------------------------------------------------------------------

-----Original Message-----

From: Damian Minkov [mailto:damencho@damencho.com]

Sent: Tuesday, 11 September, 2007 11:54
To: users@sip-communicator.dev.java.net
Subject: Re: [sip-comm] only receive audio when using sip-communicator
with trixbox (asterisk)

Hi,

On both PCs you are using sip-communicator ?
Can you send me the log. Have you access to the asterisk box ?
Is the sip-communicator account in * configured with canreinvite=no ?

damencho

FRANKOFF.S@forces.gc.ca wrote:

Further elaboration of the problem:

- there is no NAT there are 2 PC's running the client connected through a switch to a server running asterisk
- I am using the latest build (nightly snapshot)
- If I call PC "B" from PC "A" there are no issues and everything works fine
- If I call PC "A" from PC "B" the console shows a "payload change error" and I can only recive audio
- The codecs selected are GSM or ULAW
- The PC's are configured the same that is not the issue

why does this work one way and not the other?

-----Original Message-----
From: Damian Minkov [mailto:damencho@damencho.com]
Sent: Tuesday, 11 September, 2007 10:20
To: users@sip-communicator.dev.java.net
Subject: Re: [sip-comm] only receive audio when using sip-communicator
with trixbox (asterisk)

Hi,

Which build of sip-communicator are you using ?
Any NAT between PSs, what codec are you using?
Have you seen for any exceptions in the logs or in the console ?

damencho

On 9/11/07, FRANKOFF.S@forces.gc.ca <FRANKOFF.S@forces.gc.ca> wrote:
  

I only receive audio when I connect between 2 PC's using sip communicator and trixbox asterisk. Does anyone have any ideas?

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For additional commands, e-mail: users-help@sip-communicator.dev.java.net

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For additional commands, e-mail: users-help@sip-communicator.dev.java.net

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#6

Hi,

sorry for the late answer but we were quite busy these days.
Can you replace the jmf used in the JAIN sip-applet whit those from sip-communicator svn :

https://sip-communicator.dev.java.net/svn/sip-communicator/trunk/lib/os-specific/windows/installer-exclude/
or
https://sip-communicator.dev.java.net/svn/sip-communicator/trunk/lib/os-specific/linux/installer-exclude/

depending you OS

And report is the problem the same. As we are testing sip-communicator with asterisk and have no problems.

damencho

···

FRANKOFF.S@forces.gc.ca wrote:

I am using the JAIN sip-applet phone on both pc's. If you are not familiar with this it is the same as the SIP part of the sip communicator (hence me posting on this forum). In answer to your first thought yes I do get the same results when using the actual sip communicator.

I have access to the asterisk box it is currently dedicated to this testing alone and yes "canreinvite=no" for all accounts

The following is a print out of the callee from the console (the caller printout is after):

-------------------------------------------------------------------------------------------------------------------------------------------------------------
Init UDP Transmitter
Threshold enabled : true
buf length : 0
minimum Threshold : 2147483647
  - Waiting for RTP data to arrive...
  - Open RTP session for: Address: 192.168.1.10 localPort: 1590 destPort : 13074 Time To Live: 60
- number of capture devices: 2
    - name of the capture device: DirectSoundCapture
         - format accepted by this AUDIO device: LINEAR, 48000.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 48000.0 Hz, 8-bit, Stereo, Unsigned
         - format accepted by this AUDIO device: LINEAR, 48000.0 Hz, 8-bit, Mono, Unsigned
         - format accepted by this AUDIO device: LINEAR, 44100.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 44100.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 44100.0 Hz, 8-bit, Stereo, Unsigned
         - format accepted by this AUDIO device: LINEAR, 44100.0 Hz, 8-bit, Mono, Unsigned
         - format accepted by this AUDIO device: LINEAR, 32000.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 32000.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 32000.0 Hz, 8-bit, Stereo, Unsigned
         - format accepted by this AUDIO device: LINEAR, 32000.0 Hz, 8-bit, Mono, Unsigned
         - format accepted by this AUDIO device: LINEAR, 22050.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 22050.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 22050.0 Hz, 8-bit, Stereo, Unsigned
         - format accepted by this AUDIO device: LINEAR, 22050.0 Hz, 8-bit, Mono, Unsigned
         - format accepted by this AUDIO device: LINEAR, 16000.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 16000.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 16000.0 Hz, 8-bit, Stereo, Unsigned
         - format accepted by this AUDIO device: LINEAR, 16000.0 Hz, 8-bit, Mono, Unsigned
         - format accepted by this AUDIO device: LINEAR, 11025.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 11025.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 11025.0 Hz, 8-bit, Stereo, Unsigned
         - format accepted by this AUDIO device: LINEAR, 11025.0 Hz, 8-bit, Mono, Unsigned
         - format accepted by this AUDIO device: LINEAR, 8000.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 8000.0 Hz, 8-bit, Stereo, Unsigned
         - format accepted by this AUDIO device: LINEAR, 8000.0 Hz, 8-bit, Mono, Unsigned
    - name of the capture device: JavaSound audio capture
         - format accepted by this AUDIO device: LINEAR, 44100.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 44100.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 22050.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 22050.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 11025.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 11025.0 Hz, 16-bit, Mono, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 8000.0 Hz, 16-bit, Stereo, LittleEndian, Signed
         - format accepted by this AUDIO device: LINEAR, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed
Track 0 is set to transmit as:
  gsm/rtp, 8000.0 Hz, Mono, FrameSize=264 bits
Created RTP session: 192.168.1.10 dest 13074
Media Transmission started!!!
Get chat session: 202@192.168.1.10
callee 202@192.168.1.10:chatFrame gov.nist.applet.phone.ua.gui.ChatFrame[frame0,0,0,750x430,invalid,layout=java.awt.BorderLayout,title=In conversation with 202@192.168.1.10,resizable,normal,defaultCloseOperation=DISPOSE_ON_CLOSE,rootPane=javax.swing.JRootPane[,4,23,742x403,invalid,layout=javax.swing.JRootPane$RootLayout,alignmentX=0.0,alignmentY=0.0,border=,flags=16777673,maximumSize=,minimumSize=,preferredSize=],rootPaneCheckingEnabled=true]:status In a call
  - Received new RTP stream: gsm/rtp, 8000.0 Hz, Mono
      The sender of this stream had yet to be identified.
1
format list : 1
mpegControl::::null
Class for gain contolcom.sun.media.renderer.audio.DirectAudioRenderer$MCA@126d3df
Class for componentnull
-------------------------------------------------------------------------------------------------------------------------------------------------------------

Print out from the caller console:

-------------------------------------------------------------------------------------------------------------------------------------------------------------
Track 0 is set to transmit as:
  gsm/rtp, 8000.0 Hz, Mono, FrameSize=264 bits
  - Received new RTP stream: ULAW/rtp, 8000.0 Hz, 8-bit, Mono
      The sender of this stream had yet to be identified.
1
format list : 1
mpegControl::::null
Created RTP session: 192.168.1.10 dest 14554
Media Transmission started!!!
Get chat session: 201@192.168.1.251
callee 201@192.168.1.251:chatFrame gov.nist.applet.phone.ua.gui.ChatFrame[frame0,0,0,750x430,invalid,layout=java.awt.BorderLayout,title=In conversation with 201@192.168.1.251,resizable,normal,defaultCloseOperation=DISPOSE_ON_CLOSE,rootPane=javax.swing.JRootPane[,4,23,742x403,layout=javax.swing.JRootPane$RootLayout,alignmentX=0.0,alignmentY=0.0,border=,flags=16777673,maximumSize=,minimumSize=,preferredSize=],rootPaneCheckingEnabled=true]:status In a call
mpegControl::::null
Class for gain contolcom.sun.media.renderer.audio.DirectAudioRenderer$MCA@44f787
Class for componentnull
  - Received an RTP PayloadChangeEvent.
Sorry, cannot handle payload change.
-------------------------------------------------------------------------------------------------------------------------------------------------------------

-----Original Message-----
From: Damian Minkov [mailto:damencho@damencho.com]
Sent: Tuesday, 11 September, 2007 11:54
To: users@sip-communicator.dev.java.net
Subject: Re: [sip-comm] only receive audio when using sip-communicator
with trixbox (asterisk)

Hi,

On both PCs you are using sip-communicator ?
Can you send me the log. Have you access to the asterisk box ?
Is the sip-communicator account in * configured with canreinvite=no ?

damencho

FRANKOFF.S@forces.gc.ca wrote:
  

Further elaboration of the problem:

- there is no NAT there are 2 PC's running the client connected through a switch to a server running asterisk
- I am using the latest build (nightly snapshot)
- If I call PC "B" from PC "A" there are no issues and everything works fine
- If I call PC "A" from PC "B" the console shows a "payload change error" and I can only recive audio
- The codecs selected are GSM or ULAW
- The PC's are configured the same that is not the issue

why does this work one way and not the other?

-----Original Message-----
From: Damian Minkov [mailto:damencho@damencho.com]
Sent: Tuesday, 11 September, 2007 10:20
To: users@sip-communicator.dev.java.net
Subject: Re: [sip-comm] only receive audio when using sip-communicator
with trixbox (asterisk)

Hi,

Which build of sip-communicator are you using ?
Any NAT between PSs, what codec are you using?
Have you seen for any exceptions in the logs or in the console ?

damencho

On 9/11/07, FRANKOFF.S@forces.gc.ca <FRANKOFF.S@forces.gc.ca> wrote:
  

I only receive audio when I connect between 2 PC's using sip communicator and trixbox asterisk. Does anyone have any ideas?

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