asterisk users - Feb 2005 - HELP!!!!!!!!

Hi,

I have installed two X-Lite phones and they're able to login successfully.
The two phones plus the Asterisk system are all on the same LAN with private
addresses assigned to each of them.  When a call is initiated and is picked
up on the other end, there is completely no sound at all (as in the line
goes dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and
SPX.
>From the Asterisk CLI I see the following errors;
i)                     Unknown RTP codec 72 received

ii)                   RFC3389 support incomplete

Anyone got ideas on how I can go about this?

Thanks in advance.

Julius Kidubuka

"When you do the common things in life in an uncommon way, you will command
the attention of the world"

 

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Yes turn off silence suppression.

xlite - Menu - Advanced - audio settings - Silence Settings - transmite Silence:
(change to yes)
  ----- Original Message ----- 
  From: Julius Kidubuka 
  To: asterisk-users@lists.digium.com 
  Sent: Wednesday, February 16, 2005 10:04 AM
  Subject: [Asterisk-Users] HELP!!!!!!!!


  Hi,

  I have installed two X-Lite phones and they're able to login successfully.
The two phones plus the Asterisk system are all on the same LAN with private
addresses assigned to each of them.  When a call is initiated and is picked up
on the other end, there is completely no sound at all (as in the line goes
dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and SPX.

  From the Asterisk CLI I see the following errors;

  i)                     Unknown RTP codec 72 received

  ii)                   RFC3389 support incomplete

  Anyone got ideas on how I can go about this?

  Thanks in advance.

  Julius Kidubuka

  "When you do the common things in life in an uncommon way, you will
command the attention of the world"

   



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I new to this as will. But add more info like your sip.conf file.

David


On Wed, 2005-02-16 at 18:04 +0300, Julius Kidubuka
wrote:
> Hi,
> 
> I have installed two X-Lite phones and they?re able to login
> successfully. The two phones plus the Asterisk system are all on the
> same LAN with private addresses assigned to each of them.  When a call
> is initiated and is picked up on the other end, there is completely no
> sound at all (as in the line goes dead). The codecs set in the
> softphones are g711u, g711a, GSM, iLBC and SPX.
> 
> From the Asterisk CLI I see the following errors;
> 
> i)                    Unknown RTP codec 72 received
> 
> ii)                  RFC3389 support incomplete
> 
> Anyone got ideas on how I can go about this?
> 
> Thanks in advance.
> 
> Julius Kidubuka
> 
> "When you do the common things in life in an uncommon way, you will
> command the attention of the world"
> 
>  
> 
> 
> 
> -- 
> This message has been scanned for viruses and 
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-- 
David Shaw <asterisk@ke6upi.com>

Oswaldo Arratia wrote:
> Has anyone figured out how to make a Sipura to dial an extension
> automatically as soon as you pick the the handset?
Go to google and type: sipura hotline

Read the first three links.

Test.

Send us a note telling what worked for you.

-- 
Andrew Thompson
http://aktzero.com/
http://dev.asteriskdocs.org/