[Asterisk-Users] x100p to X-lite works but x-lite to x-lite not (can not transmit audio)

[Nestor A. Diaz L.]

Hello People,

I am a newbie asterisk and happy user, i have configured a x100p card and 
everything works nice, i can forward incoming connections to a x-lite
software client and works out of the box,

However when i try to make a connection between two x-lite clients then no
audio is transmited, i have followed the instructions on voip-info.org,
the tutorials on onlamp and i have read some instructions on the net,
and i still have not found the answer, in conclusion:

I have two x-lite clients, that can call each other, connection is
stablished but no audio is transmited, i follow the recomendations:

1. Install the iblc and spx registry patch (Windows 2K)
2. Work only with the alaw codec
3. Disable silence suppresion.

but i still get:

RFC3389 support incomplete. Turn off on client if possible
RFC3389: 5 bytes, level 0...
RFC3389: 5 bytes, level 0...

The above message also is showing when the call is comming from 
a zap defice and the application Dial (Zap, SIP/313) is executed (without
the RFC3389: 5 bytes, level 0...)  but it works this way.

I run asterisk from the command line as user asterisk like this:

asterisk -vvvvvgcd

This is my sip.conf:

[general]

port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
allow=all             ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here

[312]
type=friend
username=312
secret=123456
host=dynamic
disallow=all
allow=alaw
context=from-sip

[313]
type=friend
username=313
secret=123456
host=dynamic
disallow=all
allow=alaw
context=from-sip

The extensions.conf:

[from-sip]

exten => 312,1,Dial(SIP/312,10)
exten => 312,2,Voicemail(u312)
exten => 312,102,Voicemail(b312)
exten => 312,103,Hangup

exten => 313,1,Dial(SIP/313,10)
exten => 313,2,Voicemail(u313)
exten => 313,102,Voicemail(b313)
exten => 313,103,Hangup

Voicemail works, but i can not leave a message from a sip phone:

an  7 08:25:32 WARNING[393234]: app.c:615 ast_play_and_record: No audio available
 on SIP/313-47b0??
    -- User hung up
Urgent handler

but i can do that from a zap device.

I use asterisk debian's packages from testing.

ii  asterisk       1.0.2-2        Open Source Private Branch Exchange (PBX)
ii  asterisk-doc   1.0.2-2        Documentation for asterisk
ii  asterisk-sound 1.0.2-2        Sound files for asterisk

I like to have the x-lite clients working, any help will be apreciated.

Thanks you very much for your time.

--
Nestor A. Diaz Lizarazo				   Tel. +57.1.6005490
Ingeniero de Sistemas y Comp.			     Cel. 315 8190760
nestor at tiendalinux.com	       	       http://soporte.tiendalinux.com