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 -vgcd
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 1.0.2-2Open Source Private Branch Exchange (PBX)
ii asterisk-doc 1.0.2-2Documentation for asterisk
ii asterisk-sound 1.0.2-2Sound 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 LizarazoTel. +57.1.6005490
Ingeniero de Sistemas y Comp.Cel. 315 8190760
[EMAIL PROTECTED] http://soporte.tiendalinux.com
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