[asterisk-users] Help configuring Audiocodes MP-104 FXO
Daniel - Asterisk
earohuanca at gmail.com
Wed Dec 2 11:32:54 CST 2009
Hi list,
I'm trying to get ready the MP-104 FXO to use qith my box, but when I send
calls I hear only dial tone and after a few seconds I get busy signal.
I very appreciate your advices.
Command line results and SIPconfigurations follows:
*CLI>*
-- Executing [7991696900 at total:1] Playback("SIP/101-09dd8918", "beep")
in new stack
-- <SIP/101-09dd8918> Playing 'beep' (language 'es')
-- Executing [7991696900 at total:4] Dial("SIP/101-09dd8918",
"SIP/201/991696900") in new stack
-- Called 201/991696900
-- SIP/201-09ddc890 answered SIP/101-09dd8918
*sip.conf*
[201]
secret = ****
callerid = Mobile_01 <201>
type = friend
host = dynamic
context = total
dtmfmode=rfc2833
qualify = yes
call-limit=5
disallow = all
allow = gsm
allow = ulaw
allow = alaw
allow = g729
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091202/3fc2ec5f/attachment.htm
More information about the asterisk-users
mailing list