[Asterisk-Users] Welltech FXO: initial tests

[Miguel]

Caio, 

Do you have the firmware files ?, I have a 3804 h323 and I'd like to upgrade
it to SIP.

The files are:
	- 2m4sipfxo.103
	- 4fxosip.103

Kind regards,

Miguel

-------------------

From: "Caio Augusto Martimiano da Costa" <caio at furukawa.com.br>
Subject: [Asterisk-Users] Welltech FXO: initial tests
To: <asterisk-users at lists.digium.com>
Message-ID: <s1ee8021.099 at ctb-fisa5.furukawa.com.br>
Content-Type: text/plain; charset="iso-8859-1"

Dear Claudio,
 
I'm testing the welltech gateways (3804 firmware 4fxosip.102) and I am
trying to make the Asterisk answer the calls from 3508 directly (with
2nddial off) it means throw hotline service.
Do you know how to make the Asterisk answer a call  from :pstn-to-3508 and
3508-hotline-Asterisk ?
Please let me know if the question is not clear enough !
My configurations are:
 
extension.conf:
 
[general]
static=yes
writeprotect=no
 
[default]
include => oi
 
exten => s,2,Answer
exten => s,3,Wait(1)
exten => s,4,Background(vm-toenternumber)    ; Qual a extenssco desejada?
exten => 1,1,Goto(oi)
exten => 2,1,Goto(oi)
exten => 3,1,Goto(oi)
exten => 6,1,Goto(oi)
exten => 11,1,playback(beep)
exten => 0,1,Goto(default,s,2)
exten => t,1,Goto(timeout,s,1)
exten => i,1,Goto,s|2
 
[oi]
exten => s,1,Wait(1)
exten => s,2,Answer
exten => s,3,Wait(1)
exten => s,4,playback(pbx-transfer)
exten => s,5,Goto(default,s,2)
 
[bogon-calls]
exten => _.,1,Congestion

 
sip.conf
 
[general]
port=5060
bindaddr=0.0.0.0
context=from-sip
;context=bogon-calls
;context=default
maxexpirey=3600
defaultexpirey=120
disallow=all
allow=ulaw
allow=alaw

[300]
port=5060
type=friend
context=default
username=9   ; Username to use in INVITE until peer registers
secret=fisa9   
host=10.150.3.100 
disallow=all
allow=ulaw
allow=alaw
;allow=g729

 
3804:
usr/config$ sip -print
 
    Run Mode                 : PROXY MODE
    Proxy server address     : 10.150.3.4
    Domain                   : null
    Prefix string            : 1234
    Line1                    : 100
    Line2                    : 101
    Line3                    : 102
    Line4                    : 103
    SIP port                 : 5060
    RTP port                 : 16384
    Expire                   : 3600
 
usr/config$ sysconf -print
 
System information
    Inter-Digit time out        : 1
    End of Dial                 :  No end of dial
    Port status:
        port1: Enabled
        port2: Enabled
        port3: Enabled
        port4: Enabled
    DTMF selection              : In-band
        RFC2833 Payload Type        : 96
        FAX Payload Type            : 101
    2nddial:  3
    Billing: OFF
    Dial Rule
         ip side:
         filter: []   drop: []   insert: [].
         pstn side:
         filter: []   drop: []   insert: [].
    PIN prompt: 0
         set1: 1111
         set2: 2222
         set3: 3333
         set4: 4444
    Ring Detect Method: 1
    Ring before Answer: 0
 
usr/config$ bureau -print
 
Bureau line setting relate information
    PSTN number          : 4198 2000 2001 2002
    Hold tone generation : On
    Hot line / Line to Line table
=====================================================
Port     Destination Address      Remote TEL/CHANNEL
-----------------------------------------------------
1               10.150.3.4              300
2               300             300
3               300             300
4               300             300
==========================================================