Hi, I have a local lan 192.168.1.0/24 with a linux SIP server (asterisk 1.6.0.1) and 1 line connected to a FXO card.
The idea is to receive calls using this line (from pstn) and route it to extensions (PAP2 Linksys voip adapters).
Asterisk recognizes the call, plays the company´s hello msg, wait for extension and when we dial extension 101 (user jarbas) I get :
Oct 25 19:11:14] WARNING[10091]: channel.c:3181 ast_channel_make_compatible: No path to translate from SIP/jarbas-081ffad8(256)
to DGV/1(12)
[Oct 25 19:11:14] WARNING[10091]: chan_sip.c:3724 sip_write: Asked to transmit frame type 4, while native formats is 0x100 (g729)(256)
read/write = 0x100 (g729)(256)/0x100 (g729)(256)
– Got SIP response 410 “Gone” back from 192.168.1.205
– SIP/jarbas-081ffad8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel ‘DGV/1’ status is ‘CONGESTION’
[Oct 25 19:11:19] WARNING[32187]: chan_dgv.c:7460 event_handler: EV_ERRORDETECTED (1) data 1
I think my register config or my PAP2 Linksys phone adapters is not correct. Any tips ? Thanks.
IPs
Provider´s SIP server 200.30.30.30 (for example)
Local Sip linux server 192.168.1.2
router 192.168.1.1
jarbas user ATA PAP2 Linksys = dhcp (at the moment is 192.168.1.205)
paula user ATA PAP2 Linksys = dhcp (at the moment is 192.168.1.206)
; SIP Configuration example for Asterisk
[general]
context=default ; Default context for incoming calls
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=192.168.1.2 ; IP address to bind to (0.0.0.0 binds to all)
;
register => guilhermeg:abcd@200.30.30.30
register => jarbas:abcd@192.168.1.2
register => paula:abcd@192.168.1.2
;
[jarbas]
type=friend
username=jarbas
secret=abcd
dtmfmode=RFC2833
qualify=yes
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm
host=192.168.1.205
context=default
cancallforward=yes
[paula]
type=friend
username=paula
secret=abcd
dtmfmode=RFC2833
qualify=yes
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm
host=192.168.1.206
context=default
cancallforward=yes
extension.conf
[default]
exten => 101,1,Dial(SIP/jarbas,20)
exten => 101,2,Background(demo-pleaseleaveamsg)
exten => 101,4,Wait(2)
exten => 101,5,HangUp()
exten => 102,1,Dial(SIP/paula, 20)
exten => 102,3,Background(demo-pleaseleaveamsg)
exten => 102,4,Wait(2)
exten => 102,5,HangUp()
[from-pstn]
exten => s,1,Answer
exten => s,n,Background(XXXX-hello) ; "Thanks for calling XXXX, after the tone dial ext or wait for operator …"
exten => s,n,WaitExten
exten => 101,1,Goto(default,101,1)
exten => 102,1,Goto(default,102,2)