If I’m not wrong, the spa_ireland username and secret is the subscriber info you had in your Linksys SPA 3102 ATA in Voice->Line 1-> Subscriber Information ( Username and password right ) and context you had used is Ireland_local.
In my case, the global var Noram=${EXTEN:1} ( Coz I’m in north america )
[anaphone]
exten => _NXXNXXXXXX,1,Set(Noram=${EXTEN:1})
exten => _NXXNXXXXXX,2,Dial(SIP/analog_phone/${Noram},60,r);What does the r denote here?
exten => _NXXNXXXXXX,3,Goto(analog_phone${DIALSTATUS},1)
sip.conf
register= analog_phone@10.80.1.2(IP addr of my Ubuntubox where asterisk is installed )/analog_phone
In my Linksys phone adapter configuration Voice->Line1->Subscriber info Display name:Brat
password=phone_pwd
Auth_ID=analog_phone
User_ID=analog_phone
Will the above work out, for me to call from my analog phone(416-234-678) to my softphone (678) already registered with asterisk and working, with dtmf option. In sense, when I call from analog phone and press 416-234-678, an dtmf option saying, “Please enter the extension number you want to reach too” and when I press 678 it would ring the softphone 678 ( if available) and also vice-versa ( a call from softphone(678) to the analog phone (416-234-678)
P.S: I have more than 1 softphone extensions ( 456,786,234)
r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Without this option, Asterisk will generate ring tones automatically where it is appropriate to do so; however, “r” will force Asterisk to generate ring tones, even if it is not appropriate.
yes that all sounds good to me
we have a very similar setup here, with analogue phones around the factory such as door phones to gain access, somebody presses 1 which dials an internal extension which grants them access.
How can I configure system.conf correctly? I didn’t even touch that part? I’m just blank about the analog and asterisk part, I only knew softphone and asterisk.
Here’s my Linksys 3102 status info?
Product Information
Product Name: SPA-3102 Serial Number: FM600J119853
Software Version: 3.3.6(GW) Hardware Version: 1.4.5(a)
MAC Address: 000E08C56438 Client Certificate: Installed
Customization: Open
System Status
Current Time: 1/3/2003 05:48:47 Elapsed Time: 00:14:51
Wan Connection Type: DHCP Current IP: 192.168.0.56
Host Name: SipuraSPA Domain:
Current Netmask: 255.xxx.xxx.x Current Gateway: 192.168.0.1
Primary DNS: xxx.xxx.xxx.xxx
Secondary DNS: xxx.xxx.xxx.xxx
LAN IP Address: 192.xxx.x.x Broadcast Pkts Sent: 0
Broadcast Bytes Sent: 0 Broadcast Pkts Recv: 26
Broadcast Bytes Recv: 2710 Broadcast Pkts Dropped: 0
Broadcast Bytes Dropped: 0
I still didn’t coonect my analog phone to test it.
I tried as given in the link. I hanged this line to suit my requirement (as I mentioned earlier that I have more than 3 softphone under one context called brat) exten => 200,1,Dial(SIP/brat)
How does the ringing and communications takes place?
I’ll brief how I tried, with this below option given in my extensions.conf exten => 100,1,Dial(SIP/ana_phone) ; send these call to the FXS (Line1) port
exten => 200,1,Dial(SIP/brat) ; send these calls to the Xlite softphone
From my x-lite softphone I dialed 100 hoping it would route the call to analog phone.
but in my asterisk CLI I see the message: Call from 123 to extension ‘100’ rejected because extension not found.
I also presumed, that call from my analog phone by pressing 200 would ring and ask me to enter the extension I want to reach too.
I.e. I was able to make a call to my softphone extension 123 from my analog phone by pressing 123 and softphone recieves the call.
Next I want to make calls to cellphone/landlines outside the world( I meant to my home or friend ) through this softphone 123 using PSTN(Attached to the ATA ) line in my office. It gives me an error message “416-234-567” extension is not found.
Here’s my dial plan.
sip.conf
;Asterisk will route outgoing calls to this
[brat]
type = friend
username = brat
host = dynamic
port = 5061
secret = brat_pstn
dtmfmode = rfc2833
nat = no
context = pstn
insecure = very
[123]
type = friend
username = 123
secret = my_pwd
host = dynamic
context = anaphone( context of my analog phone )
)