Failing Calls thru Two Asterisk Boxes Connect Via IAX

Asterisk Asterisk Support
1

Hello Dear Friends,

I’m having troubles with my term project which is related asterisk. I have configured two RaspberryPi’s. Both have asterisk on them and they connected each other via IAX. I have also configured two IP phones and registered each of them to each RaspberryPi’s.

I want to send a SIP call from one IP phone to one RaspberryPi, and then convert it to IAX and send to other RaspberryPi, and then convert it to SIP and send the end IP Phone.

However, when I tried to make a test call, I got 403 Forbidden Error. Could you please take a look if I paste here conf files?

RaspberryPi 1

IAX.Conf

[general]
autokill=yes

register => toronto:welcome@1st_IP_Address

[osaka]
type=friend
host=dynamic
trunk=yes
secret=welcome
context=incoming_osaka

Sip.Conf

[2001]
type=friend
host=dynamic
context=phones
secret=12345

Extensions.conf

[globals]

[general]
autofallthrough=yes

[default]

[incoming_calls]

[phones]
include => internal
include => remote

[internal]
exten => _2XXX,1,NoOp()
exten => _2XXX,n,Dial(SIP/${EXTEN},30)
exten => _2XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => _2XXX,n,Hangup()

[remote]
exten => _1XXX,1,NoOp()
exten => _1XXX,n,Dial(IAX2/osaka/${EXTEN})
exten => _1XXX,n,Hangup()

[osaka_incoming]
include => internal

RaspberryPi 2

IAX.conf

[general]
autokill=yes

register => osaka:welcome@2nd_IP_Address

[toronto]
type=friend
host=dynamic
trunk=yes
secret=welcome
context=incoming_toronto

SIP.conf

[1001]
type=friend
host=dynamic
context=phones

Extensions.conf

[globals]

[general]
autofallthrough=yes

[default]

[incoming_calls]

[phones]
include => internal
include => remote

[internal]
exten => _1XXX,1,NoOp()
exten => _1XXX,n,Dial(SIP/${EXTEN},30)
exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail)
exten => _1XXX,n,Hangup()

[remote]
exten => _2XXX,1,NoOp()
exten => _2XXX,n,Dial(IAX2/toronto/${EXTEN})
exten => _2XXX,n,Hangup()

[toronto_incoming]
include => internal

If anyone has any idea to correct the issue, please advice.

P.S: I can share sip debug if anyone wants to check that too.

Thanks in advance,

2

You should also paste Asterisk logs for a call you made.
However I noticed one thing. You have context=incoming_osaka in iax.conf but I see context osaka_incoming in extensions.conf on RaspberryPi 1. Same on RaspberryPi 2.

–Satish Barot

3

[quote=“satish4asterisk”]You should also paste Asterisk logs for a call you made.
However I noticed one thing. You have context=incoming_osaka in iax.conf but I see context osaka_incoming in extensions.conf on RaspberryPi 1. Same on RaspberryPi 2.

–Satish Barot[/quote]

Hello,

Now I’m able to make successful calls. However there is oneway audio issue. I guess it is related to MODEM, and I will try to find out what is the real issue.

Anyway, thanks for your idea, I guess it was the issue at first.