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  #1 (permalink)  
Old January 4th, 2009, 03:45 PM
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Smile Addpac AP200D (FXO) + Asterisk: can't call to the PSTN

I have an AP200D (Addpac) of 2 FXO, which I want to use to call outgoing and ingoing calls (Local Area Network [with Asterisk] <-> PSTN)

Now, I can make a call from PSTN to the client of asterisk: 80, but I can't make a call from an extension of Asterisk to the PSTN

These are the configuration files:

sip.conf:

Code:
[general] 
port=5060 
disallow=all 
;allow=g729 
allow=ulaw 
;allow=alaw 
; 
;[80] 
; ...
 
[82] 
; AP200D 
type=friend 
host=dynamic 
context=internal 
language=es 
secret=82 
username=82 
callerid=82 
dtmfmode=rfc2833 
qualify=yes 
nat=no 
insecure=very 
canreinvite=no
extensions.conf
Code:
; Calls to PSTN: 
exten => _XXXXXXX,1,Dial(SIP/82/${EXTEN}) 
exten => _XXXXXXX,2,Hangup 
 
; Calls from PSTN 
exten => 11,1,Goto(ivr,s,1) 
; Here goes an IVR ...
When I want to call to the PSTN, the handset sounds like there were NO extension for that number (2636483).

Name/username Host Dyn Nat ACL Port Status
82/82 192.168.3.15 D 5060 OK (43 ms)
-- Saved useragent "AddPac SIP Gateway" for peer 82

In Asterisk it shows me:

Code:
-- Executing [2636484@internal:1] Dial("SIP/80-081d71f8", "SIP/82/2636483") in new stack 
-- Called 82/2636484 
-- SIP/82-081dc1a8 is circuit-busy 
== Everyone is busy/congested at this time (1:0/1/0) 
-- Executing [2636484@internal:2] Hangup("SIP/80-081d71f8", "") in new stack 
== Spawn extension (internal, 2636484, 2) exited non-zero on 'SIP/80-081d71f8'
When I make a debug in the AP200D, it shows me this:

Received SIP PDU from < 192.168.3.49:5060 >
INVITE sip:2636484@192.168.3.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.49:5060
From: "80" <sip.80@192.168.3.49>
To: <sip:2636483@192.168.3.15>
Contact: <sip:80@192.168.3.49>
....

Sending SIP PDU to < 192.168.3.49:5060 > from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.49:5060
From: "80" <sip.80@192.168.3.49>
To: <sip:2636484@192.168.3.15>
....

Sending SIP PDU to < 192.168.3.49:5060 > from 5060
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.3.49:5060
From: "80" <sip.80@192.168.3.49>
To: <sip:2636484@192.168.3.15>
....

What I should do in order to solve this problem? I need advices!

Thanks in advanced.
PEDRO
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  #2 (permalink)  
Old January 5th, 2009, 09:38 PM
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Default Re: Addpac AP200D (FXO) + Asterisk: can't call to the PSTN

This link help to solve my problem

http://www.voip-info.org/wiki-Asterisk+cisco+FXO

In my case It was almost the same configuration. If someone needs help with this topic about Asterisk and Addpac AP200D feel free to email me:

p.valera@pucp.edu.pe

Pedro
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