Having some troube please help

Asterisk Asterisk Support
1

I’ve just set up a asterisk box for our small office dialing out works perfect but when i try to call the number associated with the sip registering account the call ends and this is the log that i’m getting:

– Executing [1248*******@from-internal:1] SetCallerID(“SIP/704-08d81558”, “800*******”) in new stack
– Executing [1248*******@from-internal:2] Dial(“SIP/704-08d81558”, “SIP/01/1248**”) in new stack
– Called 01/1248*
– Executing [s@from-sip-external:1] GotoIf(“SIP/******01-08d94c60”, “0 ?2:3”) in new stack
– Goto (from-sip-external,s,3)
– Executing [s@from-sip-external:3] Goto(“SIP/******01-08d94c60”, “timeconditions|s|1”) in new stack
– Goto (timeconditions,s,1)
– Executing [s@timeconditions:1] GotoIfTime(“SIP/****01-08d94c60", "09:00-18:30|mon-thu||?ivr-2|s|1”) in new stack
– Executing [s@timeconditions:2] GotoIfTime(“SIP/****01-08d94c60", "09:00-18:30|fri||?ivr-2|s|1”) in new stack
– Goto (ivr-2,s,1)
– Executing [s@ivr-2:1] GotoIf(“SIP/*******01-08d94c60”, “?4”) in new stack
– Executing [s@ivr-2:2] Answer(“SIP/*******01-08d94c60”, “”) in new stack
– Executing [s@ivr-2:3] Wait(“SIP/*******01-08d94c60”, “1”) in new stack
– SIP/01-08d854c0 answered SIP/704-08d81558
== Spawn extension (ivr-2, s, 4) exited non-zero on ‘SIP/****01-08d94c60’
– Executing [h@ivr-2:1] Hangup("SIP/***01-08d94c60", “”) in new stack
== Spawn extension (ivr-2, h, 1) exited non-zero on 'SIP/01-08d94c60’
== Spawn extension (from-internal, 1248, 2) exited non-zero on 'SIP/704-08d81558’
Really destroying SIP dialog '76203d3760c6863d124842a713863f3b@226..130.’ Method: ACK
Really destroying SIP dialog '32d3ef792936428b51aee9bf3fcd9da6@226..130.’ Method: BYE

Any ideas what could be the problem?

2

Hello, check the firewall, and post your sip.conf here.

Greetings
Juan Sacco
VOIP-Argentina

3

; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn’t, try adding “nat=1” to each peer definition to
; solve translation problems.

[general]

port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=alaw&ulaw&gsm
allow=alaw
allow=g729
dtmfmode=info
relaxdtmf=no
; If you need to answer unauthenticated calls, you should change this
; next line to ‘from-trunk’, rather than ‘from-sip-external’.
; You’ll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf

;register=01:@226.**.130.***

4

Can you post a session sip with your provider without the gotoiftime ?

Thanks

Juan Sacco
VOIP-Argentina