Spawn extension exit non-zero

Asterisk Asterisk Support
1

Hi all ,
Im new here just started learning asterisk. I have start with reading and doing the lesson out of (asterisk: the definitive guide ) I have installed asterisk on ubuntu server 12.04. and setup up a sip.conf file and extensions.conf file in /etc/asterisk. Im having trouble calling extensions 100 to 101 and 101 to 100 but I can call extension 200 which plays hello world. this is what i get in asterisk -rvvvvv sorry have not learned how to setup logging hoping that comes soon in the book.

== Using SIP RTP CoS mark 5
– Executing [101@LocalSets:1] NoOp(“SIP/0000FFFF0001-0000000e”, “”) in new stack
== Spawn extension (LocalSets, 101, 2) exited non-zero on ‘SIP/0000FFFF0001-0000000e’

I made sure that all my firewalls were off on my macs and its all internal so the router should have no play in this?

thanks for the help!

here is my config.
sip.conf
[general]
context=unauthenticated
allowguest=no
srvlookup=yes
udpbindaddr=0.0.0.0
tcpenable=no

office-phone
type=friend
context=LocalSets
host=dynamic
nat=yes

secret=changed
dtmfmode=auto
disallow=all
allow=ulaw
allow=alaw

;;;;;;;;;;;;;;;;;;;;;
;;Phones;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;

0000FFFF0001 ;Mac Desktop
0000FFFF0002 ;Alienware Laptop
0000FFFF0003 ;Linksys ATA
secret=Changed
allow=gsm

extensions.conf
[LocalSets]

exten => 100,1,NoOp()
same => n,Dail(SIP/0000FFFF0001)

exten => 101,1,NoOp()
same => n,Dail(SIP/0000FFFF0002)

exten => 200,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()

2

The sample configuration enables some logging. Details are set in logger.conf.

Exited non-zero generally just means the caller hung up, although it is difficult for that to happen during Noop.

Dial is spelt i-a, not a-i.

3

so i did your changes thanks. i moved the logger.conf.sample to /etc/asterisk/logger.conf
then did a core reload and i got a new file in /var/log/asterisk/message

this is what was in the file after reload and making a call to extend. 101
[Jan 21 09:07:58] Asterisk 11.7.0 built by root @ ubuntu-asterisk on a x86_64 running Linux on 2013-12-29 15:41:49 UTC
[Jan 21 09:07:58] WARNING[5114] features.c: Could not load features.conf
[Jan 21 09:07:58] ERROR[5114] config_options.c: Unable to load config file ‘udptl.conf’
[Jan 21 09:07:58] WARNING[5114] app_voicemail.c: Failed to load configuration file.
[Jan 21 09:07:58] ERROR[5114] cdr_custom.c: Unable to load cdr_custom.conf. Not logging custom CSV CDRs.
[Jan 21 09:07:58] ERROR[5114] cel_custom.c: Unable to load cel_custom.conf. Not logging CEL to custom CSVs.
[Jan 21 09:07:58] WARNING[5114] config.c: No ‘=’ (equal sign) in line 13 of /etc/asterisk/iax.conf
[Jan 21 09:07:58] NOTICE[5114] iax2-provision.c: No IAX provisioning configuration found, IAX provisioning disabled.
[Jan 21 09:07:58] WARNING[1687] sip/config_parser.c: nat=yes is deprecated, use nat=force_rport,comedia instead
[Jan 21 09:07:58] WARNING[1687] sip/config_parser.c: nat=yes is deprecated, use nat=force_rport,comedia instead
[Jan 21 09:07:58] WARNING[1687] sip/config_parser.c: nat=yes is deprecated, use nat=force_rport,comedia instead
[Jan 21 09:07:58] ERROR[5114] res_config_sqlite3.c: Missing config file ‘res_config_sqlite3.conf’
[Jan 21 09:07:58] NOTICE[5114] res_fax.c: Configuration file ‘res_fax.conf’ not found, not changing options.

cli out put

== Using SIP RTP CoS mark 5
– Executing [101@LocalSets:1] Dial(“SIP/0000FFFF0001-00000011”, “SIP/0000FFFF0002”) in new stack
[Jan 21 09:10:52] WARNING[5439][C-00000017]: app_dial.c:2437 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/0000FFFF0001-00000011’ status is 'CHANUNAVAIL’
ubuntu-asterisk*CLI>

thanks

4

Your peer is not reacheable.

5

This noramally means not registered.

6

thanks for your help, went to the office and got 2 aastra 6731i handset and they work perfect. so its something with x-lite or mac and windows firewall setting ? anyway I can continue thanks again