Sip user register faild

Asterisk Asterisk Support
1

hello
im a elastix and gui user,and now i1m trying to work on asterisk,so im new user here
i follow the asterisk orelly ebook third edition and now i installed asterisk,dadhi and libpri on a vm machin now i want to creat new user and test it,but it dosnt register by softphone (zoiper and x-lite)
i creat sip.conf and extention.conf in /etc/asterisk
here is my sip.conf configuration (like o`relly ebook):
[general]
context=unauthenticated ; default context for incoming calls
allowguest=no ; disable unauthenticated calls
srvlookup=yes ; enabled DNS SRV record lookup on outbound calls
udpbindaddr=0.0.0.0 ; listen for UDP requests on all interfaces
tcpenable=no ; disable TCP support

office-phone ; create a template for our devices
type=friend ; the channel driver will match on username first, IP second
context=LocalSets ; this is where calls from the device will enter the dialplan
host=dynamic ; the device will register with asterisk
nat=yes ; assume device is behind NAT
; *** NAT stands for Network Address Translation, which allows
; multiple internal devices to share an external IP address.
secret=s3CuR#p@s5 ; a secure password for this device – DON’T USE THIS PASSWORD!
dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically
disallow=all ; reset which voice codecs this device will accept or offer
allow=ulaw ; which audio codecs to accept from, and request to, the device
allow=alaw ; in the order we prefer

; define a device name and use the office-phone template
0000FFFF0001

; define another device name using the same template
0000FFFF0002

and here is my extention.conf
[LocalSets]

exten => 100,1,Dial(SIP/0000FFFF0001) ; Replace 0000FFFF0001 with your device name

exten => 101,1,Dial(SIP/0000FFFF0002) ; Replace 0000FFFF0002 with your device name

;
; These will allow you to dial each of the 4 analog phones configured
; in the previous section.
;
exten => 102,1,Dial(DAHDI/1)
exten => 103,1,Dial(DAHDI/2)
exten => 104,1,Dial(DAHDI/3)
exten => 105,1,Dial(DAHDI/4)

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

i dont know i`m right,please say me 1-in softphone which username and password should i inter? username= 100 and secret = s3CuR#p@s5 or username = 0000FFFF0001 ???
2- i tryed all model above (0000FFFF0001 or 100 ) but it is always in registering mode
i reloded chan_sip.so module also
how can i find the problem ?
thank you for your attention

2

sip set debug on

Also, you mis-spelled extensions. type=peer is more secure. Are the phones really behind a different NAT from the Asterisk?

3

thankyou
for mis-spelled extentions i check and in /etc/asterisk i wrote it true
about type=peer i changed it to peer but dosent work yet about nat=yes im not behint nat but i changed it to ‘no’ but there is no difrence( i have an elastix VM in my pc also,and i check it ,it is also nat=yes and it work nice)
by enable sip debug “sip set debug on” noting appear on screen,i try reloading chan_sip.so but notind do
can you tell me i should register username = 100 or 0000FFFF0001??
and i should say that i have not chan_dahdi module,i have installed dahdi and pri but i tink becuse i have not telephony card i have not chan_dahdi module,it`s true?

4

The first things are common poor choices that don’t affect your problem.

If sip set debug on produces no output, the register is not reaching Asterisk, so you need to look for the problem outside of Asterisk, e.g. firewalls.

5

i have 2 vm 1- centos+asterisk and 2- centos + elastix and elastix work correctly for asterisk also i can ssh and work on it,also i creat another user by copy and past from elastix configuration like this
678
deny=0.0.0.0/0.0.0.0
secret=123
dtmfmode=rfc2833
canreinvite=no
context=office-phone
host=dynamic
type=friend
nat=yes
port=5060
dial=SIP/678
mailbox=678@device
permit=0.0.0.0/0.0.0.0
callerid=device <678>
callcounter=yes
faxdetect=no

and it dose not register on asterisk but when i dial some number it say " the number you has dialed is not …" so i tink problem is by my configuration
it is interesting for me that in that time that it was playing " the number you has dialed … " there was no any promp in asterisk -rvvv console,in elastix server every time a file will play it shows in console
i have alredy my question , i should register whith username = 100 or username = 0000FFFF0001 ???
how can i check my asterisk running true and enough module are installed?

6

Once you have fixed the problem that is stopping the register reaching Asterisk, you should register using the section name in sip.conf.

7

thank you or your replyes
but my problem is still there.
i have creat those usernames in sip.conf in my elastix server,and it work correctly and user can register true
i`m Suspicious to module chan_dahdi.so
becuse in my elastix server ( in same pc VM without any card ) it is result :
elastixCLI> module show like chan_dahdi.so
Module Description Use Count
chan_dahdi.so DAHDI Telephony Driver w/PRI & MFC/R2 0
1 modules loaded
elastixCLI>

but in asterisk is :
asteriskCLI> module show like chan_dahdi.so
Module Description Use Count
0 modules loaded
asteriskCLI>

i tink i have a mistake in installation process or other place.

8

how can i understand that my asterisk work correctly
it hase no any line card and user can not register,so when i open the asterisk -r console there is no data following
now i enable the “sip set debug on” but because it dosent show anything in console when im trying to register i tink asterisk have a problem,how can i sent trafic or test asterisk by ssh or other console when i can`t register user?