Can't get X-lite register with my Asterisk Server

Hello,

I have installed AsteriskNow based on CentOS Linux distribution, my problem is about the X-lite software which doesn't get registred with my Asterisk server, X-lite tries to register but it shows "time out" here is my configuration of the files sip.conf & extensions.conf, I really need your help badly, does this configuration wrong ? can you show me how to get things work out? Thank you alot guys.



sip.conf :

[general]
port=5060
bindaddr=192.168.9.231
context=sip
srvlookup=yes
allow=all
localnet=192.168.9.231/255.255.255.0

[100]
type=friend
secret=123456
context=sip
username=100
callerid="Phone1" <100>
host=dynamic
nat=no
dtmfmode=rfc2833
canreinvite=no
allow=all

[200]
type=friend
secret=123456
context=sip
username=200
callerid="Phone2" <200>
host=dynamic
nat=no
dtmfmode=rfc2833
canreinvite=no
allow=all


============================================================


Extensions.conf :

exten => 100,1,Dial(SIP/100,20)
exten => 200,1,Dial(SIP/200,20)
l1rz1mAsked:
Who is Participating?
 
bwilks99Commented:
It's not real hard it is just a lot of questions and very different to your original question and will lead to more questions. One option is to start a new question or break it down to a number of small questions.

Anyway here are bits of code I have put together this is NOT complete or TESTED or SECURED it is only to give you some idea of things you might need. There are many ways to do the same thing (you will need to customize it to suite you). Also check with voxalot for settings they require.

Here is a great free book with all the information you will need.
Asterisk: The Future of Telephony, 2nd Edition
http://downloads.oreilly.com/books/9780596510480.pdf

hope this helps
##### extensions.conf #####
[default]
;Test music on hold
exten => 10,1,Answer()
exten => 10,2,MusicOnHold(default)
exten => 10,3,Hangup()

;Voice Mail
exten => 8500,1,VoiceMailMain(${CALLERID(num)})
exten => 8500,2,Hangup()

;Local extensions that start with 3XX go to voicemail after 7seconds
exten => _3XX,1,Dial(SIP/${EXTEN},7,t)
exten => _3XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
exten => _3XX,n(unavail),Voicemail(${EXTEN},u)
exten => _3XX,n,Hangup()
exten => _3XX,n(busy),VoiceMail(${EXTEN},b)
exten => _3XX,n,Hangup()

exten => _X.,1,Dial(SIP/sip-out/${EXTEN:0},45,r)
exten => _X.,2,Hangup()


##### voicemail.conf #####
[general]
format=wav49|gsm|wav
serveremail=asterisk
attach=yes
minmessage=2
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
emaildateformat=%A, %B %d, %Y at %r
sendvoicemail=yes ; Allow the user to compose and send a voicemail while inside 
;delete=yes		; After notification, the voicemail is deleted from the server. [per-mailbox only]
			;     This is intended for use with users who wish to receive their 
			;     voicemail ONLY by email. Note:  "deletevoicemail" is provided as an
nextaftercmd=yes	; Skips to the next message after hitting 7 or 9 to delete/save current message.
			;     [global option only at this time] 

[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM

[default]
; Define maximum number of messages per folder for a particular context.
;maxmsg=50
300 => ,300,root@localhost
301 => ,301,root@localhost
302 => ,302,root@localhost


###### musiconhold.conf ######
[default]
mode=files
directory=/var/lib/asterisk/moh
random=yes

###### sip.conf ######

register => 9999999:666666666@sip.yourhost.com/9999999

[sip-out]
disallow=all
allow=g729
type=friend
fromuser=9999999
username=9999999
secret=666666666
host=sip.yourhost.com
insecure=very
context=default
qualify=no
nat=yes

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bwilks99Commented:
Hi Here a couple of things to check.

1.Here is a link to asterisk xlite config
http://www.voip-info.org/wiki/view/Asterisk%20phone%20xten%20xlite

2.At the *CLI> do you see any messages?

3. At the *CLI> type "sip show peers" what output do you get?

4. What version of xlite are you using?


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BananaskinCommented:
If you could also post the xlite log as well (which can be enabled on the softphone) this would help to diagnose the issue.
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l1rz1mAuthor Commented:
Hi,

Everything now is ok, i think there was a problem wih X-lite Version 3, but when i used the version 2 it works perfectly after i restarted the Asterisk Server, I have now some questions please :

1. I want to customize the music on hold, i mean i will make my own music on hold ".mp3 file" and i want to include it in my server, so when someone call when busy for exemple, he can hear that music on hold so how can i include it in my configurations files.

2. I want to establish a connection with an external line, it is a voxalot account that i have linked with a US phone number, so i need the code and where to include it in my sip.conf & extensions.conf.

3. I want to add a configuration line, when it rings for 7 seconds, it will be transfered to the voicemail.

4. How to configure the voicemail on asterisk server.

Thank you, i really appreciate your help experts, you're the best.
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l1rz1mAuthor Commented:
no one can answer this? is it hard or what?
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l1rz1mAuthor Commented:
hello, recently, i wanted to add a callcentric line for OutBound Calls, i have 1 problem :

When someone calls my number IPKALL which i redirected to Callcentric, how to configure the Destination incoming Call who will receive this call? I want a number 555 which i already added in my sip file configuration to receive all incoming calls, how can i do this?

Thank you
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bwilks99Commented:
Here are the instructions from callcentric for routing inbound calls with Asterisk.
Let me know if you need any more help.

http://www.callcentric.com/support/device/did_routing
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l1rz1mAuthor Commented:
i have a serious problem, i have Asterisk Server, 1 PC X-lite & 1 Sipura-1000, everything was working perfectly until the last 2 days i got an error 486 Call Failed busy here :

== Spawn extension (sip, 100, 1) exited non-zero on 'SIP/555-00000000'
    -- Executing [555@sip:1] Dial("SIP/100-00000002", "SIP/555") in new stack
    -- Called 555
    -- Got SIP response 486 "Busy Here" back from 192.168.9.9
    -- SIP/555-00000003 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
  == Auto fallthrough, channel 'SIP/100-00000002' status is 'BUSY'

When I Call from X-lite To my Sipura it doesn't ring and it says error 486 busy here

When I call from Sipura connected with a phone adapter to X-lite, it works, i really don't understand what's the problem, here is my files configuration:

Sip.conf

[general]
port=5060
bindaddr=0.0.0.0
context=entrant
srvlookup=yes


[100]
type=friend
secret=123456
context=sip
username=100
callerid="Zoubir" <100>
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
allow=all

[200]
type=friend
context=sip
secret=123456
username=200
callerid="fa9ir" <200>
host=dynamic
nat=yes
dtmfmode=rfc2833
allow=all


[555]
type=friend
context=sip
secret=123456
username=555
callerid="Server" <555>
host=dynamic
nat=yes
dtmfmode=rfc2833
allow=all


[mysipprovider-out]
type=peer
secret=londontown
username=17772794539
host=callcentric.com
fromuser=17772794539
fromdomain=callcentric.com
canreinvite=no
insecure=very
qualify=yes
nat=yes
context=from-mysipprovider


extensions.conf

[general]
static=yes

[sip]
exten => 100,1,Dial(SIP/100)
exten => 200,1,Dial(SIP/200)
exten => 555,1,Dial(SIP/555)

[incoming]
exten => s,1,Set(Var_TO=${SIP_HEADER(TO)})
exten => s,2,GotoIf($["${Var_TO}" = "<sip:17772794539@callcentric.com>"]?extension2,s,1:3)
exten => s,3,GotoIf($["${Var_TO}" = "<sip:17772794539@in.callcentric.com>"]?extension1,s,1:4)
exten => h,4,Macro(hangupcall)

[extension1]
exten => s,1,Dial(SIP/100)

[from-callcentric]
exten => s,1,Goto(incoming,s,1)


Please help and Thank you Experts :)
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l1rz1mAuthor Commented:
thank you
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