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

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?

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

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.

no one can answer this? is it hard or what?

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

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

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 :)

thank you