[asterisk-users] Cannot make calls

[Ott Rose]

you can get debug info a couple of ways from the asterisk CLI. I like this command the best. sip set debug ip xxx.xxx.xx.xxx where xxx.xxx.xxx.xxx is the of the x-lite phone. It will give you a lot of info. I haven't figured out how to redirect output yet.

Date: Fri, 30 Oct 2009 13:05:35 -0700
From: cliconnect at cliconnect.com
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Cannot make calls






  




No change on it.



 Do I have to enter a command ?



I have changed the port to 5060 in both clients. Still the same problem.



thanks



Jair Santos



Warren Selby wrote:
You're attempting to connect on ports 5061-5062 but are
bound to port 5060...?

  

What does your CLI look like during a failed call attempt?

  

Thanks,

--Warren Selby

  

  On Fri, Oct 30, 2009 at 2:18 PM, Cliconnect <cliconnect at cliconnect.com>
wrote:

  
    
    

    

    
Thank you,
    

    

>>How are you setting up xlite and the ata?

    

    
Xlite

    

User name : 1000

Domain: IP of the server running Asterisk

Register with domain and receive incoming calls: clear

Port used in local computer : manually specify range : 5061-5062

    

ATA

SIP server address: IP of the server running Asterisk

Outbond Proxy : IP of the server running Asterisk

SIP User id : 1001

Accoount ID : 1001

Use DNS SRV : yes

User id is phone number : yes

SIP registration : no

Local sip port : 5062
    

    

>>Which version of Asterisk are you using?

    
Asterisk 1.6.1.6, Copyright (C) 1999 - 2009 Digium,
    

    

>> What does the general section of your sip.conf look like?

    

    
[general]

context=default

allowoverlap=no

bindport=5060

bindaddr=0.0.0.0

srvlookup=yes

    

When I 

    

sip show peers

    

Name/username              Host            Dyn Nat ACL Port    
Status     

1000                       (Unspecified)    D          5060    
Unmonitored 

1001                       (Unspecified)    D          5060    
Unmonitored 

2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0
offline]

    

regards

    

Jair Santos

    

    

    

    
    

    
    

    

Warren Selby wrote:
    
      
      How are you setting up xlite and the ata?  Which
version
of Asterisk are you using?  What does the general section of your
sip.conf look like?

      

      On Fri, Oct 30, 2009 at 1:01 PM,
Cliconnect <cliconnect at cliconnect.com>
wrote:

      
        Hi all,

        

I can only get a line signal when  I set the phones to not register
with domain . 

        

All phones are in the same NAT and I cannot make calls.

        

I am getting "Call failed : Proxy Authentication Required" in Xlite
and  a busy signal when using an ATA.

        

Here is my extensions.conf

[internal]

exten => 1000,1,Verbose(1|Extension 1000)

;exten => 1000,n,Echo()

;exten => 1000,n,Hangup()

exten => 1000,n,Dial(SIP/1000,30)

exten => 1000,n,Hangup()

        

exten => 1001,1,Verbose(1|Extension 1001)

exten => 1001,n,Dial(SIP/1001,30)

exten => 1001,n,Hangup()

        

[phones]

include => internal

        

        

and sip.conf

[1000]

type=friend

context=phones

host=dynamic

[1001]

type=friend

context=phones

host=dynamic

        

        

I am not setting a password .

        

Any help will be appreciated.

        

TIA

        

Jair Santos

        -- 

        

        
        
        

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