Thank you ,

I did that
sip set debug ip 198.163.0.103
SIP Debugging Enabled for IP: 198.163.0.103

I checked the /var/log/asterisk files and there is no information there.

Could you please inform where am I suppose to see the debug information ?

tks

Jair


Ott Rose wrote:
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: cliconn...@cliconnect.com
To: asterisk-users@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 <cliconn...@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 <cliconn...@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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