[asterisk-users] Asterisk uses "Anonymous", but why?

[Murthy Gandikota]

----------------------------------------
> From: murthy64 at hotmail.com
> To: asterisk-users at lists.digium.com
> Date: Thu, 6 Aug 2015 17:33:37 +0000
> Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why?
>
>
>
> ________________________________
>> Date: Thu, 6 Aug 2015 12:07:35 -0500
>> From: rmudgett at digium.com
>> To: asterisk-users at lists.digium.com
>> Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why?
>>
>>
>>
>> On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota
>> <murthy64 at hotmail.com<mailto:murthy64 at hotmail.com>> wrote:
>> Tested with X-Lite and it worked fiine. Is there some way to replace
>> "Anonymous" with a config parameter?
>>
>> Thanks for your kind help
>>
>> ----------------------------------------
>>> From: murthy64 at hotmail.com<mailto:murthy64 at hotmail.com>
>>> To: asterisk-users at lists.digium.com<mailto:asterisk-users at lists.digium.com>
>>> Subject: Asterisk uses "Anonymous", but why?
>>> Date: Wed, 5 Aug 2015 21:38:16 +0000
>>>
>>> Hi All
>>>
>>> I am trying to dial out using SIP and Vonage using the instructions :
>>>
>>> <a href="http://www.voip-info.org/wiki/view/Asterisk+and+Vonage"
>> target="_blank"
>> class="newlyinsertedlink">http://www.voip-info.org/wiki/view/Asterisk+and+Vonage</a>
>>>
>>> It was not working. So I downloaded X-PRO Vonage, the vonage sip
>> phone, and wiresharked the port. I see that a significant difference is
>> the vonage phone uses "Vonage User" where
>>> asterisk uses "Anonymous". Is that the problem? The Inbound call
>> works fine. Here is my sip.conf
>>>
>>> [general]
>>> context = demo ; Default context for incoming calls
>>> bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)
>>> bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
>>> srvlookup = yes ; Enable DNS SRV lookups on outbound calls
>>> context=incoming
>>> disallow=all
>>> allow=ulaw
>>> allow=alaw
>>> allow=g729
>>> allow=g723
>>> externip=72.220.28.226
>>> localnet=192.168.0.0
>>> nat=yes
>>> maxexpiry=15
>>> minexpiry=14
>>> ;rtautoclear=no
>>> ;autofallthrough=yes
>>>
>>> register
>> =><did>:<password>@69.59.234.67:5060/202<http://69.59.234.67:5060/202>
>>>
>>> [vonage-out]
>>> username=<did>
>>> type=friend
>>> secret=<password>
>>> port=5061
>>> nat=yes
>>> host=69.59.234.67
>>> fromuser=<did>
>>> fromdomain=69.59.234.67
>>> dtmfmode=rfc2833
>>> auth=md5
>>> context=from-pstn
>>> canreinvite=no
>>>
>>> Here is the CLI command used:
>>>
>>> ubuntu*CLI> originate SIP/732-xxx-xxxx at vonage-out application dial
>>> == Using SIP RTP CoS mark 5
>>> [Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160
>> handle_response_invite: Received response: "Forbidden" from
>> '"Anonymous"
>> <sip:<did>@69.59.234.67<http://69.59.234.67>>;tag=as69898393'
>>> ubuntu*CLI>
>>
>> Use the AMI Originate action or a call file. You can specify a caller
>> id there. You cannot specify one from the command line.
>>
>> Richard
>
>
> Hi Richard
> What should I use for extension? Since I am not bridging an extension with outbound, but making an outbound call and playing a sound file, what would be the extension?
>
> Here is my Asterisk-Java code:
>
> managerConnection.addEventListener(this);
> originateAction = new OriginateAction();
> originateAction.setChannel("SIP/"+ani);
> originateAction.setContext("from-pstn");
> originateAction.setExten(????);
> originateAction.setPriority(new Integer(1));
> originateAction.setCallerId("murthy");
> originateAction.setTimeout(new Integer(30000));
>
> // connect to Asterisk and log in
> managerConnection.login();
>
> // send the originate action and wait for a maximum of 30 seconds for Asterisk
> // to send a reply
> originateResponse = managerConnection.sendAction(originateAction, 30000);
>
> I get error with this.
>
>
> Here is from-pstn context in extensions.ael
>
> context from-pstn {
> 1619xxxxxxx => {
> Answer();
> Playback(welcomesystole);
> Read(digito1,,3);
> Playback(diastole);
> Read(digito2,,3);
> Agi(agi://10.10.22.171:4573/hello.agi?systole=${digito1}&diastole=${digito2});
> Hangup()
> }


I used the "s" for exten, and added extension s to the from-pstn context thus:

 managerConnection.addEventListener(this);
	        originateAction = new OriginateAction();
	        originateAction.setChannel("SIP/"+ani+"@vonage-out");
	        originateAction.setContext("from-pstn");
	        originateAction.setExten("s");
	        originateAction.setPriority(new Integer(1));
	        originateAction.setCallerId("Vonage User");
	        originateAction.setTimeout(new Integer(30000));

	        // connect to Asterisk and log in
	        managerConnection.login();

	        // send the originate action and wait for a maximum of 30 seconds for Asterisk
	        // to send a reply
	        originateResponse = managerConnection.sendAction(originateAction, 30000);

	        // print out whether the originate succeeded or not
	        System.out.println(originateResponse.getResponse());
 
context from-pstn {
        s => {
                 Answer();
                Playback(welcomesystole);
                Read(digito1,,3);
                Playback(diastole);
                Read(digito2,,3);
                Agi(agi://10.10.22.171:4573/hello.agi?systole=${digito1}&diastole=${digito2});
                Hangup();

        }
}

Now I get 

[Aug  6 10:50:32] WARNING[25977][C-0000000b]: chan_sip.c:23160 handle_response_invite: Received response: "Forbidden" from '"Vonage User" <sip:1619xxxxxxx at 69.59.234.67>;tag=as46f9ddef'
ubuntu*CLI> 

Regards