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

Murthy Gandikota murthy64 at hotmail.com
Thu Aug 6 13:25:49 CDT 2015



________________________________
> Date: Thu, 6 Aug 2015 12:55:28 -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 12:33 PM, Murthy Gandikota 
> <murthy64 at hotmail.com<mailto:murthy64 at hotmail.com>> wrote: 
> 
> 
> ________________________________ 
>> Date: Thu, 6 Aug 2015 12:07:35 -0500 
>> From: rmudgett at digium.com<mailto:rmudgett at digium.com> 
>> To: asterisk-users at lists.digium.com<mailto:asterisk-users at lists.digium.com> 
>> Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? 
> 
> <snip> 
> 
>>> 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><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 => { 
> 
> This looks like a dialplan pattern match exten but you do not have a 
> leading '_' to indicate 
> that it is a pattern so this exten will only match a literal "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}<http://10.10.22.171:4573/hello.agi?systole=$%7bdigito1%7d&diastole=$%7bdigito2%7d>); 
> Hangup() 
> } 
> 
> It is up to you where you want to send the originated call to in your 
> dialplan. Since you 
> appear to want to send it to an extension that is a pattern you need to 
> use a value that 
> the pattern will match such as 16190000000. 
> 
> Richard 

Hi Richard

Thank you for your suggestions. The responses received are:

[Aug  6 11:20:28] NOTICE[25977][C-0000001a]: chan_sip.c:23147 handle_response_invite: Failed to authenticate on INVITE to '"Vonage User" <sip:1619xxxxxxx at 69.59.234.67>;tag=as0bf485e8'
       > Channel SIP/vonage202-00000019 was never answered.
  
I don't understand the "Channel SIP/vonage202-00000019 was never answered".... your kind clarification is sought.

Regards

 		 	   		  


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