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

[Murthy Gandikota]

________________________________
> Date: Thu, 6 Aug 2015 13:33:11 -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 1:25 PM, Murthy Gandikota 
> <murthy64 at hotmail.com<mailto:murthy64 at hotmail.com>> wrote: 
> 
> 
> ________________________________ 
>> Date: Thu, 6 Aug 2015 12:55:28 -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? 
>> 
>> 
>> 
>> On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota 
>> 
> <murthy64 at hotmail.com<mailto:murthy64 at hotmail.com><mailto: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><mailto:rmudgett at digium.com<mailto:rmudgett at digium.com>> 
>>> To: 
> asterisk-users at lists.digium.com<mailto:asterisk-users at lists.digium.com><mailto: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><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><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<mailto:sip%3A1619xxxxxxx 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. 
> 
> What do you think "Failed to authenticate" on the call you just 
> originated means? 
> Your call was rejected and thus the call was never answered. You have an 
> authentication problem. Vonage could not authenticate the call you 
> originated. 
> 
> Richard 


I use the same password for INBOUND and it works fine! Something amiss with Asterisk OUTBOUND 
because I used the same password with X-Lite and X-Pro Vonage soft phones with successful calls.

Regards