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

[Richard Mudgett]

On Thu, Aug 6, 2015 at 1:25 PM, Murthy Gandikota <murthy64 at hotmail.com>
wrote:

>
>
> ________________________________
> > 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.
>

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
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