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

Richard Mudgett rmudgett at digium.com
Thu Aug 6 12:07:35 CDT 2015


On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota <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
> > To: 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
> >
> > [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>;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
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