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: [email protected] > To: [email protected] > 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@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> > > > > Thanks for your help > murthy > > > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
