________________________________ > Date: Thu, 6 Aug 2015 12:07:35 -0500 > From: rmudg...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > > > > On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota > <murth...@hotmail.com<mailto:murth...@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: murth...@hotmail.com<mailto:murth...@hotmail.com> >> To: asterisk-users@lists.digium.com<mailto:asterisk-users@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<http://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<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 => { Answer(); Playback(welcomesystole); Read(digito1,,3); Playback(diastole); Read(digito2,,3); Agi(agi://10.10.22.171:4573/hello.agi?systole=${digito1}&diastole=${digito2}); Hangup() } -- _____________________________________________________________________ -- 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