---------------------------------------- > From: murth...@hotmail.com > To: asterisk-users@lists.digium.com > Date: Thu, 6 Aug 2015 17:33:37 +0000 > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > > > > ________________________________ >> 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() > }
I used the "s" for exten, and added extension s to the from-pstn context thus: managerConnection.addEventListener(this); originateAction = new OriginateAction(); originateAction.setChannel("SIP/"+ani+"@vonage-out"); originateAction.setContext("from-pstn"); originateAction.setExten("s"); originateAction.setPriority(new Integer(1)); originateAction.setCallerId("Vonage User"); 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); // print out whether the originate succeeded or not System.out.println(originateResponse.getResponse()); context from-pstn { s => { 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(); } } Now I get [Aug 6 10:50:32] WARNING[25977][C-0000000b]: chan_sip.c:23160 handle_response_invite: Received response: "Forbidden" from '"Vonage User" <sip:1619xxxxxxx@69.59.234.67>;tag=as46f9ddef' ubuntu*CLI> Regards -- _____________________________________________________________________ -- 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