Alex Samad
2009-Jul-15 04:34 UTC
[asterisk-users] how to enable dial to alex@asterisk.blurb.com
Hi The subject line says it all how do I enable this style of call. Pointers to the dns setup and asterisk setup would be great or even search words for google, as I am not sure how to search for this type of request. Alex -- There is no instance of a country having benefited from prolonged warfare -- Sun Tzu - The Art of War -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: Digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090715/7ad32e51/attachment.pgp
Darrick Hartman
2009-Jul-15 05:39 UTC
[asterisk-users] how to enable dial to alex@asterisk.blurb.com
Alex Samad wrote:> Hi > > The subject line says it all how do I enable this style of call. > Pointers to the dns setup and asterisk setup would be great > > > or even search words for google, as I am not sure how to search for this > type of request. > > AlexAlex, Here's a good place to start. http://www.voip-info.org/wiki/view/DNS+SRV Then you would need to enable a few things in /etc/asterisk/sip.conf [general] allowguest=yes context=yourdefaultcontext domain=yourdomain.com Then configure the default context in your extensions.conf file to include routing for your calls. There may be a few more steps, but this should get you going down the right road. Darrick
John A. Sullivan III
2009-Jul-15 05:50 UTC
[asterisk-users] how to enable dial to alex@asterisk.blurb.com
On Wed, 2009-07-15 at 14:34 +1000, Alex Samad wrote:> Hi > > The subject line says it all how do I enable this style of call. > Pointers to the dns setup and asterisk setup would be great > > > or even search words for google, as I am not sure how to search for this > type of request. > > Alex<snip> If I understand what you are seeking, you can try these URIs: http://www.voip-info.org/wiki/view/Asterisk+tips+SIP+URI+Dial http://www.blyon.com/blog/index.php/2009/06/22/p2p-sip-uri-dialing/ However, I found I changed mine substantially. I am very new to Asterisk so if this seems like a silly idea, it probably is and I would appreciate being told so! We generally use numeric extensions - old habits I suppose. We found that the catch-all _. for uri dialing was also catching mis-dialed extensions. That led us to this solution: [dial-uri] ; Always include this last because of its broad matches exten => _[a-zA-Z0-9].,1,GotoIf($[${SIPDOMAIN}!=pbx01.ssiservices.biz]?:_.,1) ; non-URIs will automatically append @pbx01.ssiservices.biz ; this logic separates mistyped extensions from valid URI attempts exten => _[a-zA-Z0-9].,n,Macro(uridial,${EXTEN}@${SIPDOMAIN}) exten => _.,1,Answer(0.5) exten => _.,n,Playback(im-sorry) exten => _.,n,Wait(0.0.5) exten => _.,n,Playback(you-dialed-wrong-number) exten => _.,n,Wait(0.4) exten => _.,n,Playback(vm-goodbye) exten => _.,n,Hangup() Here is the macro: [macro-uridial] exten => s,1,NoOp(Calling remote SIP peer ${ARG1}) exten => s,n,Dial(SIP/${ARG1},60) exten => s,n,Congestion() As I think about it, I wonder if that NoOp should be replace with a Verbose. In any event, I hope this helps. Oh, of course, this is for outbound. For inbound, one creates explicit entries for each SIP URI and map these to the appropriate extensions. For example, for users, we typically map to their email address (which is different than their internal ID; for security purposes, publicly exposed IDs are different from internally used IDs). We also create direct SIP extensions for things like voicemail, office numbers, world headquarters, so that direct SIP calls can be used just like regular calls and enter our calling tree: [a100in] ; direct inbound SIP dialing exten => conference,1,Goto(a100pub,6000,1) exten => someone,1,Goto(a100pub,314,1) exten => helpdesk,1,Goto(a100pub,302,1) exten => someoneelse,1,Goto(a100pub,312,1) exten => mycompany-hq,1,Goto(a100pub,99999,welcome) exten => mycompany-europe,1,Goto(a100pub,99999,welcome) exten => mycompany-us,1,Goto(a100pub,99999,welcome) exten => vmail,1,Goto(a100pub,7000,1) Since we are a secure, multi-tenant environment, we do not place these in the default inbound context for sip. Instead, we only allow designated domains in our sip.conf and specify a separate inbound context for each which lands them into these sip directories, e.g., : autodomain=no domain=pbx01.mycompany.com domain=172.x.y.8 ; define client domains domain=yourcompany.com,a100in domain=theircompany.com,a10in domain=pbx01.theircompany.com allowexternaldomains=yes Hope this helps. If someone sees a better way, please say so. Thanks - John -- John A. Sullivan III Open Source Development Corporation Street Preacher: Are you SAVED?????!!!!!! Educated Skeptic: Saved from WHAT?????!!!!!! Educated Believer: From our selfishness that hurts the ones we love and condemns us to an eternity of hurting each other. http://www.spiritualoutreach.com Christianity that makes sense