I Had the same Problem as you did... I used the following from the list as a template and Setup up my dial Plan Accordingly...
http://lists.digium.com/pipermail/asterisk-users/2004-September/062564.html Hope it helps. Dave Chris wrote: > I haven't gotten to keys yet. >The documentation out there doesn't seem to be very good. > >Chris > > >----- Original Message ----- >From: "Tim Pushor" <[EMAIL PROTECTED]> >To: "Asterisk Users Mailing List - Non-Commercial Discussion" ><[email protected]> >Sent: Thursday, May 05, 2005 4:06 PM >Subject: Re: [Asterisk-Users] Connecting 2 * Together-Pulling hair out > > > > >>Personally, if I owned both boxes and had full control of the dialplan >>on both, I'd stay away from passwords. (but be careful what I say, I'm >>a >>hack) >> >>I have a bunch of boxes connected together via IAX and authenticating >>via RSA. The entries in iax.conf are simple, and dialing across the >>connection is simple (no passwords in the dialplan) (thanks again Rich >>for taking the time). >> >>Tim >> >>Here is a sample of iax.conf entries on machine a: >> >>[machineb] >>type=user >>host=machineb.internal.net >>auth=rsa >>inkeys=machineb >>username=machineb >>context=inbound >> >>[machineb] >>type=peer >>host=machineb.internal.net >>auth=rsa >>outkey=machinea >>username=machinea >> >>And an example dialplan entry to dial an extention on machineb (in the >>inbound context): >> >>exten => 333,1,Dial(IAX2/machineb/333) >> >>And on machinea, the opposite of machineb: >> >>[machinea] >>type=user >>host=machinea.internal.net >>auth=rsa >>inkeys=machinea >>username=machinea >>context=inbound >> >>[machinea] >>type=peer >>host=machinea.internal.net >>auth=rsa >>outkey=machineb >>username=machineb >> >>To generate the keys: >> >>on machinea: >> >>astgenkey -n machinea >>mv machinea.* /var/lib/asterisk/keys >> >>copy machinea.pub to machineb's /var/lib/asterisk/keys >> >>on machineb: >> >>astgenkey -n machineb >>mv machineb.* /var/lib/asterisk/keys >> >>copy machineb.pub to machinea's /var/lib/asterisk/keys >> >> >>Chris wrote: >> >> >> >>> I have something similar. Both of my servers are behind a firewall and NAT. You will need to allow UDP 4569 through the firewall for IAX2. If you have NAT you will need to redirect 4569 to the internal server. >>> >>> I would suggest using AMP and then looking at IAX_ADDITIONAL.CONF to see how it's done. You can modify the IAX.CONf because I don't believe AMP rewrites that file. >>> >>> I think the user and passwords are required. I would suggest using a strong password or someone may decide to make a few phone calls. After this you will need the routing in Extensions.conf to allow calls to be made on this trunk. >>> >>> Asterisk will handle the SIP > IAX. All my clients are SIP and they have no trouble going over a IAX trunk to other SIP devices on the other server. >>> >>>This is what my IAX_ADDITIONAL.CONF looks like >>> >>>SiteA - Dynamic IP >>>-------------- >>>[boxb-peer] >>>username=boxa-user >>>type=peer >>>trunk=yes >>>secret=mypassword >>>host=thehost.dyndns.org >>> >>>[boxb-user] >>>type=user >>>secret=mypassword2 >>>host=thehost.dyndns.org >>>context=from-internal >>> >>>--------------- >>>Site b - Static IP >>>---------------- >>> >>>[boxa-peer] >>>username=boxb-user >>>type=peer >>>trunk=yes >>>secret=mypassword2 >>>host=xxx.xxx.xxx.xxx >>> >>>[boxa-user] >>>type=user >>>secret=mypassword >>>host=xxx.xxx.xxx.xxx >>>context=from-internal >>> >>> >>>Regards, >>> >>>Chris >>> >>> >>>----- Original Message ----- >>>From: "mr. barker" <[EMAIL PROTECTED]> >>>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" >>><[email protected]> >>>Sent: Thursday, May 05, 2005 1:58 PM >>>Subject: RE: [Asterisk-Users] Connecting 2 * Together-Pulling hair >>>out >>> >>> >>> >>> >>> >>> >>>>Yes trying to connect to boxes together. >>>> >>>>One sits outside the internal firewall and is on the inside. >>>> >>>>I am using AMP. However I can just put whatever I need in the >>>>custom.conf sections. >>>>The users agents are SIP .. can SIP call go over a IAX trunk ? if so great. >>>>To create the trunk do I need to use a users name and password ? or ? >>>> >>>>I need to have the *box that is behind the firewall to be able to >>>>place a call out through the *box that has a public ip. >>>> >>>>Thank you >>>> >>>>-----Original Message----- >>>>From: [EMAIL PROTECTED] >>>>[mailto:[EMAIL PROTECTED] On Behalf Of Chris >>>>Sent: Thursday, May 05, 2005 8:20 AM >>>>To: Asterisk Users Mailing List - Non-Commercial Discussion >>>>Subject: Re: [Asterisk-Users] Connecting 2 * Together-Pulling hair >>>>out >>>> >>>> I am not sure what you are trying to do. I have created an IAX2 trunk >>>>between the servers over an internet connection. >>>>Then all you have to do is put in call routing on the trunks to >>>>forward the call to the right place. Are you using AMP or trying to do it manually. >>>>I found everything a little confusing as well, but it is simple now >>>>that I understand it. >>>> >>>> >>>>Chris >>>> >>>>----- Original Message ----- >>>>From: "mr. barker" <[EMAIL PROTECTED]> >>>>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" >>>><[email protected]> >>>>Sent: Thursday, May 05, 2005 4:43 AM >>>>Subject: [Asterisk-Users] Connecting 2 * Together-Pulling hair out >>>> >>>> >>>> >>>> >>>> >>>> >>>>> _____ >>>>> >>>>>Subject: [Asterisk-Users] Connecting 2 * Together-Pulling hair out >>>>> >>>>> >>>>> >>>>>I have read the docs on connecting 2* together but am unsure of a >>>>>few >>>>> >>>>> >>>>> >>>>> >>>>things >>>> >>>> >>>> >>>> >>>>>Do I need a different account for each number that will be called >>>>>from one box to the other ? ie. Do I set up a user account on one >>>>>and then have the other box log into that account when it whats to make a call ? >>>>> >>>>> >>>>> >>>>>I have 2 asterisk boxes and only one of them has the ability to >>>>>access a VoipAccount and PSTN connections.(*box 1). The other holds >>>>>the SIP extensions for the internal SIP users/exten(*box2) >>>>> >>>>>I would like to be able to have the box with the Sip UA(*box2) on >>>>>it to be able to place a call using the box that has the >>>>>VoipAccount and PSTN connection. I am able to make multiple UA >>>>>calls on the VoipAccount and 3 >>>>> >>>>> >>>>> >>>>> >>>>on >>>> >>>> >>>> >>>> >>>>>the PSTN lines (only have 3 lines coming in). I can get it to work >>>>>if I create a user exten on *box1 and map a trunk(which is really >>>>>only an >>>>> >>>>> >>>>> >>>>> >>>>exten) >>>> >>>> >>>> >>>> >>>>>using the user/password login to that exten from *box2. However >>>>>when I >>>>> >>>>> >>>>> >>>>> >>>>try >>>> >>>> >>>> >>>> >>>>>to place a second call when the VOIP line is in use it gives me >>>>>error ( basically saying can't use the trunk because it is in use) >>>>>I would like >>>>> >>>>> >>>>> >>>>> >>>>to >>>> >>>> >>>> >>>> >>>>>be able to have this exten/trunk to be able to use multiple >>>>>connections on it. >>>>> >>>>> >>>>> >>>>>There must be an easier way to do this I am just not sure how. I >>>>>looked >>>>> >>>>> >>>>> >>>>> >>>>at >>>> >>>> >>>> >>>> >>>>>creating IAX trunks but still come up with the Trunk is really an >>>>>Exten name/password . >>>>> >>>>> >>>>> >>>>>Any help would be appreciated. (my brain is boiling eggs) >>>>> >>>>> >>>>> >>>>>Thank you. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>-------------------------------------------------------------------- >>>>-------- >>>>---- >>>> >>>> >>>> >>>> >>>> >>>> >>>>>_______________________________________________ >>>>>Asterisk-Users mailing list >>>>>[email protected] >>>>>http://lists.digium.com/mailman/listinfo/asterisk-users >>>>>To UNSUBSCRIBE or update options visit: >>>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>>> >>>>> >>>>> >>>>> >>>>_______________________________________________ >>>>Asterisk-Users mailing list >>>>[email protected] >>>>http://lists.digium.com/mailman/listinfo/asterisk-users >>>>To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>>>-------------------------------------------------------------------- >>>>---- >>>> >>>>_______________________________________________ >>>>Asterisk-Users mailing list >>>>[email protected] >>>>http://lists.digium.com/mailman/listinfo/asterisk-users >>>>To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>>> >>>> >>_______________________________________________ >>Asterisk-Users mailing list >>[email protected] >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >>---------------------------------------------------------------------- >>-- >> >>_______________________________________________ >>Asterisk-Users mailing list >>[email protected] >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.5 - Release Date: 4/05/2005 _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
