What I do is the following and keep in mind I only use one register statement with my provider:
exten => 18665551234,1,SetVar(FROM_DID=18665551234) ; exten => 18665551234,2,Goto(from-pstn,s,1) ; exten => 5185551234,1,SetVar(FROM_DID=5185551234) ; exten => 5185551234,2,Goto(custom-callid,s,1) ; On 4/2/06, Marco Mouta <[EMAIL PROTECTED]> wrote: > Hi, > > I'm not an expert, but as far as i know, your incoming calls will > arrive with DID in ${EXTEN} > so the only thing you need is: > > exten => 1234,1,GoTo(context1,1234,1) ; example for context extension > and priority > exten => 2345,1,GoTo(context2,2345,1) > exten => 3456,1,GoTo(context3,3456,1) > > Be sure that you have created context1 context2 and context3 in your > extensions.conf > And in this context1 context2 and context3 you must have handler for > 1234; 2345; and 3456; > > example: > [context1] > exten => 1234,1,Answer() > exten => 1234,2,Playback(vm-goodbye) > exten => 1234,3,Hangup() > > > I didn't test this code, but this is my tip the main idea is that you > need to catch de DID and make a GoTo for the context you want. > > > Best regards, > Marco Mouta > > > On 4/2/06, Rich Adamson <[EMAIL PROTECTED]> wrote: > > Steve Gladden wrote: > > >> What version of asterisk? (been lots of changes happening to the sip > > >> code over the last year) > > > > > > > > > SVN-branch-1.2-r9156 > > > > > >> Have you looked at the sample configs in /usr/src/asterisk/configs? > > > > > > Yes I have and my own configs are pretty much copies of them. > > > They do not detail, do or explain the simple concept that I am > > > trying to accomplish. > > > > > > If they do.... I don't see it. > > > > > > #1 I have more than one incoming SIP account > > > #2 I would like to have them come into the context of > > > my choice when a call comes in. > > > HOW do I do this? > > > > > > currently I have 3 register lines > > > there is no way to specify in a register line > > > some way of making the call start in any other context > > > other than what is specified in the [general] section > > > of sip.conf > > > > > > It seems that somehow maybe if there is a peer tat is somehow > > > matched to the register line (how???) it may work. > > > > > > > > > There may be some crazy way to do this within a peer > > > if so this is the information I am looking for... > > > > > > > > > The examples and descriptions are not at all clear to me.... > > > > > > I have 3 accounts with the same provider.... > > > > > > How do I get incoming calls to come into three different contexts > > > that I will create is the question. > > > > > >>From the example file I see: > > > > > > > > > Asterisk can register as a SIP user agent to a SIP proxy (provider) > > > ; Format for the register statement is: > > > ; register => user[:secret[:[EMAIL PROTECTED]:port][/extension] > > > ; > > > ; If no extension is given, the 's' extension is used. The extension > > > needs to > > > ; be defined in extensions.conf to be able to accept calls from this SIP > > > proxy > > > > > > > > > I actually need to do 3 of these..... > > > > > > ;register => 2345:[EMAIL PROTECTED]/1234 > > > ; > > > ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider > > > ; connect to local extension 1234 in extensions.conf, default context, > > > ; unless you configure a [sip_proxy] section below, and configure a > > > ; context. > > > > > > Ok I have 3 accounts from the same provider.... > > > one [sip_proxy] section just puts me in the same problem boat I'm already > > > in.... using a register line > > > > > > the calls some into the context specified in [general] section of sip.conf > > > > > > I need to somehow differentiate the three SIP 'lines' and give > > > them different contexts to start in. > > > > > > > > > > > > > > > ; Tip 1: Avoid assigning hostname to a sip.conf section like > > > [provider.com] > > > > > > > > > OK sure then how will this associate with my register line that > > > uses provider.com > > > This makes no sense to me... > > > I mean It really makes no sense... > > > Sorry for my confusion. > > > > > > Do I need the register line or do I not need the register line? > > > > > > Why even have a register line if you don't need it and can somehow > > > do this in a peerf, riend or user section..... > > > and if you need the register line ---- the instructions say > > > not to use [provider.com] as the peer, then how the heck do you > > > get that register line to work with an associated [peer]. > > > > > > I need to get a handle on how this works before I go posting my > > > sporatic attempts to get a friend,peer or user to 'register' > > > which is not working. > > > > > > The only way I've been able to get my system to take incoming calls > > > from our sip provider so far is to use register lines and keep > > > the system 'registered' with our provider. > > > > I don't use any sip providers, so be careful with what I say here. > > > > Based on the current sip.conf.sample comments (as of today), it would > > appear you need to do something like this: > > > > register => 2345:[EMAIL PROTECTED]/1234 > > register => 2346:[EMAIL PROTECTED]/2345 > > register => 2347:[EMAIL PROTECTED]/3456 > > > > The above register statements are used to inform your sip provider which > > IP address you are coming from, and calls for each of those three > > accounts (eg, 2345, 2346, and 2347) will be routed to your system. In > > your extensions.conf, you would need something like: > > > > exten => 1234,1,Dial(SIP/3000) > > exten => 2345,1,Dial(SIP/3001) > > exten => 3456,1,Dial(SIP/3002) > > > > Note the comments in the sample config relative to not using a host= > > statement in the type=peer section. Also note the above register > > statements assume the use of three different account names (eg, 2345, > > 2346, and 2347). > > > > As I mentioned above, I don't use any sip providers. But, if I read the > > sample file correctly, the key to the above working is having three > > different account names. > > > > Olle has made several changes to the sip implementation in asterisk over > > the last year or so, so there might be variations of how this is done > > that are asterisk version dependent. He has also posted (several times) > > comments relative to how incoming sip calls match the various > > definitions in sip.conf. > > > > Again, since I don't use sip providers, I'll go from memory to try and > > repeat at least a portion of his posts. Be careful as I don't have any > > recent practical experience on this. It goes something like this: > > > > If you specify a host= statement in sip.conf, incoming calls will match > > the "first" section in sip.conf that includes that statement > > (essentially disregarding username and secret, etc). > > > > If you don't specify a host= statement in sip.conf and you have a > > section that includes a username and secret plus type=peer, it will > > match on username and secret. (That implies that if you have three > > different numbers registered with your sip provider all under one > > username, calls for all three will match the "first" section in sip.conf > > that contains that username and secret.) > > > > Olle has also mentioned the entire type= stuff is going away in favor of > > another sip approach. I don't know where that effort stands or even if > > any of it appears in current code. > > > > Hopefully, some other folks will comment on the above as I'm sure others > > have multiple numbers from a single sip provider working. > > > > Rich > > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users