Thanks for the help! What I have gathered mentally so far is that asterisk can't do exactly what I am asking/expecting it to do.
Problem being that I am trying to get multiple inbound contexts from multiple peers ( 3 of them in sip.conf) from one single provider. What happens is that it matches the first peer (for my provider) and never matches the next two that I also want to use. Seems that it will only do a match based on IP Address/Host and not on accountname or incoming phone number. The help I have recieved here has not really addressed the origial question of how to get the calls to come directly into a context from the sip peer itself, however they have pointed out some work-arounds to what asterisk seemingly does not support doing directly. If I am wrong with this conclusion please help me out! I have been able to accomplish what is needed by simply having an initial context that everything comes into (possible security issue) and then immediately issue a Goto() to get the call into the context where it belongs. This 'feels' very hokey and wrong, but it works for now! Thanks for the help! Take care! Steve > 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 > _______________________________________________ --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