[Asterisk-Users] Newbie question - sip.conf incoming contexts

[Marco Mouta]

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 <radamson at routers.com> 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[:authuser]]@host[: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:password at sip_proxy/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:password at sip_proxy/1234
> register => 2346:password at sip_proxy/2345
> register => 2347:password at sip_proxy/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
>
>
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