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

Sun Apr 9 07:28:45 MST 2006

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 <marco.mouta at gmail.com> 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 <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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>
>
> --
> Tom Vile
> Baldwin Technology Solutions, Inc
> Consulting - Web Design - VoIP Telephony
> www.baldwintechsolutions.com
> Phone: 518-631-2855 x205
> Fax:     518-631-2856
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