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
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