[asterisk-users] Two phone numbers, one SIP provider

Mat Stace mps at colewood.net
Thu Jul 20 08:30:56 MST 2006


I'm not exactly sure on the /how/ * mathes items from the sip.conf (I
suspect it goes to the latter for whichever provider), but the way
configured my extenions.conf to handle multiple incoming accounts from
sipgate is like this (obviously much simplified for ease of explanation):
 
 
[incoming_sipgate]
 
exten => 1111,1,Answer
exten => 1111,2,Dial(SIP/ciscophone,12)
 
exten => 2222,1,Answer
exten => 2222,2,Dial(SIP/pcsoftphone,12)
 
 
 
Also, in the sip.conf, each peer has context=incoming_sipgate in it.
HTH,
 
Mat
 
 -----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Benjamin
Stocker
Sent: 20 July 2006 16:05
To: Asterisk Users Mailing List
Subject: [asterisk-users] Two phone numbers, one SIP provider



Hi

I have two phone numbers from my SIP provider sippro.com, say 1111 and 2222.
I use two sip.conf entries to register this phone numbers:

register => 1111:pass at sippro.com/1111
register => 2222:pass at sippro.com/2222

[1111]
type=friend
username=1111
secret=pass
insecure=very
host=  <http://sip.sippro.com> sip.sippro.com
context=incoming-1111

[2222]
type=friend
username=2222
secret=pass
insecure=very
host=sip.sippro.com
context=incoming-2222

Now, from my dialplan I can use them to do outgoing calls, like
Dial(SIP/9999 at 1111). That works pretty fine. The problem are incoming calls.
According to [1] asterisk should lookup a match in sip.conf when somebody
(outside sippro.com) calls 1111 or 2222. For example, a call to 1111 should
look for a extension in context 'incoming-1111'. A call for 2222 should go
to context incoming-2222. But in the above scenario, asterisk always gets a
match on '2222'. As a result, context 'incoming-2222' is always used. 

How does asterisk search for a match in sip,conf for incoming calls and how
can I get it to use the context specified in the account settings?

1. http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf



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