[asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

Alex Villací­s Lasso a_villacis at palosanto.com
Fri Apr 25 18:29:44 CDT 2014

I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has been 
heavily modified. Currently asterisk runs on localhost and only listens on SIP/RTP at . Therefore, all of the SIP traffic appears to come from localhost, from the point of view of asterisk.

Currently I have a model on which internal SIP phones get identified by the authentication username, and then the contact names at From: and To: get massaged to incorporate the SIP domain, in order to emulate multiple-domain support. The 'sip' table in 
Asterisk defines all such contacts as SIP accounts of the form name_domain.com, and the SIP phones are configured to use 'name' as authentication username for domain 'domain.com'. However, SIP providers that register on the server with authentication names 
are left with their original names, since in the model, SIP trunks are available to all domains.

Now I have to add support for SIP providers which are to be authorized on the basis of IP only. Apparently, the kamailio module permissions.so (WITH_IPAUTH) is made for just this purpose, so I enabled it. After authentication, I need to route the INVITE to 
asterisk, and asterisk must somehow match the account for the SIP trunk from the available information on the INVITE request.

A typical INVITE for this scenario looks like this, before being processed by kamailio:

INVITE sip:6008010 at;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP;branch=z9hG4bK+676ea13f680e853fd847230512a347561+32e3da76+1
Call-ID: FBE75B3A at 32e3da76
From: <sip:042294440 at;user=phone>;tag=32e3da76+1+544c000c+52be771c
To: <sip:6008010 at;user=phone>
CSeq: 975469826 INVITE
Expires: 180
Organization: SetelGYE
Min-SE: 90
Session-Expires: 18000
Supported: replaces, 100rel, timer
Contact: <sip:042294440 at;transport=udp;user=phone>
Content-Length: 149
Content-Type: application/sdp
Max-Forwards: 70

o=- 0 0 IN IP4
c=IN IP4
t=0 0
m=audio 5388 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Here, 6008010 is the phone number that was dialed at the provider in order to reach my system, and 042294440 is the provider-supplied Caller-ID, which I want to preserve all the way to Asterisk. In particular, 042294440 appears as the value that ends up as 
$fU (From: username) while being processed in kamailio. If I pass the SIP packet as-is to asterisk, asterisk first tries to match by the value of $fU, which obviously fails to match the trunk name. It then tries to match by incoming IP, which also fails 
because asterisk received this packet from . Finally, asterisk sort of matches to the first record in the sip table, which is *not* the SIP account for this trunk, but some other random account.

I have a partial solution that uses sqlops to make a query to the sip table, using the $si (source IP) and reads the trunk name in order to replace $fU. This works, as now $fU will have the trunk name and asterisk will now recognize the intended SIP 
account for the trunk. However, this has the unfortunate side effect of throwing out the Caller-ID information.

What is the standard/proper way to deal with this situation? Is there a well-known way to make Asterisk match the trunk name, without overwriting the Caller-ID information? Before you ask, requesting the provider to modify its INVITEs is not an option. I 
believe there is a standard way to deal with this, since this scenario should also arise with a kamailio that faces the internet, and relays INVITEs (after authentication) to an asterisk in a local network. As far as I can tell, the fact that in my case 
the 'local network' is localhost should be irrelevant.
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