2 May
2014
2 May
'14
6:40 p.m.
El 24/04/14 19:09, Alex Villacís Lasso escribió:
I am currently preparing a kamailio-asterisk
combination. The asterisk installation uses realtime for SIP. The kamailio configuration
(attached) 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 127.0.0.1 . 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 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@172.28.161.218:5060;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP
200.25.144.58:5060;branch=z9hG4bK+676ea13f680e853fd847230512a347561+32e3da76+1
Call-ID: FBE75B3A@32e3da76
From:
<sip:042294440@200.25.144.58:5060;user=phone>;tag=32e3da76+1+544c000c+52be771c
To: <sip:6008010@172.28.161.218:5060;user=phone>
CSeq: 975469826 INVITE
Expires: 180
Organization: SetelGYE
Min-SE: 90
Session-Expires: 18000
Supported: replaces, 100rel, timer
Contact: <sip:042294440@200.25.144.58:5060;transport=udp;user=phone>
Content-Length: 149
Content-Type: application/sdp
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, PRACK, UPDATE, INFO, REFER
v=0
o=- 0 0 IN IP4 201.217.79.3
s=-
c=IN IP4 201.217.79.3
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 127.0.0.1 . 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.
I tried appending a
P-Asserted Identity header to the incoming INVITE before routing it to asterisk, like
this:
#!ifdef WITH_IPAUTH
if((!is_method("REGISTER")) && allow_source_address() &&
$au == "")
{
# Attempt to create a P-Asserted-Identity if none exists, to preserve
# incoming Caller-ID
if (!is_present_hf("P-Asserted-Identity"))
{
append_hf("P-Asserted-Identity: <sip:$fU@$fd>\r\n");
}
# Loading $fU from database using IP
sql_pvquery("elxpbx", "SELECT name FROM sip WHERE host =
'$si' AND sippasswd IS NULL", "$fU");
# source IP allowed
return;
}
#!endif
With tcpdump, I can see that the header is indeed appended to the SIP headers of the
INVITE, but there is no effect in Asterisk. From examination of the Asterisk 11.8.1 source
code, I see that channels/chan_sip.c contains a get_pai() function that is
supposed to process P-Asserted-Identity and extract a caller ID. I am still studying the
code, but I would appreciate help on this issue, to see why my attempt is not working.