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<div class="moz-text-flowed" style="font-family: -moz-fixed;
font-size: 14px;" lang="x-western">I am currently preparing a
kamailio-asterisk combination. The asterisk installation uses
realtime for SIP. The kamailio configuration was based on the
reference at <a class="moz-txt-link-freetext"
href="http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb">http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb</a>
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.
<br>
<br>
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.
<br>
<br>
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.
<br>
<br>
A typical INVITE for this scenario looks like this, before being
processed by kamailio:
<br>
<br>
INVITE <a class="moz-txt-link-abbreviated"
href="mailto:sip:6008010@172.28.161.218:5060;transport=udp;user=phone">sip:6008010@172.28.161.218:5060;transport=udp;user=phone</a>
SIP/2.0
<br>
Via: SIP/2.0/UDP
200.25.144.58:5060;branch=z9hG4bK+676ea13f680e853fd847230512a347561+32e3da76+1<br>
Call-ID: FBE75B3A@32e3da76
<br>
From: <a class="moz-txt-link-rfc2396E"
href="mailto:sip:042294440@200.25.144.58:5060;user=phone"><sip:042294440@200.25.144.58:5060;user=phone></a>;tag=32e3da76+1+544c000c+52be771c
<br>
To: <a class="moz-txt-link-rfc2396E"
href="mailto:sip:6008010@172.28.161.218:5060;user=phone"><sip:6008010@172.28.161.218:5060;user=phone></a>
<br>
CSeq: 975469826 INVITE
<br>
Expires: 180
<br>
Organization: SetelGYE
<br>
Min-SE: 90
<br>
Session-Expires: 18000
<br>
Supported: replaces, 100rel, timer
<br>
Contact: <a class="moz-txt-link-rfc2396E"
href="mailto:sip:042294440@200.25.144.58:5060;transport=udp;user=phone"><sip:042294440@200.25.144.58:5060;transport=udp;user=phone></a>
<br>
Content-Length: 149
<br>
Content-Type: application/sdp
<br>
Max-Forwards: 70
<br>
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, PRACK, UPDATE,
INFO, REFER
<br>
<br>
v=0
<br>
o=- 0 0 IN IP4 201.217.79.3
<br>
s=-
<br>
c=IN IP4 201.217.79.3
<br>
t=0 0
<br>
m=audio 5388 RTP/AVP 8 101
<br>
a=rtpmap:101 telephone-event/8000
<br>
a=fmtp:101 0-15
<br>
<br>
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 <b class="moz-txt-star"><span class="moz-txt-tag">*</span>not<span
class="moz-txt-tag">*</span></b> the SIP account for this
trunk, but some other random account.
<br>
<br>
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.
<br>
<br>
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.
<br>
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