[asterisk-users] multiple registration to sip trunking provider.

[Andrea Borghi]

a strange problem of multiple sip registrations and peer selection in 
sip.conf is calling for your suggestions!!

let's examine this scenario:

some numbers and passwords hidden with HHHs to protect the guilty :)

I have 3 distinct sip subscriptions with cordiaip.net provider in US. For 
each of these i insert in sip.conf (with peer name differences and relefant 
number/password differences, of course]

-------
register => 1646HHHHH25:HHHHH at soft1.ny.cordiaip.net/1646HHHHH25

[cordiaus1]                              
type=friend                             
secret=HHHHH
username=1646HHHHH25                    
fromuser=1646HHHHH25                    
fromdomain=soft1.ny.cordiaip.net
host=soft1.ny.cordiaip.net
call-limit=5                            
outboundproxy=soft1.ny.cordiaip.net     
disallow=all
allow=gsm
allow=alaw
allow=ulaw
context=DID_cordia
insecure=port
-------

the sip registrations are OK and all seeems fine, BUT
i have difficulties to map the incoming call because * is making mistakes in 
matching the incoming sip INVITE to the relevant peer.

Please note that ALL the peers share the very same host and sip port.

When i make a call to one of the subscribed cordia number, in sip debug i get 
a packes similar to this:

--------
<--- SIP read from 38.98.115.34:5060 --->
INVITE sip:16462487925 at 87.241.44.202 SIP/2.0
Via: SIP/2.0/UDP 38.98.115.34:5060;branch=z9hG4bK22981681-bdb335
To: <sip:1646HHHHH25 at 38.98.115.34>
From: <sip:39347HHHHH22 at 38.98.115.34>;tag=2298168-fdb335
Call-ID: 4926-0-1232058433 at 38.98.115.7
CSeq: 1 INVITE
Contact: <sip:393477135822 at 38.98.115.34:5060;transport=udp>
Server: Sansay-SIP/8.0
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 201

v=0
o=Sansay-SPX 11 11 IN IP4 38.98.115.9
s=Session Controller
c=IN IP4 38.98.115.9
t=0 0
m=audio 15986 RTP/AVP 0 18
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
--------

please note the From: and To: lines, I receive a From: with the caller CID 
(my mobile phone, in this case) and a To: with the sip number the call is 
directed to; this seems OK to me.

I have read the chan_sip.c source file and it seems that
when * receives this invite, it wals through the list of sip 
users/peers/friends to search for the correct entry from which cloning the 
sip parameters for the channel (as moh class, call limits, codecs and such) 
using the host IP as the key (if type=peer) or the caller number (if 
type=user) getting the values from the From: header.

This seems very strange, because the user part of the From: header is 
potentially ANY number and the host part (and not the port, because is is 
always 5060 and there is insecure=port in place) in this scenario is not 
unique due to the 3 peers definitions.

Please keep in mind that if i utilize only one registration i have absolutely 
no problems and can configure * correctly. The problem presents itself ONLY 
with multiple peers with multiple registrations to the same host/port.

I cannot request cordia to forward me the numbers via an unique sip 
registration (sip trunking) because it seems that they don't offer this 
service. (but it may well be that i hadn't asked the right question)

Can anyone suggest how to implement a correct sip trunking for this scenario, 
in which I have the incoming calls of the three registration going in a 
specific context (not the default, see context=DID_cordia in the peer 
definitions) and the outgoing calls going out via a specific user (so i can 
choose at the dialplan level with which number i am presenting myself in 
outgoing calls)

I have spent some days trying various combinations of peers and users 
definitions, going in all cases to crash on the wall of the algorithm * uses 
to select the correct peer for the incoming calls.