You can start with the following:


 # Change URI(s)
        $ru = "sip:" + $rU + "@3.3.3.3";
        $tu = "sip:" + $tU + "@3.3.3.3";
        $fu = "sip:" + $fU + "@2.2.2.2";

        $var(contact_username) = $cU;

        # Remove existing Contact header
        remove_hf("Contact");

        # Insert new Contact header using the stored username
        insert_hf("Contact: <sip:$var(contact_username)@2.2.2.2:5060>\r\n");
        OR 
        # Insert new Contact header using the stored username
        insert_hf("Contact: <sip:+61123123123@2.2.2.2:5060>\r\n");

and then add the remaining modifications if needed as per your upstream carrier requirements. 


Regards, 
Shah Hussain 
From: Markus <universe@truemetal.org>
Sent: Friday, September 22, 2023 8:58 AM
To: sr-users@lists.kamailio.org <sr-users@lists.kamailio.org>
Subject: [SR-Users] Modifying SDP as drop-in replacement for overloaded Asterisk box - looking for help/paid consulting fast
 
Hi list,

I'm trying to use Kamailio 4.4.4 with rtpengine in a self-inflicted
emergency situation (didn't monitor traffic growth properly and now
encountering packet loss during peak times) as a drop-in replacement for
an overloaded Asterisk box in a call-termination-to-upstream-carrier
scenario.

My test scenario is to make a call from a SIP softphone to Asterisk IP
1.1.1.1 -> Kamailio/rtpengine IP 2.2.2.2 -> Upstream carrier 3.3.3.3

sngrep on Kamailio box 2.2.2.2 - the following SDP will not work -
carrier is rejecting it. Carrier is authenticating our calls based on
our IP address 2.2.2.2, no username/pass involved.

2023/09/22 02:06:49.216136 2.2.2.2:5060 -> 3.3.3.3:5060
INVITE sip:+32xxxxxxxx@2.2.2.2;user=phone SIP/2.0
Record-Route: <sip:2.2.2.2;lr>
Via: SIP/2.0/UDP
2.2.2.2;branch=z9hG4bKd9c3.d6fa3abe5d52b827e2054de5573028e0.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK473270e8
Max-Forwards: 69
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 20.0.0
Date: Fri, 22 Sep 2023 00:06:50 GMT
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: <sip:+61xxxxxxxxx@2.2.2.2;user=phone>
Content-Type: application/sdp
Content-Length: 314
X-SIP: 1.1.1.1

v=0
o=root 1093000903 1093000903 IN IP4 1.1.1.1
s=Asterisk PBX 20.0.0
c=IN IP4 2.2.2.2
t=0 0
m=audio 25742 RTP/AVP 8 9 0 101
a=maxptime:150
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:25743
a=ptime:20

I'm comparing this rejected INVITE to a successful INVITE sent by the
original Asterisk box at IP 2.2.2.2 (now Kamailio box) to the carrier
without Kamailio in the path, and these are the differences I noticed,
and probably the things I have to mimick with Kamailio in order to make
it work:

INVITE sip:+32xxxxxxxxx@2.2.2.2;user=phone SIP/2.0
should be
INVITE sip:+32xxxxxxxxx@3.3.3.3;user=phone SIP/2.0

To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
should be
To: <sip:+32xxxxxxxx@3.3.3.3;user=phone>

From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
should be
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@2.2.2.2>;tag=as3d75aadd

Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
should be
Contact: <sip:+61xxxxxxxxx@2.2.2.2:5060>

Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
should be
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@2.2.2.2:5060

o=root 1093000903 1093000903 IN IP4 1.1.1.1
should be
o=root 1093000903 1093000903 IN IP4 2.2.2.2

My kamailio.cfg can be found here: https://pastebin.com/6PKcRjPU

These are the Asterisk boxes I want to originate calls from to Kamailio:

[root@voip30 ~]# kamctl address show
+-----+-----+----------+------+------+-----------+
| id  | grp | ip_addr  | mask | port | tag       |
+-----+-----+----------+------+------+-----------+
| 195 |   1 | 1.1.1.1  |   32 |    0 | voip20.sv |
| 196 |   1 | 1.1.1.2  |   32 |    0 | voip21.sv |
| 197 |   1 | 1.1.1.3  |   32 |    0 | voip22.sv |
| 198 |   1 | 1.1.1.4  |   32 |    0 | voip23.sv |
| 199 |   1 | 1.1.1.5  |   32 |    0 | voip24.sv |
| 200 |   1 | 1.1.1.6  |   32 |    0 | voip25.sv |
| 201 |   1 | 1.1.1.7  |   32 |    0 | voip26.sv |
| 202 |   1 | 1.1.1.8  |   32 |    0 | voip27.sv |
| 203 |   1 | 1.1.1.9  |   32 |    0 | voip28.sv |
+-----+-----+----------+------+------+-----------+

This is the upstream carrier I want Kamailio to proxy calls to:

[root@voip30 ~]# kamctl dispatcher show
dispatcher gateways
+----+-------+------------------+-------+-------+------------+------+
| id | setid | destination      | flags | prio. | attrs      | desc |
+----+-------+------------------+-------+-------+------------+------+
| 12 |     1 | sip:3.3.3.3:5060 |     0 |     0 | weight=100 |      |
+----+-------+------------------+-------+-------+------------+------+
(output manually slightly modified to look properly over E-Mail)

As you might have guessed I'm a Kamailio noob... and don't have the
resources to learn it as fast as I must to avoid further packet loss. If
there's anyone available who can help me to get this done today,
optionally in exchange for money, I'd be grateful.

Thank you!
Markus
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