[SR-Users] Re: Modifying SDP as drop-in replacement for overloaded Asterisk box - looking for help/paid consulting fast

You can start with the following:    # Change URI(s)           $ru = "sip:" + $rU + &quot;(a)3.3.3.3&quot;.3.3";           $tu = "sip:" + $tU + &quot;(a)3.3.3.3&quot;.3.3";           $fu = "sip:" + $fU + &quot;(a)2.2.2.2&quot;.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 &lt;universe(a)truemetal.org&gt; *Sent:* Friday, September 22, 2023 8:58 AM *To:* sr-users(a)lists.kamailio.org &lt;sr-users(a)lists.kamailio.org&gt; *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 <https://pastebin.com/6PKcRjPU> <https://pastebin.com/6PKcRjPU <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 __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: