Re: [SR-Users] Kamailio/RTPengine as a proxy for FreePBX/Asterisk...

Thank you for looking at this Sergiu. Yes, you are correct that Asterisk's INVITE to the receiving extension is bouncing between 10.252.1.14 and 192.168.123.10. Based on the above, how do you expect this call to reach the softphone at 10.0.0.142? I suppose that is the question at issue. And you'll have to forgive me for any ignorance on the details of SIP; I'm still learning. - Should I be adding a Via header to the message? - Should I be modifying the Contact header? (BTW, I'm already modifying the contact header to point to 10.252.1.14. If I don't, the endpoint shows offline/unavailable. As I understand it, Contact header has nothing to do with message routing.) - Should I be saving the location of the endpoint in Kamailio and doing a lookup on inbound messages? Given that I've been assured that this is a common use case for Kamailio, I have to admit that all of the howto's and example .cfg files I've read and tried do not seem to completely address this situation. ------------------------------ *From:* sr-users &lt;sr-users-bounces(a)lists.kamailio.org&gt; on behalf of Sergiu Pojoga &lt;pojogas(a)gmail.com&gt; *Sent:* Tuesday, September 28, 2021 9:25 AM *To:* Kamailio (SER) - Users Mailing List &lt;sr-users(a)lists.kamailio.org&gt; *Subject:* Re: [SR-Users] Kamailio/RTPengine as a proxy for FreePBX/Asterisk... This says it all: 2021/09/28 04:45:07.358826 192.168.123.10:7330 -> 10.252.1.14:5060 INVITE sip:1093@10.252.1.14 SIP/2.0 Based on the above, how do you expect this call to reach the softphone at 10.0.0.142? Also, it's pretty easy to see from the provided traces that the call is bounding back and forth between 10.252.1.14 <=> 192.168.123.10, never reaching the softphone. You have a rather long way to making this scenario work, which is good, you'll get to learn a few new things as an ITSP. Good luck. On Tue, Sep 28, 2021 at 1:00 AM Micah Quinn &lt;micah.quinn(a)sipiq.com&gt; wrote: OK, then some more details and some questions. My network configuration is as follows: 10.0.0.142 10.0.0.200 10.252.1.14 10.252.1.1 192.168.123.5 192.168.123.10 [softphone] <--------> [kamailio/rtpengine] <---------VPN---------> [VPN server] <------------------> [FreePBX} There is no NAT'ing involved/enabled. I'm running RTPEngine on the same machine as Kamailio. With my current configuration I can call the PBX directly without issue. (i.e. access my voicemail, IVRs, conference rooms, etc.). However, I can still not make an extension-to-extension call. Asterisk responds to the INVITE with a "401 Unauthorized" message.I have two extensions registered (1093 and 10931): Endpoint: 1093/1093 Not in use 0 of inf InAuth: 1093-auth/1093 Aor: 1093 10 Contact: 1093/sip:1093@10.252.1.14 a49a850887 Avail 85.409 Endpoint: 10931/10931 Not in use 0 of inf InAuth: 10931-auth/10931 Aor: 10931 10 Contact: 10931/sip:10931@10.252.1.14 3690dfd96d Avail 85.225 Below are two packet captures from the Kamailio machine and the Asterisk machine. If more information is needed, I'll be happy to supply the specifics. Thanks to anyone that's willing to take the time to look this over. (Alternatively, if somebody wants to suggest a kamailio.cfg file for my specific use case, I'd be happy to test that on my setup as well.) On the Kamailio machine: --------------------------------------- 2021/09/28 04:45:07.358826 192.168.123.10:7330 -> 10.252.1.14:5060 INVITE sip:1093@10.252.1.14 SIP/2.0 Via: SIP/2.0/UDP 192.168.123.10:5060 ;rport;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e From: "10931" <sip:10931@192.168.123.10 To: <sip:1093@10.252.1.14> Contact: <sip:asterisk@192.168.123.10:5060> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93 CSeq: 13326 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "10931" <sip:10931@192.168.123.10> Max-Forwards: 70 User-Agent: FPBX-16.0.10.27(17.9.4) Content-Type: application/sdp Content-Length: 341 v=0 o=- 585379038 585379038 IN IP4 192.168.123.10 s=Asterisk c=IN IP4 192.168.123.10 t=0 0 m=audio 18074 RTP/AVP 0 8 3 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv 2021/09/28 04:45:07.365188 10.252.1.14:5060 -> 192.168.123.10:7330 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.123.10:5060 ;rport=7330;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e;received=192.168.123.10 From: "10931" <sip:10931@192.168.123.10 To: <sip:1093@10.252.1.14> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93 CSeq: 13326 INVITE Server: kamailio (5.3.2 (x86_64/linux)) Content-Length: 0 2021/09/28 04:45:07.366400 10.252.1.14:5060 -> 192.168.123.10:5060 INVITE sip:1093@10.252.1.14 SIP/2.0 Via: SIP/2.0/UDP 10.252.1.14;branch=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0 Via: SIP/2.0/UDP 192.168.123.10:5060 ;received=192.168.123.10;rport=7330;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e From: "10931" <sip:10931@192.168.123.10 To: <sip:1093@10.252.1.14> Contact: <sip:asterisk@192.168.123.10:5060> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93 CSeq: 13326 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "10931" <sip:10931@192.168.123.10> Max-Forwards: 69 User-Agent: FPBX-16.0.10.27(17.9.4) Content-Type: application/sdp Content-Length: 349 v=0 o=- 585379038 585379038 IN IP4 10.252.1.14 s=Asterisk c=IN IP4 10.252.1.14 t=0 0 m=audio 14618 RTP/AVP 0 8 3 111 9 101 a=maxptime:150 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:14619 a=ptime:20 2021/09/28 04:45:07.409622 192.168.123.10:5060 -> 10.252.1.14:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.252.1.14;rport=19725;received=192.168.123.5;branch=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0 Via: SIP/2.0/UDP 192.168.123.10:5060 ;rport=7330;received=192.168.123.10;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93 From: "10931" <sip:10931@192.168.123.10 To: <sip:1093@10.252.1.14 CSeq: 13326 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1632804307/c98b5b90e7cdc94fd7ab1974b7d3c44b",opaque="6e3e077334bf1910",algorithm=md5,qop="auth" Server: FPBX-16.0.10.27(17.9.4) Content-Length: 0 2021/09/28 04:45:07.412926 10.252.1.14:5060 -> 192.168.123.10:5060 ACK sip:1093@10.252.1.14 SIP/2.0 Via: SIP/2.0/UDP 10.252.1.14;branch=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0 From: "10931" <sip:10931@192.168.123.10 To: <sip:1093@10.252.1.14 Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93 CSeq: 13326 ACK Max-Forwards: 69 Content-Length: 0 2021/09/28 04:45:07.413090 10.252.1.14:5060 -> 192.168.123.10:7330 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.123.10:5060 ;rport=7330;received=192.168.123.10;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93 From: "10931" <sip:10931@192.168.123.10 To: <sip:1093@10.252.1.14 CSeq: 13326 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1632804307/c98b5b90e7cdc94fd7ab1974b7d3c44b",opaque="6e3e077334bf1910",algorithm=md5,qop="auth" Server: FPBX-16.0.10.27(17.9.4) Content-Length: 0 2021/09/28 04:45:07.455640 192.168.123.10:7330 -> 10.252.1.14:5060 ACK sip:1093@10.252.1.14 SIP/2.0 Via: SIP/2.0/UDP 192.168.123.10:5060 ;rport;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e From: "10931" <sip:10931@192.168.123.10 To: <sip:1093@10.252.1.14 Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93 CSeq: 13326 ACK Max-Forwards: 70 User-Agent: FPBX-16.0.10.27(17.9.4) Content-Length: 0 On the FreePBX machine: --------------------------------------- 2021/09/28 04:45:07.342242 192.168.123.10:5060 -> 10.252.1.14:5060 INVITE sip:1093@10.252.1.14 SIP/2.0 Via: SIP/2.0/UDP 192.168.123.10:5060 ;rport;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e From: "10931" <sip:10931@192.168.123.10 To: <sip:1093@10.252.1.14> Contact: <sip:asterisk@192.168.123.10:5060> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93 CSeq: 13326 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "10931" <sip:10931@192.168.123.10> Max-Forwards: 70 User-Agent: FPBX-16.0.10.27(17.9.4) Content-Type: application/sdp Content-Length: 341 v=0 o=- 585379038 585379038 IN IP4 192.168.123.10 s=Asterisk c=IN IP4 192.168.123.10 t=0 0 m=audio 18074 RTP/AVP 0 8 3 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv 2021/09/28 04:45:07.390644 10.252.1.14:5060 -> 192.168.123.10:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.123.10:5060 ;rport=7330;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e;received=192.168.123.10 From: "10931" <sip:10931@192.168.123.10 To: <sip:1093@10.252.1.14> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93 CSeq: 13326 INVITE Server: kamailio (5.3.2 (x86_64/linux)) Content-Length: 0 2021/09/28 04:45:07.392235 192.168.123.5:19725 -> 192.168.123.10:5060 INVITE sip:1093@10.252.1.14 SIP/2.0 Via: SIP/2.0/UDP 10.252.1.14;branch=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0 Via: SIP/2.0/UDP 192.168.123.10:5060 ;received=192.168.123.10;rport=7330;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e From: "10931" <sip:10931@192.168.123.10 To: <sip:1093@10.252.1.14> Contact: <sip:asterisk@192.168.123.10:5060> Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93 CSeq: 13326 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "10931" <sip:10931@192.168.123.10> Max-Forwards: 69 User-Agent: FPBX-16.0.10.27(17.9.4) Content-Type: application/sdp Content-Length: 349 v=0 o=- 585379038 585379038 IN IP4 10.252.1.14 s=Asterisk c=IN IP4 10.252.1.14 t=0 0 m=audio 14618 RTP/AVP 0 8 3 111 9 101 a=maxptime:150 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:14619 a=ptime:20 2021/09/28 04:45:07.393454 192.168.123.10:5060 -> 192.168.123.5:19725 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.252.1.14;rport=19725;received=192.168.123.5;branch=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0 Via: SIP/2.0/UDP 192.168.123.10:5060 ;rport=7330;received=192.168.123.10;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93 From: "10931" <sip:10931@192.168.123.10 To: <sip:1093@10.252.1.14 CSeq: 13326 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1632804307/c98b5b90e7cdc94fd7ab1974b7d3c44b",opaque="6e3e077334bf1910",algorithm=md5,qop="auth" Server: FPBX-16.0.10.27(17.9.4) Content-Length: 0 2021/09/28 04:45:07.438326 192.168.123.5:19725 -> 192.168.123.10:5060 ACK sip:1093@10.252.1.14 SIP/2.0 Via: SIP/2.0/UDP 10.252.1.14;branch=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0 From: "10931" <sip:10931@192.168.123.10 To: <sip:1093@10.252.1.14 Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93 CSeq: 13326 ACK Max-Forwards: 69 Content-Length: 0 2021/09/28 04:45:07.438558 10.252.1.14:5060 -> 192.168.123.10:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.123.10:5060 ;rport=7330;received=192.168.123.10;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93 From: "10931" <sip:10931@192.168.123.10 To: <sip:1093@10.252.1.14 CSeq: 13326 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1632804307/c98b5b90e7cdc94fd7ab1974b7d3c44b",opaque="6e3e077334bf1910",algorithm=md5,qop="auth" Server: FPBX-16.0.10.27(17.9.4) Content-Length: 0 2021/09/28 04:45:07.439339 192.168.123.10:5060 -> 10.252.1.14:5060 ACK sip:1093@10.252.1.14 SIP/2.0 Via: SIP/2.0/UDP 192.168.123.10:5060 ;rport;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e From: "10931" <sip:10931@192.168.123.10 To: <sip:1093@10.252.1.14 Call-ID: 4ec009f0-34c5-4356-bef9-a52b862c7a93 CSeq: 13326 ACK Max-Forwards: 70 User-Agent: FPBX-16.0.10.27(17.9.4) Content-Length: 0 ------------------------------ *From:* Henning Westerholt &lt;hw(a)skalatan.de&gt; *Sent:* Saturday, September 11, 2021 3:16 PM *To:* Kamailio (SER) - Users Mailing List &lt;sr-users(a)lists.kamailio.org&gt; *Cc:* Micah Quinn &lt;micah.quinn(a)sipiq.com&gt; *Subject:* RE: Kamailio/RTPengine as a proxy for FreePBX/Asterisk... Hello Micah, using Kamailio as front-end/balancer for one or more asterisk instance(s) is a classic use case for Kamailio. Have a look to the Asterisk log why you get some authentication request, probably you need to “tell” Asterisk to trust the Kamailio (IPs). Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com *From:* sr-users &lt;sr-users-bounces(a)lists.kamailio.org&gt; *On Behalf Of *Micah Quinn *Sent:* Friday, September 10, 2021 1:05 AM *To:* sr-users(a)lists.kamailio.org *Subject:* [SR-Users] Kamailio/RTPengine as a proxy for FreePBX/Asterisk... Hello all, I'm new to Kamailio, so bear with me as I stumble through this. First, I'll describe what I'm trying to achieve at a high level and then perhaps somebody can advise me on whether Kamailio is a good fit for this solution or not. I'd like to be able to deploy a small appliance type server to our customer's sites that just runs Kamailio and a VPN connection back to our datacenter. At our datacenter, we run virtualized instances of Asterisk for each of our customers. The idea is that Kamailio would act as a transparent proxy through to the Asterisk instance under nominal conditions and as a basic SIP router in the case that the Asterisk instance is unavailable. This degraded functionality would then at least allow extension to extension calling even if the Internet or Asterisk instance is down. I'm currently using dispatcher with a single entry in preparation for a time when we might want to failover to another Asterisk instance. I'm forwarding all REGISTER and INVITE messages to the server chosen from ds_select_dst. Initially this all seems to work as I can register with a softphone and pjsip show endpoints shows my softphone connected. However, when I attempt to call any extension (my own or another) Asterisk responds to the INVITE message with a "401 Unauthorized" message and the typical "The person at extension XXXX is unavailable...". I know that more details might be necessary to troubleshoot this, but I didn't want to include everything in one post and risk cluttering it up with unnecessary information. If anyone can confirm that this is a reasonable way to approach the problem, I can then provide whatever relevant data is necessary to get deeper into it. (I've used sngrep, logging, asterisk cli, etc.) Thanks in advance for any help. __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(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: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(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: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users