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

Sergiu Pojoga pojogas at gmail.com
Tue Sep 28 16:25:52 CEST 2021


This says it all:

2021/09/28 04:45:07.358826 192.168.123.10:7330 -> 10.252.1.14:5060
INVITE sip:1093 at 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 <micah.quinn at sipiq.com> 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 at 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 at 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 at 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 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14>
> Contact: <sip:asterisk at 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 at 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 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 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 at 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 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14>
> Contact: <sip:asterisk at 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 at 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 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14
> >;tag=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> 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 at 10.252.1.14 SIP/2.0
> Via: SIP/2.0/UDP
> 10.252.1.14;branch=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> From: "10931" <sip:10931 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14
> >;tag=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> 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 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14
> >;tag=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> 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 at 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 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14
> >;tag=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> 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 at 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 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14>
> Contact: <sip:asterisk at 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 at 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 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 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 at 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 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14>
> Contact: <sip:asterisk at 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 at 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 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14
> >;tag=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> 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 at 10.252.1.14 SIP/2.0
> Via: SIP/2.0/UDP
> 10.252.1.14;branch=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> From: "10931" <sip:10931 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14
> >;tag=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> 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 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14
> >;tag=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> 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 at 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 at 192.168.123.10
> >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54
> To: <sip:1093 at 10.252.1.14
> >;tag=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0
> 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 <hw at skalatan.de>
> *Sent:* Saturday, September 11, 2021 3:16 PM
> *To:* Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>
> *Cc:* Micah Quinn <micah.quinn at sipiq.com>
> *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 <sr-users-bounces at lists.kamailio.org> *On Behalf Of *Micah
> Quinn
> *Sent:* Friday, September 10, 2021 1:05 AM
> *To:* sr-users at 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 at lists.kamailio.org
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