RFC3327 On Wed., Sep. 29, 2021, 12:50 a.m. Micah Quinn, <[email protected]> wrote:
> 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 <[email protected]> on behalf of > Sergiu Pojoga <[email protected]> > *Sent:* Tuesday, September 28, 2021 9:25 AM > *To:* Kamailio (SER) - Users Mailing List <[email protected]> > *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:[email protected] 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 <[email protected]> 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:[email protected] a49a850887 > Avail 85.409 > > Endpoint: 10931/10931 Not in > use 0 of inf > InAuth: 10931-auth/10931 > Aor: 10931 10 > Contact: 10931/sip:[email protected] 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:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 192.168.123.10:5060 > ;rport;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e > From: "10931" <sip:[email protected] > >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54 > To: <sip:[email protected]> > Contact: <sip:[email protected]: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:[email protected]> > 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:[email protected] > >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54 > To: <sip:[email protected]> > 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:[email protected] 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:[email protected] > >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54 > To: <sip:[email protected]> > Contact: <sip:[email protected]: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:[email protected]> > 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:[email protected] > >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54 > To: <sip:[email protected] > >;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:[email protected] SIP/2.0 > Via: SIP/2.0/UDP > 10.252.1.14;branch=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0 > From: "10931" <sip:[email protected] > >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54 > To: <sip:[email protected] > >;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:[email protected] > >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54 > To: <sip:[email protected] > >;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:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 192.168.123.10:5060 > ;rport;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e > From: "10931" <sip:[email protected] > >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54 > To: <sip:[email protected] > >;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:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 192.168.123.10:5060 > ;rport;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e > From: "10931" <sip:[email protected] > >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54 > To: <sip:[email protected]> > Contact: <sip:[email protected]: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:[email protected]> > 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:[email protected] > >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54 > To: <sip:[email protected]> > 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:[email protected] 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:[email protected] > >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54 > To: <sip:[email protected]> > Contact: <sip:[email protected]: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:[email protected]> > 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:[email protected] > >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54 > To: <sip:[email protected] > >;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:[email protected] SIP/2.0 > Via: SIP/2.0/UDP > 10.252.1.14;branch=z9hG4bKe019.4be37ea094ac7d8f4c0a037c7887e071.0 > From: "10931" <sip:[email protected] > >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54 > To: <sip:[email protected] > >;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:[email protected] > >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54 > To: <sip:[email protected] > >;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:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 192.168.123.10:5060 > ;rport;branch=z9hG4bKPj68d21815-beeb-4631-b8ba-e2b979331e0e > From: "10931" <sip:[email protected] > >;tag=a3c6bf40-aa29-4b58-963d-36952a617a54 > To: <sip:[email protected] > >;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 <[email protected]> > *Sent:* Saturday, September 11, 2021 3:16 PM > *To:* Kamailio (SER) - Users Mailing List <[email protected]> > *Cc:* Micah Quinn <[email protected]> > *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 <[email protected]> *On Behalf Of *Micah > Quinn > *Sent:* Friday, September 10, 2021 1:05 AM > *To:* [email protected] > *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 > * [email protected] > 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 > * [email protected] > 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 * [email protected] 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
