Kamailio and Siremis with FreePBX

sngrep is a very good tool for a quick look, run it on both boxes and watch them

Wow yeah fantastic little tool! Just installed that on both FreePBX and kamilio both, basically freePBX is transmitting but the kamilio box doesn’t get them. In fact it gets nothing, just a blank sngrep screen.

There isn’t a firewall installed on that box either…

On the instructions for install of dSipRouter I read this:

Install (No Proxy audio (RTP) traffic)

git clone https://github.com/dOpensource/dsiprouter.git
cd dsiprouter
./dsiprouter.sh install

Install (Proxy audio (RTP) traffic)

If you need to proxy RTP traffic then add the -rtpengine parameter. So, the command to install dSIPRouter and the RTPEngine would be

git clone https://github.com/dOpensource/dsiprouter.git
cd dsiprouter
./dsiprouter.sh install -rtpengine

I went with option 1 (No Proxy audio (RTP) traffic) - as I believe that’s only needed with NAT, is that correct?

hmm just tried to add the kamilio server onto a softphone (with wrong details) and sip packets appeared in sngrep which tells me FreePBX isn’t sending them properly?

A Media Proxy is indeed needed if the inbound interface is not the same as the outbound interface, this does not ONLY apply to NAT situations, but also perhaps VPN’s Internal connections etc. also. It is best to avoid it’s use if possible, you want the proxy to “hand of both” SIP and SDP both.

OK I won’t use that then and stick with what I have got.

Not sure why kamailio isn’t receiving what freepbx sends it though? It is receiving other test packets so I’m assuming it’s a config error on FreePBX.

For the outgoing and incoming trunk details, I have this:

type=friend
host=xx.xx.xxx.xx

OK added lots more context to my trunk data, and now I’m seeing lots of traffic in my kamilio box from my FreePBX box = PROGRESS!

Now I’m getting a:
(cause 20 - Subscriber absent)
in FreePBX though…

Outgoing data:
canreinvite=yes
dtmfmode=rfc2833
host=xx.xx.xxx.xx
outboundproxy=xx.xx.xxx.xx
progressinbound=yes
qualify=300
type=peer
disallow=all
allow=ulaw

If you change the type to “friend”, the system will use the same settings for the outbound as well as the inbound trunk. Might save you some typing…

Im still getting

(cause 20 - Subscriber absent)

Im thinking because there is no register string?

Did you search for the problem?

I did: Cause 20 Solutions

Let us know if that helps.

Of course I did, and that offers me no solution.

Go back to your softphone, set it’s ip address up as a PBX in dSIPRoputer, make sure that you have a mappng through outbound global routes to a carrier that is known to work with direct IP routing (trunks that need registration/username/passwords will have to be handled in Kamailio/Siremis) . When you get that working then move on to FreePBX

I managed to get dSIPRouter to send some packets to and from my SIP trunk however it just wouldn’t work properly (even with softphone)

I’m at the point of giving up now.I can’t find anywhere that explains exactly the terminology / couple of fields you need to fill in Siremis in order to create and proxy through a SIP trunk - it looks quite feature rich but the documentation is clearly written by the people that have either developed the system or already have an overwhelming knowledge of how it alll works - not for anyone that actually wants to figure out how to use it.

I’m closer to just setting up another PBX now in order to route basic SIP traffic! :frowning:

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Tweet or reach out to the dsiprouter creator? Maybe he can get you over the last leg?

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It is very interesting to note that dSIPRouter has recently added domain support directly for FusionPBX (read Multi-tenant here on one (or more ) servers) it is seamless with Flowroute which supports SRV records and almost painless with VI and ThinQ which don’t. Still no XML required. If you need real HA and have lots of clients, you might want to explore further . . .

(Still works great with as many FreePBI’s as you want though)

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So you suggest I have a FusionPBX work above all my FreePBX servers, instead of kamailio?

No, FusionPBX is a pbx based on freeswitch, FreePBX a pbx based on asterisk, neither one is a good choice for a router for the other one

my post was just a sideline as dSIProuter came up

Ah OK. Have you ever used FusionPBX? It looks good, but a lot to take in and change compared to FreePBX - plus the community isn’t there as much as here.

Are there any benefits to FreeSwitch over Asterisk?

There are many “pros and cons” articles out there, I use both, Asterisk is easier, freeswitch more complicated, there are reasons for that and there are advantages to both…

Asterisk is fine for all but the lartgest monolithic PBX If you have a few hundred concurrent calls I will wager that freeswitch will be a better match for you., but if you have a need of more than one PBX then freeswitch has a lot of advantages, which become more obvious as the number of PBX’s grow.

FreePBX is a very mature and well supported frontend, FusionPBX is the newer guy on the block but is quite sufficient to manage all but the most complicated freeswitch sytem without resorting to “custom” stuff.

JM2CWAE

(as I was editing anyway, I would also mention another advantage, it is trivial to configure FreeSwitch/FusionPBX to ONLY allow domain based connections which largely precludes most attack vectors, both on SIP and web services which sniff your raw IP services, fail2ban can premptively trap such attempts before the knuckle-draggers escalate their efforts , a side effect is that such a setup allows your users to get a nice VOIP portal for fax/call forwarding/etc./etc. without too much angst or black/whitelisting in iptables)

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