Installing OpenSIPS & Kamailio is....

[TimGuyUK]

Bloody infuriating. Never mind getting to the fun stuff of trying to get teams going.

I cant find a good howto that gets me to a point of even trying.

Anyone suggest a good howto, or even a appliance/virtual that's more or less ready to go?

I had better success with Kamailio but I still don't think I really got to a point it was good to go.

Tim

 

[ewdpb]

I am sorry to hear that @TimGuyUK but I am also a bit happy for I thought I was the only one having problems with it. I figured I was just stupid and could not get it - which is still possible. I got fed up at some point but I recently found this tutorial. I meant to give it a go but I haven't had the time.

BTW, I feel the same way everytime I need to do something new on freeswitch.

Good luck.

 

[gflow]

Bloody infuriating. Never mind getting to the fun stuff of trying to get teams going.

I cant find a good howto that gets me to a point of even trying.

Anyone suggest a good howto, or even a appliance/virtual that's more or less ready to go?

I had better success with Kamailio but I still don't think I really got to a point it was good to go.

Tim

Follow this guide for the install, it will get your started. I've followed it and it works perfectly:

http://www.powerpbx.org/content/opensips-v30-debian-v10-mariadb-apache-v1


Also you might want to install RTPengine first before installing opensips, here is a good link for that:

https://nickvsnetworking.com/rtpengine-installation-configuration/

When you have all that installed you can start following the Teams OpenSIPS integration guide.

 

[TimGuyUK]

Thank you gentleman, I will get onto it with the powerpbx install guide asap.

@gflow What does RTPengine bring to the install? (Anything specific to Teams integration?)

Tim

 

[TimGuyUK]

I am sorry to hear that @TimGuyUK but I am also a bit happy for I thought I was the only one having problems with it. I figured I was just stupid and could not get it - which is still possible. I got fed up at some point but I recently found this tutorial. I meant to give it a go but I haven't had the time.

BTW, I feel the same way everytime I need to do something new on freeswitch.

Good luck.

@ewdpb Your defo not stupid im sure, I can normally bludgeon my way through install guides, maybe even use a couple and go back and forth but the OpenSIPS & Kamailio guides are a nightmare.

Tim

 

[gflow]

@ewdpb Your defo not stupid im sure, I can normally bludgeon my way through install guides, maybe even use a couple and go back and forth but the OpenSIPS & Kamailio guides are a nightmare.

Tim

At least OpenSIPS has a good book and there are some basic training videos on udemy, Kamailio is harder to get started but they are both so similar if you can use one you can figure the other one out.

For me RTPengine had some more features and was easier to find support for, it seemed as though most in the OpenSIPS community use RTPengine. I was using OpenSIPS on AWS and RTPproxy didnt have the features to set an external IP so I needed to use RTPengine, you may not use it if you have direct internet access.

I can help you get to the same point I was at so keep posting your questions here and hopefully you can get it fully functional and post the full config for the rest of us.

 

[KonradSC]

I used that powerpbx guide as well for OpenSIPS, however I went with Debian 9 b/c of issues I ran into with OpenSSL. I need to go back and see if Debian 10 will work with my updated config, but I know that 9 works.

I have Team working with OpenSIPS & Fusion and didn't need RTPengine. It depends on your environment. I have public IP's for both OpenSIPS and FreeSWITCH.

 

[Instigata]

Hi all.

First post here. Trying to setup an RTP Proxy and kind of in the same boat. I have used the powerpbx guide to setup Opensips on Amazon EC2, 2 questions:

1. Is it necessary to setup RTPEngine first?
2. Any suggested configuration for OpenSips to work with RTP?

Best wishes and thanks. (Apologies if I have to open a new thread!)

 

[robvandenbulk]

Yes install RTPengine, RTPengine can convert SRTP to RTP
Kamailio can control RTPengine, to translate RTP>SRTP SRTP>RTP

 

[Instigata]

Alright - thanks a lot. Would anyone have a working config file for Opensips with RTPEngine please? Thanks again.

 

[robvandenbulk]

I cant help you with OpenSips, but i can show you the way in Kamailio.
Yes the config file is a lot of work, but not only the config file, setting up a server with kamailio , RTPengine , Teams and the config file it a lot of work. And to maintain it you need to make the config your self! so you now whats in it when it not work anymore. But i can help you, its opensource,
Greetings Rob

 

[Instigata]

I cant help you with OpenSips, but i can show you the way in Kamailio.
Yes the config file is a lot of work, but not only the config file, setting up a server with kamailio , RTPengine , Teams and the config file it a lot of work. And to maintain it you need to make the config your self! so you now whats in it when it not work anymore. But i can help you, its opensource,
Greetings Rob

Thanks for the offer. I have setup RTPEngine. However, I am getting no-audio and one-way audio calls. Attached is the opensips.cfg - let me know what I have done wrong pls Thanks for your help.

 

[robvandenbulk]

I don't now how to handle opensips, i am working with kamailio, the config look similar,
Do you working with a MS-teams server and opensips? a lot of thinks are missing in this file.
or is this a test config file?
I can try to help you with kamailio, but opensips i must learn to (i have no time for opensips)

Install Kamailio Devel Version From Git Repository

None

kamailio.org

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skalatan.de

Greetings Rob

 

[Instigata]

Hi again Rob.

Thanks. This is a test config file - we are not using MS-Teams at all. If you can fix for Kamailio, we can try and modify for Opensips. Thanks again.

 

[gflow]

Thanks for the offer. I have setup RTPEngine. However, I am getting no-audio and one-way audio calls. Attached is the opensips.cfg - let me know what I have done wrong pls Thanks for your help.

I kinda got it working with OpenSIPS on Amazon I could call in and out but couldn't transfer or hold resume. Remember there a few settings you need to change when running in the cloud behind NAT, have a read here:

Running OpenSIPS in the Cloud

Cloud computing is a more and more viable option for running and providing SIP services. The question is how compatible are the SIP services with the Cloud environment ? So let’s have a look …

blog.opensips.org

 

[KonradSC]

In order for hold to work properly, you need to strip out the "HOLD" method from the "Allow" header. If Microsoft thinks that you support hold it won't work.

I put this in my route to handle replies from freeswitch. This was the trickiest part to get right.

t_on_reply("handle_allow_methods");

The onreply_route looks like this.

onreply_route[handle_allow_methods] {
        xlog("You hit the onreply_route[handle_allow_methods]...\n");
        if(remove_hf("Allow")) {
                xlog("Removed Allow Header was removed: onreply_route[handle_allow_methods] \r\n");
                if (append_hf("Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY,MONKEY\r\n")) {
                        xlog("Added New Allow Header: onreply_route[handle_allow_methods] \r\n");
                }
        }
}