QUESTION Opensips integration

atux_null

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Hi. Looking to integrate Opensips with my PIAF system and i have a few questions. I have seen the following link and i have created 2 systems, one for PIAF and one for OPENSIPS.
i am using the same numbering scheme as in link.
  1. Where do i register?To PIAF or to OPENSIPS?
  2. Where does the trunk resides from my ITSP. It uses a domain name and a username/passwd.
  3. How do i create a link between the 77xx to 7xx numbers so when a call comes to OPENSIPS from the sip client, to be routed to the PIAF and the opposite?
 

dicko

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I don't claim to be an expert in anything but I will offer my comments,

opensips nor openser (now kamailio) make for terrible PBX's . they both make excellent proxies however.

If you have a single pbx , then neither are necessary , but if you have several, then either make good sense as a trunking front-end

Neither are easy to manage if you need a gooey.

dsiprouter is a point and click front end to kamailio that is hard to fault and supports all sorts of PBX's (PBI ?) and all your likely NANP carriers.

It also has built in support for FusionPBX which is itself an excellent multidomain PBX with built in auto-provisioning for most well known phones and working T38 fax.

If you choose that route then you have a choice of registering your extensions to dsiprouter in a pass-through fashion to your PBI or continue to register your extensions directly to the underlying PBI.

So if you do it that way, for about 10 bucks a month plus DID's and call cost you can support several hundred phones on dozens of separate clients, each easily managed (more bucks, more sh*t :) )

For me its a no-brainer and works without problems.
 

atux_null

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Thanks a lot for your reply. my intention is to have more than one PBXs to loadbalance and support multiple users and as a proxy to have an Opensips. If you have any links that could help me on how to setup the whole thing, then it will be ideal.
I have seen FusionPBX and currently i would stay with the Asterisk solution.
Regarding dsiprouter i gave it a try, but i had difficulties to register the ITSPs with domain and username/passwd. Also i never managed to make it send the call to the asterisk boxes.
Prefererably i would go to the Opensips with control panel and Asterisk solution. If you could help me setup Asterisk boxes to the Opensips and make incoming/outgoing calls, i would be more than grateful, please.
 

smarks

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I don't claim to be an expert in anything but I will offer my comments,

opensips nor openser (now kamailio) make for terrible PBX's . they both make excellent proxies however.

If you have a single pbx , then neither are necessary , but if you have several, then either make good sense as a trunking front-end

Neither are easy to manage if you need a gooey.

dsiprouter is a point and click front end to kamailio that is hard to fault and supports all sorts of PBX's (PBI ?) and all your likely NANP carriers.

It also has built in support for FusionPBX which is itself an excellent multidomain PBX with built in auto-provisioning for most well known phones and working T38 fax.

If you choose that route then you have a choice of registering your extensions to dsiprouter in a pass-through fashion to your PBI or continue to register your extensions directly to the underlying PBI.

So if you do it that way, for about 10 bucks a month plus DID's and call cost you can support several hundred phones on dozens of separate clients, each easily managed (more bucks, more sh*t :) )

For me its a no-brainer and works without problems.
Can something like dsiprouter (easily) work on an Amazon/Google/Azure instance or is it better to use something like DigitalOcean where you don't have to deal with NAT? Either with the Asterisk servers also behind NAT or not.
 

billsimon

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Can something like dsiprouter (easily) work on an Amazon/Google/Azure instance or is it better to use something like DigitalOcean where you don't have to deal with NAT? Either with the Asterisk servers also behind NAT or not.
opensips and kamailio work fine with NAT, and you pair them with rtpproxy (old guard) or rtpengine (newer) to relay media, which also can handle NAT conversions. Of course it's not automatic; you have to do it in your routing script.
 
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smarks

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opensips and kamailio work fine with NAT, and you pair them with rtpproxy (old guard) or rtpengine (newer) to relay media, which also can handle NAT conversions. Of course it's not automatic; you have to do it in your routing script.
I could be wrong but it sounds like you are talking about Kamailio in front of NAT. I am talking about putting it behind NAT, which is necessary if you want to use a service like Amazon/Azure/Google.
 

billsimon

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I could be wrong but it sounds like you are talking about Kamailio in front of NAT. I am talking about putting it behind NAT, which is necessary if you want to use a service like Amazon/Azure/Google.
It works both ways.

opensips and kamailio can be put behind the NAT. In the header of the config you specify both the internal and the NAT address as "listeners," like:

listen=udp:10.0.0.1:5060 as 1.2.3.4:5060

and when you engage rtpproxy or rtpengine, you can specify whether you are dealing with an endpoint outside your NAT (so you advertise the public IP), or an endpoint inside the NAT (so you advertise your internal IP).

To deal with other people coming from behind a NAT, there are modules for detecting the situation (specifically, identifying an RFC1918 address in the SIP header or SDP) and using the sender's IP and port number instead of what they put in the SIP.
 

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