Where are examples for openSER/Asterisk integration?

My straw poll found that the community support for the Openser project to the worst for a given open source programme.

'Having examined 100's of posts on the net, far from embracing the spirit of co-operation, it would appear that those in the know are keeping the knowledge to themselves. One of the most requested subjects was Asterisk/Openser integration. We did not find a single complete example, though it is well known that this has been successfully achieved numerous times'


<edit by Dean>See rest of thread</edit>

A quick scan through the openSER forum here shows that to be untrue.

It's not right to expect any support "system" to provide full blow by blow details of how to achieve X. All it's possible to do is to lead the horse to water and deal with specific problems.

I've not yet seen any post or reference anywhere (here or otherwise) with a user detailing their integration of Asterisk to openSER and asking why it isn't working.

dean :

A quick scan through the openSER forum here shows that to be untrue.

I agree that voipuser is beacon in the darkness, but please, show me, as the poll suggests, simply one example of a 'how to' It's not an onerous request surely?

We're not talking of a complex system here, simply how would you front end asterisk with openser in its very basic form.

Spcifically, how are attended transfers handled.
How do you integrate asterisk as a backend

These are questions that have been asked again and again, with almost zero response. I can understand reluctnance to repeat basics (read the manual in other words), surely basic concepts should be covered, and to be frank, they're not.

Whenever I find out something usefull, I document it for the community, this simply doesn't happen with Openser and I might add SER,

Do you have a link to that poll? Knowing the reasons presented there can give the chance to explain and improve. I couldn't find it quickly with google.

To give you answers about "how-to", there are some at:

http://www.voip-info.org/wiki/view/Open ... ingOpenSER

You find there openser and asterisk, freeradius, nat traversal and other examples. The dokuwiki at openser.org site has some other tutorials.

This forum and the mailing lists have good response time and answers, but don't expect that when someone asks "I want to make a platform with authentication, authorization, pstn integration, call hunting, call transfer, redirect on busy, not found, etc, voice mail and audio conferencing, a.s.o) will get it as a step-by-step how-to.

As Dean said, the community helps to solve punctual issues, not to build platforms for something else, where the time to do it is humongous.

... and just on top of the openser page of this forum there are some sticky posts, one of them is a good introductive tutorial of how to install it.

Will get you to the state of having a basic sip platform running.

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simply how would you front end asterisk with openser in its very basic form....please show me.......simply one example of a 'how to'

http://www.voip-info.org/wiki/view/Real ... th+OpenSER

There's also some good information on the architecture of this here http://www.mit.edu/afs/athena/project/s ... scoGW.html which relates to Cisco gateways and not Asterisk (but the overall topology and theory is exactly the same).

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Spcifically, how are attended transfers handled.

That question in itself raises 10 more about specific requirements.

If I had a specific set of requirements it's something I could probably build in a day. Badly, I suspect But I could post up the configuration details here and I bet get some guidance within 48 hours as to how to fix it up.

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basic concepts should be covered, and to be frank, they're not.

I don't agree. The basics are covered well enough in the available documentation. Building a network with specific requirements takes work and effort for the specific use case. That needs to be undertaken by the company with the requirement. If it's a commercial use case, the company requiring it can hire expertise to achieve it. If it's experimental/educational, the user can post up a complete set of (non-working) configuration data and get support to get it working.

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but don't expect that when someone asks "I want to make a platform with authentication, authorization, pstn integration, call hunting, call transfer, redirect on busy, not found, etc, voice mail and audio conferencing, a.s.o) will get it as a step-by-step how-to.

Exactly. No-one here can afford to spend a couple of days building that for the sake of it. It would be unreasonable to expect it of them, and no-one here that has done this already for a commercial organisation could post the configuration for IP reasons (which almost certainly will be owned by the paying party).

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Whenever I find out something usefull, I document it for the community

Thank you. You're not alone - there are a number of people here who do just that, and it's people like you that help develop it into a useful resource.

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Do you have a link to that poll? Knowing the reasons presented there can give the chance to explain and improve.

Ditto. Can you post the link?

Dean,

Feel free to delete this thread if you so wish. The poll in question was a straw poll of peers trying to get to the bottom of Asterisk/Openser integration and the sheer frustration of weeks of reading posts on this subject with no responses. The title of the initial post is misleading and I apologize. However I still assert that the basis of my gripe is valid.

While there are many posts on how OpenSer<->Asterisk is a very powerful combination, nowhere is there even sudo code to document how it can be achieved. I’m not talking about complex systems with load balancing etc, but just a simple example of front ending asterisk with Ser/OpenSer such that calls can be transferred between the two and functions like attended/Blind call transfers and MOH work.

I will gladly pay for someone’s time to write a short document in Sudo Code for the benefit of the community as this is sadly lacking and this question comes up time and time again.

Regards

I don't see a need to delete the thread but I do think the title was misleading and I've edited it.

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sheer frustration of weeks of reading posts on this subject with no responses.

Can you provide links to those posts.

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I will gladly pay for someone’s time to write a short document in Sudo Code

Try a PM to Adam (x-console) or Daniel (Miconda) for a quote. Bear in mind they may be constrained by conflicts of interest if either have already done this work for a paying client.

The best option though I think is for you to post where exactly you got stuck with this configuration, and what you've tried so far. From there I think you'll find the answers you need forthcoming.

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this question comes up time and time again.

Can you provide links to those original questions?

Ok, let me start by outlining what's being attempted here. A lot of work is going into the A2Billing project to make it carrier ready. Clearly Asterisk has limits as to the number of sip transactions it can reliably handle. In addition there is an issue with 302 redirects, in that they are not accounted correctly (A local channel is created with no reference to the initial call leg to the device the invite was sent to). Why do we want to do this? Simply so as to allow a user to redirct calls from their phone to another number and not having to use a web interface.

The Idea is to use Openser as a registration and proxy server that handles phone to phone calls. Calls to PSTN are directed to Asterisk which handles both outbound and inbound calls, with A2B doing the accounting. Accounts are setup within A2B and a view of the account data is generated for Openser. So far so good. However the problem lies with attended call transfers.

I simply for the life of me cannot get them to work. Clearly Asterisk needs to be in the media path to handle MOH while the attended conversation is happening, and then the call needs to be bridged.

Any suggestions in this area would be fantastic

regards

Are you trying to achieve the typical scenario? how far do you get? how does your broken callflow match up against the one I've linked?

http://www.allvoipsolutions.org/mydrupal/node/212