dialing SIP destination via specific trunk

Hi,

I've a sip destination (user@domain.com) which is only reachable via specific trunk.

I know how to map a SIP extension to SIP destination; but how to make PBX route it via specified trunk? I do not belive outbound routing can be made applicable to SIP destinations.

Thanks!

--igor

RE: dialing SIP destination via specific trunk

Hey, please don't get me wrong, but I believe your statement is a bit confusing.

As far as I know, most SIP URIs don't require any special routing to reach their destination, when they are routed via the default PBXes' SIP Trunk. If your SIP URI is not reaching its' destination, most probably the remote SIP Proxy does not accept inbound OnNet calls, from other SIP Proxies or User Agents.

If a call to your SIP destination requires a registration to the remote SIP Proxy (also known as Trunk), we need to look into this further. Please provide us with additional details regarding this SIP destination (URI).

Have a good day.

RE: dialing SIP destination via specific trunk

That's correct, usually SIP URI's are globally reachable - but it's not always the case.

I'm trying to use SIP <-> Skype gateway, available to Sipnet.ru (Tario) customers. It's possible to call skypeusername@skype.sipnet.ru (or even i believe skypeusername@skype.com) via Sipnet. Yes, it does require SIP registration; and i do not see how i can make PBX to route this URI via Sipnet trunk.

I guess if Sipnet would be my default trunk it may work; i'd test it; but this is not setup i can really use (it would require setting up full routing of numerical prefixes first, leaving Sipnet as last resort default). [LATER] I've tried using Sipnet as default trunk, but PBXes still attempts to reach SIP URI directly - as it should in general case.

I think i'm probably asking to add some support for SIP URI in dial patterns for outbound routing. It can be just regexp, or may be special magic wild-card meaning "matches any SIP URI".

Ideally it should be possible to route different SIP URI's into different trunks, but even global default route for SIP URI's would be helpful for now.

Thanks!

--igor

This post has been edited 2 time(s), it was last edited by sia on 06.09.2009 at 01:08.

RE: dialing SIP destination via specific trunk

Igor,

I now understand what you are trying to achieve, and I will describe a workaround which will allow you to route your Skype calls. But, this not the proper method to achieve routing calls to Skype. There is a proper alternative mentioned in bobmats' last message on this thread asterisk / skype, for which I would like to see a show of hands from you, as well as other users looking to route Skype calls.

In order to achieve routing SIP URI calls to a specific trunk, create a new range of extension numbers (e.g. 5xxxx). The complete SIP URI from each of your contacts, should be placed in the "dial" field, in the form of SIP/skypeusername@skype.sipnet.ru

Then, create an Outbound Route with 5xxxx in its' Custom Dial Patterns and Sipnet's trunk in the Trunk Sequence. Following the last reload, you should be able to dial a 5xxxx extension number and get to the appropriate Skype contact, via the Sipnet trunk.

Obviously, this is not the proper way to achieve this, but it's all we have for now. For a more elegant way, please voice your opinion in the above mentioned thread.