Pass Through Registration / Proxy

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I need to create a SIP proxy to be placed in front of a legacy PBX. When a phone registers with the proxy, I would like Asterisk to register with the PBX behind it. (To tell the PBX to send calls to the proxy and then to the SIP phone).

Can I use Asterisk to create a proxy like this? Is there a way to cause the Asterisk to register with another host when it receives a successfully registration?

Thanks!

8 thoughts on - Pass Through Registration / Proxy

  • I’ve been tasked with building the whole thing in just Asterisk (as an exercise). Trying to figure out how/if Asterisk alone can do thi.

    From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] I need to create a SIP proxy to be placed in front of a legacy PBX. When a phone registers with the proxy, I would like Asterisk to register with the PBX behind it. (To tell the PBX to send calls to the proxy and then to the SIP phone).

    Can I use Asterisk to create a proxy like this? Is there a way to cause the Asterisk to register with another host when it receives a successfully registration?

    Thanks!

  • Asterisk is not a proxy, so it doesn’t have functionality like this built in. There may be ways to sort of do such things, like listening to an AMI event for a successful inbound registration, updating configuration, reloading it, and causing an outbound registration to get sent. It’s hackish at best.


    Joshua Colp Digium, Inc. | Senior Software Developer
    445 Jan Davis Drive NW – Huntsville, AL 35806 – US
    Check us out at: http://www.digium.com & http://www.asterisk.org

  • Asterisk is not a proxy. Kamailio is a proxy and can do what you want – no need for Asterisk. At best, you would be using Asterisk as a b2bua between your endpoint(s) and legacy PBX.

    -Barry Flanagan

  • You can, but maybe you should use a sip proxy (like kamailio) for this task instead of a back to back user agent like asterisk.

    You can listen to events triggered on registration to asterisk and with realtime intergration add the register to the PBX (or manipulate sip.conf). This still might be easier to implement compared to (for example) kamailio if you are new to that.

  • OK – I’ll have to rethink how to solve this problem. Maybe I made some assumptions…here’s what I’m trying to accomplish:

    I’ve been given a legacy PBX with SIP capabilities. I need to have SIP
    phones connect to Asterisk (for other features, part of the next step) which passes the calls through to the legacy PBX. And conversely, calls to that extension number on the legacy PBX have to ring the SIP phone connected to Asterisk.

    Maybe proxy is the wrong word I chose. Asterisk is something like a peer to the legacy PBX. I thought about setting up individual SIP accounts on the Asterisk box to connect to the legacy PBX, or maybe a SIP trunk to the legacy PBX (assuming it can route calls through the SIP trunk to a peer to reach a phone). The legacy PBX is a Nortel in case that matters.

    I’m supposed to figure this out and present options but having trouble figuring out if Asterisk would be a peer, or pretend to be many sip agents registering on the legacy Sip pbx, etc. I think I’m stuck at the conceptual level. (Still a beginner in training – but having fun learning Asterisk)

    —–Original Message—

  • One of my first integrations was similar but with a Siemens. Easiest might be a SIP trunk (peer) between Asterisk/Nortel and have different prefixes for the Norted (1xx) and Asterisk (2xx) and route these to the other. The SIP endpoints simply register to Asterisk.

  • The challenge is that calls to extensions on the Nortel have to ring the nortel phone AND the sip phone connected to Asterisk.

    —–Original Message—