multiple SIP registrations

Discussion in 'General' started by franklin, Jan 28, 2011.

  1. franklin

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    dicko, You once said that in Asterisk you can only have 1 extension/SIP binding per IP. Is it a problem in Elastix to have 2 separate extensions (ie.1000,1001) on one phone -- one on each line reg? I seem to be having some issues, and I think this is the case.
     
  2. dicko

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    I believe it depends on the phone, some have a little SIP port translation thingy built in, many don't.

    So the safest thing is to use a separate port for each line registered on the same host device.
     
  3. fmvillares

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    hi dick!!!
    yealink phones, polycom phones, cisco phones and grandstream phones support multiple lines registered to the same ports 5060 or anything yopu have cvonfigured for...
    but remember asterisk only support 1 place (ip and port) for each registration line
     
  4. dicko

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    Hey Fernando!

    Not so much Linksys/Sipura :)
     
  5. franklin

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    Hi dicko and fm. Thanks for the replies. I have line reg 1 5060 on reg.1.server.1.port and for the outbound proxy port. When I tried to change reg2 to 5061 for both, reg2 would not register. Changing reg.2.outboundProxy.port="5060" made it reg. So reg2 is talking to the server on 5061 and outbondproxy on 5060. How does this work with the outbound proxy? Is the proxy a global setting that is in my trunk?
     
  6. franklin

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    To be more clear, is it kosher to load up your registrations to the max of the Polycom phone with individual extensions if each extension is on a different port to the server? And why is it different with the proxy? Is there anything I need to do to make the reg talk on a different port for the outboundproxy? Thanks.
     
  7. dicko

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    Make sure the extension port matches the phones port in an extension by extension fashion.

    The proxy would be the server that handles your outbound calls. You probably don't need it, as the SIP sessions will otherwise be negotiated through the registration server (Asterisk)
     
  8. franklin

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    I have 2 regis on and IP335, one on 5060 and one on 5061. Both use 5060 reg.x.outboundproxy.port and both are registered. Ports match in the Extensions.

    What is the difference in the authentication/negotiation between server reg and proxy reg? Asterisk only likes one port/IP. Is this only for the server reg? Am I going to run into problems because each reg is using 5060 for the proxy?

    You say "I don't need it" I understand Asterisk is negotiating over 5060 with my VSP, so why even put anything in reg.x.outboundProxy.port? Is there some kind of auth done by Asterisk with the phone to match up the proxy port? And it has nothing to do with the "binding?"

    Where is the port # set for communicating with your VSP?

    Thanks.
     
  9. dicko

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    You need the registration service to receive calls, if you are registered, you can normally also make calls once registered, if you want to use another service than your Asterisk box to make calls, or don't need to receive calls, then so point your proxy server settings to it, if they accept your request it will go through.

    Asterisk is a Back-to-back User agent so generally you will use the same box for both services, as such leave out the proxy settings, asterisk will handle that for you.

    regards

    dicko
     
  10. franklin

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    Interesting thing: I took both IP addresses out of the reg.x.outboundProxy.address. Only the reg1 on 5060 would register. When I put the address back on on reg.2.outboundProxy.address it registered. Hence, for the back to back to work you have to be on 5060?

    Thanks. I'll see how the multiple regis per phone works out in the field. It's only randomly that using 5060 on mult regis/phone presents a problem.
     
  11. dicko

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    Not really, you just need the two end-points to agree on the SIP registration port. Of course those port connections must be transparent over your network.

    The way Polycoms work is a little opaque, they tend to retain settings from obscure places, maybe you need to 456 them to default and use a good provisioning template (not the Elastix one in this case :) )

    Please explore the sip debug options in Asterisk, you will see almost immediately the failures in negotiation of both registration and INVITEs if and when the occur.

    regards

    dicko
     

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