Remote phones cannot answer calls

Discussion in 'General' started by jburgin, Nov 23, 2010.

  1. jburgin

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    I recently switch from an asterisk server to an Elastix server. I have 2 Polycom IP335 phones that work normally when connected to my server. I can call out and they can answer inbound calls. When they are connected remotely they can both dial out and seem to work fine but they cannot answer an inbound call. When I call them they ring but you cannot answer the call. If you pick up the handset they keep ringing. It is like you never picked it up. One of the phones was originally connected to my Asterisk phone and it worked fine.

    Thanks
     
  2. Bob

    Bob

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    jburgin,

    Under Unembedded Freepbx, under tools menu, under Asterisk SIP settings, you will find the following items...

    IP Configuration
    Public IP [ ] Static IP[ ] Dynamic IP[ ]
    External IP
    Local Networks

    my suggestion is use the auto configure button, if your network is simple, otherwise have a read up on this section.

    After you have pressed the submit button, see if they now work.

    If not, you need to look at opening/forwarding the ports on the remote routers to the phone.

    Regards

    Bob
     
  3. jburgin

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    Thanks Karma

    When I went to the Asterisk SIP settings page all of my settings were correct but I had an error message stating that sip_nat.conf overrode the settings made on this screen. I had edited my sip_nat.conf file to match my Asterisk setup with my external IP address. Once I commented the line out the error message went away and everything works now.

    My advice to everyone out there is to limit the file editing you do to the bare minimum and use the configuration screens if at all possible.

    Thanks again

    Jim
     
  4. danardf

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    Hi jburgin and wellcome to our Elastix forum, and hi Bob too. :)

    Try to find your issue in this forum. (Search Forum). Lots of threads about "how to configure a remote extension".

    The Bob's way, seems doesn't configure sip_nat.conf but another file. (same thing, not in the same place). So try to verify if you haven't 2 same parameters in your config. One could disable the other.

    I can do a mistake, but verify this.

    Regards at all. ;)
     
  5. Bob

    Bob

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    Thank you Danardf, great to be here :laugh:

    Danardf, you are correct, it configures the sip_general_additional_conf. The file sip_nat.conf should remain empty, if you put something in here it may conflict with the settings done in Freepbx.

    In the original world of Asterisk, there is general one file called SIP.CONF. Over the years to make it easier for products like Freepbx, and in general to break up what is a very long configuration file, they broke it up and chained all of the files together so to Asterisk, it looks like a SIP.CONF file.

    When freepbx came up with this GUI SIP settings, I was expecting to see them in a file called SIP_NAT_ADDITIONAL.conf, which would have stayed in line with past changes to config files, but alas, they changed their rules. Not a big deal and I supposed we don't want to get to the other end of the spectrum of breaking it down to far.

    Elastix/Freepbx is driving real hard to make it possible on 98% of system setups to do everything via the GUI. For those other 2% that need to make changes directly with the files, the best thing is to learn the structure, so you know that your changes won't be overridden or that your changes will have an impact on the GUI based configurations.

    The biggest problem is that many of the guides out in Internet land are now getting out of date, or are for pure Asterisk systems and do not apply directly to Freepbx based systems (directly meaning you need to understand what is possible to apply in the GUI (e.g. translation into a Freepbx based GUI.

    Danardf, I know you know this stuff :cheer: , just stating it for others that read this post now and in the future...

    Regards

    Bob
     
  6. danardf

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    Yes, of course Bob. ;)

    Have a nice night. :laugh:
     

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