How to change codec for each SIP trunk and RTP por

Discussion in 'General' started by azadato, Apr 13, 2010.

  1. azadato

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    Hi all,

    i'm new at this forum.
    i have install my first elastix freePBX system.
    everything going well expet one thing (for now).

    i'm tring to connect SIP trunk to elastix system, as at seems the system was register but i'm not able to get calls or dial out.

    i think that the problem are codec or RTP port.
    So, i would like to know how can i change the RTP ports range and how can i choice diffrents codec for each SIP trunk.

    i don't have any knowledge with linux commands or asterix.

    Please help.

    Thank you.
     
  2. Patrick_elx

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    You need to read Elastix without tears available in the download section.
    You will find the command that you need to access the cli or read the log to understand why your setup is not working.

    regarding codec, you can define it on a trunk by trunk basis by just adding in your setup for the trunk a:
    disallow=all
    allow=alaw,ulaw,g729 etc... or whatever you want.
    Again this is explained in Elastix without Tears.

    For the RTP port, I don't think that you can define it on a trunk by trunk basis, but only for your pbx as a whole.
    Again the answer is in the setup chapter of Elastix without Tears.
     
  3. Patrick_elx

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    Also please do not post the same question in multiple thread.
    Post it once, and wait until someone answer.
    Writing the same post in different areas of this forum will only irritate the same people that are taking on their own time to answer you.

    Also I would suggest that you use the search feature and do some reading before asking.

    A lot of contributors here will be happy to guide you in the right direction for you to find your own answers but wont do all your job. The first step is to read the faq that will tell you to read EWT.
     

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