No audio on remote SIP extension

Discussion in 'General' started by eNoisy, Jul 24, 2010.

  1. eNoisy

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    Hi all!

    I'm having a real headache with an external SIP account on an Aastra phone.

    I managed to connect the telephone to the PBX but still I can't get any audio on both ends. I can even monitor other lines through the programmable keypad. Additionally to that, when I call a mobile the call is connected fine but after a few seconds the it's hanged up and also if I call a land line the connection process takes around 1 minute.

    Not sure if everything is related but troubleshooting the audio issue, I've followed what I've read on some other posts and also on Elastix Without Tears.

    Here's what I've done so far:

    Configured port forwarding on the routers for both sides: ports 5060, 4569 and 10000-20000 from any external IP to my internal Elastix IP (Elastix side) and the Phone IP (Phone side)

    I also included the following lines on the sip_nat.conf file as suggested by pnaves (http://www.elastix.org/es/component/kun ... 606/#38854) and also tried to do it directly on the sip.conf.

    bindport=5060
    bindaddr=0.0.0.0
    nat=yes
    externhost=(IP for the router at the phone side)
    localnet=10.0.0.0/255.255.255.0 (Net address at the Elastix side)
    externrefresh=10
    allowguest=yes
    context=from_internal
    rtptimeout=60
    rtpholdtimeout=120
    useragent=Elastix

    What could I be missing? Do I also need to configure IAX settings on the config files or SIP is enough?

    Thanks!
     
  2. dicko

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    if you have an public ip address, use

    externip=<static.external.quad.ip>

    if you have a dynamically tracked host, use

    externhost=<dynamic.name>

    use one OR the other but whatever you announce must resolve "on the tubes" to your VOIP server.
     
  3. eNoisy

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    Thanks dicko!

    I do have a static public IP. I tried with externip=myIP and once again it didn't work. Right now I'm even trying to connect to another remote machine, whose router's port forwarding is configured as mentioned before and I'm still getting the same results. Still not sure if I'm missing something on either the routers or the asterisk config, or it might just be these Aastra phones. I'll give it a try with another brand and check the results... unless you or someone else has something different to suggest... {:(
     
  4. Moi3020

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    I have been the same problem three times, and always solve it check the rtp ports
    maybe in your router or you firewall, those are filtering ... creates rules that permit the passage of RTP ports 10000 to 20000 in both directions.
     
  5. rgranados

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    Re: Re:No audio on remote SIP extension

    eNoisy escribió:
    I have almost same problem because i have audio only in one way.. let me explain.

    when i call direct to an external number (e. 800117) call is connected but i have no voice path from the Called Party

    -- SIP/CyberSite-Trk-08f76d88 answered SIP/1202-b6c23d18
    -- Native bridging SIP/1202-b6c23d18 and SIP/CyberSite-Trk-08f76d88

    but

    When i call to a extension forwarded to the same party call has complete voice path

    -- SIP/CyberSite-Trk-08fc60d0 answered Local/800117@from-internal-3728,2
    -- Local/800117@from-internal-3728,1 answered SIP/1202-b6c23d18

    I am Using Elastix 1.6
    extensión 1202 is a Linksys PAP2 device

    what is most weird is that if i dial from a Softphone it is working on both manners.
     

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