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sip trunk issue

Discussion in '3CX Phone System - General' started by dlazure, Apr 2, 2014.

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  1. dlazure

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    Hi

    We are sertup on a version 12 with SIP-32G from Yealink latest firmware. using sip trunk, we dial out and after around 10 minutes we no longer hear the other end but they can still hear us.

    this is not happening on each calls. We have now installed a dsl loop from our sip trunk provider and we still have this issue.
    we are connected using a cisco ISA-500 with sip alg disabled.

    any ideas ?
     
  2. craigreilly

    craigreilly Well-Known Member

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    What else does the cisco have setup? Do you have any NAT's setup? Have you run the Firewall tester?

    I have configured SonicWalls and Watchguards and not encountered this issue.
     
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  3. carolinainnovative

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    Make sure that the appropriate ports - port 5060 UDP and the RTP ports - are open and forwarded to your 3CX box. If you are extra concerned about security, you can always restrict the IP range the IPs of your provider's switch.

    If the ISA500 allows it, you may also want to look at the nat timeouts - ie how long it keeps the various translations open after a packet is sent outbound.
     
  4. sigma1

    sigma1 Active Member

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    Is that after 10 minutes, when you make a new call, you have one way audio or after you are on a call for 10 minutes you have this issue?
    Enable keep alives, make sure that ports 9000-9049 are fwded, do you use registration or IP based authentication?
     
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  5. ian.watts

    ian.watts Active Member

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    I've run a trunk through an ISA 500.. was pretty straightforward.
    Port Forwards:
    5000 TCP
    5060 UDP
    5090 UDP
    5090 TCP
    9000-9099 UDP


    Also ran one through an ASA 5505.. ios remains unpretty but worked fine. Didn't have the luxury of ASDM at the time.. but I'm old-school like that anyway. PIX and the Cats ftw.

    A good reference: http://www.3cx.com/blog/docs/ports-used/
     
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