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trying to call out with a VOIP provider: no sound +error

Discussion in '3CX Phone System - General' started by johandm_be, Jan 4, 2007.

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  1. johandm_be

    Jan 4, 2007
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    I am trying to make a setup as follows:

    using (free) 3CX Phone system, I have set up 3 internal extensions (100 - 102).
    I have one external VOIP line, it is a Gizmo Project (SIP) account ("dinjafr_be") Currently I only allow outgoing calls.
    I have one outbound rule that says: all outgoing calls use the Gizmo VOIP provider.

    on a 2nd machine I have Gizmo Client running (logged in as "johandm_be") and on a 3rd machine I have 3CX softphone running. It is connected to the 3CX phone system, it is extension 101

    if I try to call my gizmo ("johandm_be" or the actual number number, the result is the same) from 3CX softphone on ext 101, then Gizmo client rings (OK!!), but if I pick up there is no sound (in either direction).
    The connection gets dropped several seconds later, and on the 3CX phone I see : "protocol error, layer 2".

    I verified that I opened ports 9000-9003 on my firewalls (since I did not know what protocol, I did it for all of UDP, TCP and ICMP)

    I've pasted my log from the server below. Any advise would be welcome.


    16:03:27.968 Terminated c6 "Dinja"<sip:101@Piv2200> <sip:017470421506@proxy01.sipphone.com> Call ended
    16:03:21.171 Established c6 "Dinja"<sip:101@Piv2200> <sip:017470421506@proxy01.sipphone.com> Call is established
    16:03:20.171 CallConf::eek:nOutgoingResp: Got response from "17470421506"<sip:17470421506@proxy01.sipphone.com:5060>;tag=00007B621B29EC7C on invite from <sip:dinjafr_be@proxy01.sipphone.com>;tag=3417d52b. Response line: SIP/2.0 200 OK
    16:02:53.968 CallConf::eek:nOutgoingResp: Got response from "17470421506"<sip:17470421506@proxy01.sipphone.com:5060>;tag=8760b34a on invite from <sip:dinjafr_be@proxy01.sipphone.com>;tag=3417d52b. Response line: SIP/2.0 180 Ringing
    16:02:50.437 Calling c6 "Dinja"<sip:101@Piv2200> <sip:017470421506@proxy01.sipphone.com> Send INVITE to [#17470421592 @Gizmo]
    16:02:50.296 Routed c6 "Dinja"<sip:101@Piv2200> <sip:017470421506@proxy01.sipphone.com> From: Ext:101; To: [#17470421592 @Gizmo]
    16:02:50.296 CallConf::findDestination: Found destination [#17470421592 @Gizmo] for caller Ext:101
    16:02:50.265 Registrar::checkAor: Registrar resolved sip:101@Piv2200 as <sip:101@>
    16:02:50.265 Incoming c6 "Dinja"<sip:101@Piv2200> <sip:017470421506@proxy01.sipphone.com> Incoming call from 101
    16:02:50.265 AuthMgr::eek:nAuthSuccess: Extension "Dinja"<sip:101@Piv2200>;tag=21608 has been registered successfully
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