No outbound audio

Discussion in '3CX Phone System - General' started by gscherer, Aug 29, 2009.

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  1. gscherer

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    Im having an issue with not being able to send outbound audio. I only have a skype online number for dialing in.
    It shows in the firewall log tcp port 5090 has translation. I dont know how to resolve that. I have all the correct ports forwarded in my PIX firewall.
    The server log says something about recv only.
    Does anyone know how to fix this?
    Thanks,
    Gary

    Firewall Checker Log-----------------------------------
    3CX Firewall Checker, v1.0. Copyright (C) 3CX Ltd. All rights reserved.

    <12:21:04>: Phase 1, checking servers connection, please wait...
    <12:21:04>: Stun Checker service is reachable. Phase 1 check passed.
    <12:21:04>: Phase 2a, Check Port Forwarding to UDP SIP port, please wait...
    <12:21:09>: UDP SIP Port is set to 5060. Response received correctly with no translation. Phase 2a check passed.
    <12:21:09>: Phase 2b. Check Port Forwarding to TCP SIP port, please wait...
    <12:21:14>: TCP SIP Port is set to 5060. Response received correctly with no translation. Phase 2b check passed.
    <12:21:14>: Phase 3. Check Port Forwarding to TCP Tunnel port, please wait...

    <12:21:18>: TCP TUNNEL Port is set to 5090. Response received WITH TRANSLATION 10543::5090. Phase 3 check passed with WARNINGS. Some functionality will be LIMITED. For more information, please visit http://www.3cx.com/support/firewall-checker.html

    <12:21:18>: Phase 4. Check Port Forwarding to RTP external port range, please wait...
    <12:21:23>: UDP RTP Port 9000. Response received correctly with no translation. Phase 4-01 check passed.
    <12:21:28>: UDP RTP Port 9001. Response received correctly with no translation. Phase 4-02 check passe

    Server Log File--------------------------------------
    12:13:00.197 [CM503008]: Call(1): Call is terminated

    12:13:00.187 [CM503008]: Call(1): Call is terminated

    12:12:25.007 [CM503007]: Call(1): Device joined: sip:11@192.168.1.101:3446;rinstance=c0d9b5962e11a062

    12:12:24.977 [CM503007]: Call(1): Device joined: sip:10000@127.0.0.1:6060

    12:12:23.695 [CM505001]: Ext.11: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CXVoipPhone;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXVoipPhone 3.1.8571.0] Transport: [sip:192.168.1.8:5060]

    12:12:23.695 [CM503002]: Call(1): Alerting sip:11@192.168.1.101:3446;rinstance=c0d9b5962e11a062

    12:12:23.415 [CM503024]: Call(1): Calling RingAll80:11Ext.1113Ext.13@[Dev:sip:11@192.168.1.101:3446;rinstance=c0d9b5962e11a062]

    12:12:23.394 [CM503004]: Call(1): Route 1: RingAll80:11Ext.1113Ext.13@[Dev:sip:11@192.168.1.101:3446;rinstance=c0d9b5962e11a062]

    12:12:23.374 [CM503010]: Making route(s) to <sip:80@192.168.1.8:5060>

    12:12:23.364 [CM505002]: Gateway:[Skypegw] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXSkypeGateway 2.0.8742.0] Transport: [sip:127.0.0.1:5060]

    12:12:23.324 [CM503001]: Call(1): Incoming call from +16318316983@(Ln.10000@Skypegw) to <sip:80@192.168.1.8:5060>

    12:12:23.294 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10000 forwards to DN:80

    12:09:46.058 [CM504001]: Ext.11: new contact is registered. Contact(s): [sip:11@192.168.1.101:3446;rinstance=c0d9b5962e11a062/11]

    12:08:26.484 [CM504001]: Ext.99: new contact is registered. Contact(s): [sip:99@127.0.0.1:40600;rinstance=5735744cbf040f0c/99]

    12:08:20.816 IP(s) added:[192.168.1.8]

    12:08:13.245 [CM112000] Media Server is connected

    12:08:12.364 [CM506002]: Resolved SIP external IP:port (68.194.30.175:5060) on Transport 192.168.1.8:5060

    12:08:12.153 [CM506001]: STUN request to resolve SIP external IP:port mapping is sent to STUN server 75.101.138.128:3478 over Transport 192.168.1.8:5060

    12:08:10.951 [CM501006]: Default Local IP address: [192.168.1.8]

    12:08:10.921 [CM501007]: *** Started Calls Controller thread ***

    12:08:10.751 [CM501002]: Version: 7.1.7060.0

    12:08:10.751 [CM501001]: Start 3CX PhoneSystem Call Manager

    12:08:10.270 Unknown system [DBProvider] tries to connect!
    12:08:10.250 [CM501010]: License Info: Load Failed
     
  2. StefanW

    StefanW Head of Customer Support and Training
    Staff Member 3CX Support

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    Hello,

    on witch Platform do u run the latest version of 3CX-Gateway?
    VMware ESX and HyperV dont work.
    When it is an normal PC did you connect via RDP while u started the channels?
    If you did so, have you changed the settings in the RDP Config to play sound at remote computer and not lokal?
     
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  3. gscherer

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    Its running on a dell win2003 server. I was not using remote desktop at all. I was using my workstation pc with the soft phone app. I got it to work by reconfiguring the sound card drivers on my pc.

    Now what is the procedure to be able to have more than 1 caller dialing my skype in number? Ive read about slave lines but i cant seem to find any setup info. Is it even possible to have more than 1 person calling in on a single skype in number?

    Thanks for the help

    Gary
     
  4. StefanW

    StefanW Head of Customer Support and Training
    Staff Member 3CX Support

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    The Feat. is called Master Line.

    Configure line one as master.
    this is the line u publish on your website then.
    u will be never able to call out over that account in that scenario, so don't load it up with credits

    Example:
    Master: SkypeAccount01*
    Now make another 4 Accounts as u name them is randomy
    Slave1: Skype1
    Slave2: SkyperInbound1
    and so on

    Now what happens, somebody calls SkypeAccount01* the gateway takes the call, keeps the caller in rining and trans fairs the call to any slave line witch is get picked randomly witch forwards the call to the PBX then. As u see the Master accound is idel again...
     
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  5. StefanW

    StefanW Head of Customer Support and Training
    Staff Member 3CX Support

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