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Problems with calling and hanging up.

Discussion in 'Windows' started by yaro, Oct 15, 2008.

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  1. yaro

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    It often happens when I'm making a phone call using the soft client I can hear the ringing tone for ages so then I try to hang up but the presence indicator stays orange indicating that I'm still ringing. And indeed it is so as after a while someone answers the call. Another thin is that on the customer side they can hear only 2 or three rings before they pick up while I can hear like 20 before the connection happens.
    So how do we fix that?
    Yaro
     
  2. mickp

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    Sounds weird. Could you post your server status logs?

    Mick.
     
  3. yaro

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    15:31:23.325 MediaServerReporting::RTPReceiver [MS105000] C:355.1: No RTP packets were received:remoteAddr=192.168.42.30:42004,extAddr=0.0.0.0:0,localAddr=192.168.42.3:7162
    15:31:23.262 Call::Terminate [CM503008]: Call(355): Call is terminated
    15:31:23.246 Call::RouteFailed [CM503015]: Call(355): Attempt to reach [sip:020783548@192.168.42.3:5060] failed. Reason: Busy
    15:31:23.246 CallLeg::eek:nFailure [CM503003]: Call(355): Call to sip:141020783548@voip.zen.co.uk:5060 has failed; Cause: 486 Busy Here; from IP:212.23.7.228:5060
    15:31:22.356 Line::printEndpointInfo [CM505003]: Provider:[Zen] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Zen Internet Telephony Service] Transport: [sip:192.168.42.3:5060]
    15:31:22.356 CallCtrl::eek:nAnsweredCall [CM503002]: Call(355): Alerting sip:02072310411@voip.zen.co.uk:5060
    15:31:13.184 Call::Terminate [CM503008]: Call(357): Call is terminated
    15:31:13.184 Call::RouteFailed [CM503018]: Normal call termination. Reason: Away status is OFF
    15:31:13.153 CallCtrl::eek:nSelectRouteReq [CM503010]: Making route(s) to [sip:*60@192.168.42.3:5060]
    15:31:13.153 Extension::printEndpointInfo [CM505001]: Ext.20: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX VoIP Client;Rev: 1] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Phone 6.0.727.0] Transport: [sip:192.168.42.3:5060]
    15:31:13.153 CallCtrl::eek:nIncomingCall [CM503001]: Call(357): Incoming call from Ext.20 to [sip:*60@192.168.42.3:5060]
    15:31:09.309 Call::Terminate [CM503008]: Call(356): Call is terminated
    15:31:09.309 Call::RouteFailed [CM503018]: Normal call termination. Reason: Away status is ON
    15:31:09.293 CallCtrl::eek:nSelectRouteReq [CM503010]: Making route(s) to [sip:*61@192.168.42.3:5060]
    15:31:09.278 Extension::printEndpointInfo [CM505001]: Ext.20: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX VoIP Client;Rev: 1] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Phone 6.0.727.0] Transport: [sip:192.168.42.3:5060]
    15:31:09.278 CallCtrl::eek:nIncomingCall [CM503001]: Call(356): Incoming call from Ext.20 to [sip:*61@192.168.42.3:5060]
    15:30:15.294 MediaServerReporting::STUN [MS101003] C:355.2: Possible firewall problem. Address mapping failed on STUN server 192.168.42.3:3478 for local address ":9000"
    15:30:09.153 CallCtrl::eek:nSelectRouteReq [CM503004]: Call(355): Calling: VoIPline:141020783548@(Ln.10004@Zen)@[Dev:sip:02072310411@voip.zen.co.uk:5060]
    15:30:09.137 CallCtrl::eek:nSelectRouteReq [CM503010]: Making route(s) to [sip:020783548@192.168.42.3:5060]
    15:30:09.137 Extension::printEndpointInfo [CM505001]: Ext.20: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX VoIP Client;Rev: 1] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Phone 6.0.727.0] Transport: [sip:192.168.42.3:5060]
    15:30:09.137 CallCtrl::eek:nIncomingCall [CM503001]: Call(355): Incoming call from Ext.20 to [sip:020783548@192.168.42.3:5060]
     
  4. yaro

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    The above is an example when I tried hanging up but it didn't work.
    yaro
     
  5. mickp

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    15:30:15.294 MediaServerReporting::STUN [MS101003] C:355.2: Possible firewall problem. Address mapping failed on STUN server 192.168.42.3:3478 for local address ":9000"

    looks a bit dodgy. Does your sip provider pass the firewall test?

    I'd suggest changing your sip provider to use safe settings, pbx handles audio, no redirect, no reinvite to see if that helps. From the troubleshooting guide http://sites.google.com/a/3cx.com/3cx-wiki/Home/troubleshooting

    What's your network configuration like? Have you done any specific firewall configuration to support 3cx?

    Mick.
     
  6. yaro

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    The STUN isn't configured and from what I red in the troubleshooting bit it doesn't need to for internal extensions. The settings in the management console are as you indicated. We use just the firewall that was shipped with our Netgear router. It's open on 3478 and 5060.
    No other modifications for 3cx were done on it. Firewall test points to "No STUN server was specified. Please specify a STUN server and try again." No other users in the office experience problems with their voip softphones, well except for the dude who's running it on Vista.
    yaro
     
  7. mickp

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    I think you should specify a stun server (stun.3cx.com works) and give it another go. Your logs indicate that you are attempting to communicate with an external sip provider.

    Mick.
     
  8. DeWild1

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    I am using two ADSL lines.
    I had problems with it hanging up and other weird stuff too..
    On the first ADSL line, I am also streaming video. I notice it would drop the connection every 3 seconds or so.
    No matter what I tried, new ADSL modems, not streaming, new router, new server, etc, ETC, etc, I still had the same problem including the voice would cut out part of the words.
    I switched the VOIP to the other ADSL line and all is well now.

    I would check your connection.
     
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