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Phone rings but no sound ?

Discussion in '3CX Phone System - General' started by andrelau, Jul 4, 2009.

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  1. andrelau

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    Hello.
    I'm facing a problem i don't understand.

    1st i am able to call / and receive call , on the same computer using
    - 3CX VoIP Phone , configured to use the SIP account of my SIP.

    So what's the problem then ? :D

    I'm playing with the 3CX Phone system (free edition , registered).
    - I configured 3 extensions for 3 different computers on my Lan.
    - i configured a VOIP provider with my SIP account. I can see in the log that it registers.
    - i can call one extension from another
    - i can call outside (my cellular) from my extention : Cellular rings. but when i take the call : NO SOUND , in both the cellular and the extention .....
    - i can call my SIP phone number and make all extension ring at the same time (using group). If i take the call : NO SOUND !

    any idea of what's going wrong ?

    I checked the firewall using the dedicated menu . Everything is "checked passed".
     
  2. andrelau

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    Here's a log of what happens when i receive a call from my cellular and take the call.
    Code:
    [size=85]09:04:32.901  [MS105000] C:2.1: No RTP packets were received:remoteAddr=212.27.52.130:37882,extAddr=0.0.0.0:0,localAddr=82.xxx.xxx.xxxx:9000
    09:04:32.899  [CM503008]: Call(2): Call is terminated
    09:04:32.896  [CM503008]: Call(2): Call is terminated
    09:04:12.066  Session 38 of leg C:2.1 is confirmed
    09:04:11.909  [CM503007]: Call(2): Device joined: sip:10@192.168.xx.xx:54525;rinstance=2921cb3e0b7a4804
    09:04:11.905  [CM503007]: Call(2): Device joined: sip:172.17.20.241:5062
    09:04:11.903  [MS210003] C:2.1:Answer provided. Connection(transcoding mode):82.238.7.140:9000(9001)
    09:04:11.901  [MS210001] C:2.2:Answer received. RTP connection: 192.168.xx.xx:40012(40013)
    09:04:11.898  Remote SDP is set for legC:2.2
    09:04:08.900  [CM505001]: Ext.10: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CXVoipPhone;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXVoipPhone 3.1.6288.0] Transport: [sip:192.168.xx.xx:5060]
    09:04:08.900  [CM503002]: Call(2): Alerting sip:10@192.168.xx.xx:54525;rinstance=2921cb3e0b7a4804
    09:04:08.448  [CM503024]: Call(2): Calling Ext:Ext.10@[Dev:sip:10@192.168.xx.xx:54525;rinstance=2921cb3e0b7a4804]
    [/size]
    I'm very surprised with the MS210001 C:2.2 Answer received RTP Connection on port 40012 ?? , i opened this port for another soft and it looks like it's using it, despite the 9000 - 9049 range ?
    and the 09:04:32.901 [MS105000] C:2.1: No RTP packets were received:remoteAddr=212.27.52.130:37882,extAddr=0.0.0.0:0,localAddr=82.xxx.xxx.xxxx:9000

    Edit: I suspected a bad "static" configuratio, so i checked the STUN "back".

    but still no RTP packets :s
    Code:
    09:33:27.652  [MS105000] C:3.1: No RTP packets were received:remoteAddr=212.27.52.129:37214,extAddr=82.xxx.xxx.xxx:40002,localAddr=82.xxx.xxx.xxx:40002
     
  3. andrelau

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    i have analysed the traffic using Wireshark, as suggested in the wiki.
    I can see UDP packet outgoing from the local IP of the 3CX server to the sip domain ? , but when i select one to analyse RTP traffic it tells my that it's not RTP and that i have 0 RTP packet (which is consistent ! :d )
     
  4. Vali_3CX

    Vali_3CX Well-Known Member
    Staff Member 3CX Support

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    Hi
    About 3CX Voip Phone configuration - if you dial 999, do you have audio?
    Regards
    vali
     
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  5. andrelau

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    Hi, thanks for the answer.
    i'll check 999. but everything worked fine, sound included. Except the Audio when i call phone number outside or when i take an incoming call from my SIP provider (my isp, which offers a SIP account attached to our phone number).

    At this moment, i uninstalled 3CX and i'm trying AsteriskNow.
    With asterisk I am able to make call using the voip and hear the sound. So they're hope :)

    I'll try to reinstall 3CX cause i find it much easier to configure.
     
  6. SY

    SY Well-Known Member
    3CX Support

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    Hi,

    How did you manage to compare functionality of 3CX PBX and AsteriskNow on the SAME (windows) environment? Is there some version of "AsteriskNow for windows"?

    Thanks
     
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  7. andrelau

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    Nope.
    AsteriskNow comes on a CD (.iso to burn). Then you boot and it installs ALL (os + database, asterisk, etc). Then you have to connect to this server from another computer using http:// and ... look for the admin / passwd ! :D .

    I have a dedicated machine i used to test asterisk now. It was far less intuitive (to my taste) than 3CX web interface. (i tested 3CX on my desktop but now i plan to install it on this dedicated computer)

    I"m NOT saying that one is better than another :D, just that it was EASIER for me to do my first step in voip and pbx configuration with 3CX.

    Now, I'm reinstalling windows on this computer. Then, i'll reinstall 3CX.

    I also changed my network configuration.
    Before, my ISP box was configured in router mode and pc where plugged on the switch of the ISP box.
    Now, the ISP box is no more in router configuration, but on the WAN port of a D-Link router (that was sleeping on a shelf) DGL-4300 , latest firmware.

    I'll be back ;)

    /installmode : on
     
  8. James21

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    Just curious, but how does 'Asterisk Now' work for you? Do you like it? I am looking for a new system, and my coworker recommended this program. If anyone has tried it, please let me know how it works for them....
     
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  9. andrelau

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    I uninstalled it. 3CX is installed back. I found asterisk much more complcated to configure for my Voip provider.
    At this moment, i was not able to make them work fine , both of them. With 3CX, I'm facing problem with sound when it comes to call outside. "No RTP Packet found".

    - With 3CX, i can easily configure my voip provider, make it register, create a route, configure voice mail etc... ALL THIS WITH THE GUi. No conf file edit.
    it was as simple as , download, install , follow the wizard, create 2 extension to play between lan computers and "understand" the concept of extensions . That's it! all was working , fine and quick in less than half a day (including some testing, firewall configuration, reconfiguration of extension with stronger password, etc..)

    With all these new knowledges of the Voip world, i tried AsteriskNOW.

    With asterisk , all this was much more complicated. (conf file editing, etc..). The learn "curve" seems much longer.
    The only way i found to make it work with my voip provider was to manually edit the config file. did not work with the gui.... :s
     
  10. SY

    SY Well-Known Member
    3CX Support

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    Hi,

    Media server component of our PBX takes care of audio delivery/negotiation between endpoints and controls it.
    3CXMediaServer.log and 3CXMediaServer.trace.log(especially in verbose mode) may provide enough information regarding audio problems.

    Thanks
     
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  11. andrelau

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    thanks, i'll check cause after a complete reinstall, new "lan config" with SIP compliant router, the problem still exists.

    From the office, using tunnel i can reach the PBX and when i press 999 i have sound.
    I created a route "out" to my SIP provider.
    when i press 0, then dial and call my cellular for example, the cellular rings. if i take the call i have no sound ! (both ways)
    When i stop and look at the log in web interface it tells NO RTP packets.

    I'll check those media log.
     
  12. andrelau

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    Here's the Mediaserver.trace, just after a server restart, waiting for the Voip to Register.
    then Make a call from cellular to my Voip phone number.
    Softphone rings on my computer, i take the call . No sound . I hang on the cellular.

    XXX.XXX.7.140 is my public IP. (text replacement)
    212.27.52.5 is Freephonie.net, but i don't know for 212.27.52.130

    192.168.0.135 is my PBX
    192.168.0.45 is my computer

    Port check via the web user interface of 3CX is ok.

    Any help welcomed.

    Code:
    2009/07/09 >>
    23:19:07.629|.\Service.cpp(244)|Log2||??:*** Initialize ***
    23:19:07.659|.\Service.cpp(320)|Log2||??:*** Start Server ***
    23:19:07.659|.\Service.cpp(80)|Log2||??:*** Listen ***
    23:19:07.669|.\Service.cpp(96)|Log2||??:*** Connecting to [127.0.0.1:5482] ***
    23:19:07.669|.\Service.cpp(107)|Log2||??:*** Connected to pbx-3CX:5482/CallManager at 127.0.0.1:5482 ***
    23:19:13.658|.\Service.cpp(159)|Trace5||??:ParametersUpdated:=====[03070440]
    STUN=ARRAY
      [0]=stun.3cx.com:3478
      [1]=stun2.3cx.com:3478
    ADD2833=1
    ANSP=0
    CS=62914560
    EXTADDR=XXX.XXX.7.140
    EXTIF=192.168.0.135
    FEP=9000
    FLP=7000
    LEP=9049
    LLP=7499
    LOCALADDR=
    MCFS=5242880
    MOH=C:\ProgramData\3CX\Data\Ivr\Prompts\onhold.mp3
    MS_BLINDEXTPORTS=0
    STUNTOUT=3000
    TOS=0
    allowExternalOn=0.0.0.0/0
    allowSourceAsOutbound=0
    externalIPRange=
    reportSourceConflicts=0
    =================================
    
    23:38:54.090|.\MSEndPoint.cpp(846)|Trace5||??:Source ports: 9000 STUN RTP:XXX.XXX.7.140:9000 RTCP:XXX.XXX.7.140:9001 mapped to XXX.XXX.7.140, ports 9000,9001
    23:38:54.200|.\MediaServer.cpp(636)|Trace5||??:EndPoint created: (destination=212.27.52.5)
    EndPoint: ID=00000001@(EXTERNAL)
      LOGID=C:1.1  Status: MSEP_STUN
      RTP:XXX.XXX.7.140:9000
      RTCP:XXX.XXX.7.140:9001
      STUN RTP:XXX.XXX.7.140:9000
      STUN RTCP:XXX.XXX.7.140:9001
      Coder:
        NOT SET
        101:telephony-event
        Party ptime:20
      Party RTP:0.0.0.0:0
      Party RTCP:0.0.0.0:0
      Decoders:
        <empty>
    
    23:38:54.210|.\MediaServer.cpp(1082)|Trace5||??:EP 00000001@ joined to call 1
    23:38:54.210|.\MediaServer.cpp(1091)|Trace5||??:starting send on new call 1
    23:38:54.280|.\MediaServer.cpp(733)|Trace5||??:Set Party EP:C:1.1SDP:v=0
    o=cp10 124717553612 124717553612 IN IP4 212.27.52.130
    s=SIP Call
    c=IN IP4 212.27.52.130
    t=0 0
    m=audio 31810 RTP/AVP 8
    b=AS:64
    a=rtpmap:8 PCMA/8000/1
    a=ptime:30
    
    23:38:54.280|.\MSEndPoint.cpp(2482)|Trace5||??:New state00000001@ SILENCE
    23:38:54.280|.\MSEndPoint.cpp(1494)|Log2||??:[MS210000] C:1.1:Offer received. RTP connection: 212.27.52.130:31810(31811)
    23:38:54.490|.\MSEndPoint.cpp(591)|Trace5||??:00000002@: RTP bind address 192.168.0.135:7000
    23:38:54.490|.\MSEndPoint.cpp(602)|Trace5||??:00000002@: RTCP bind address 192.168.0.135:7001
    23:38:54.490|.\MediaServer.cpp(636)|Trace5||??:EndPoint created: (destination=192.168.0.45)
    EndPoint: ID=00000002@(LOCAL)
      LOGID=C:1.2  Status: MSEP_LOCAL
      RTP:192.168.0.135:7000
      RTCP:192.168.0.135:7001
      STUN RTP:0.0.0.0:0
      STUN RTCP:0.0.0.0:0
      Coder:
        NOT SET
        101:telephony-event
        Party ptime:20
      Party RTP:0.0.0.0:0
      Party RTCP:0.0.0.0:0
      Decoders:
        <empty>
    
    23:38:54.490|.\MediaServer.cpp(1082)|Trace5||??:EP 00000002@ joined to call 1
    23:38:54.490|.\MediaServer.cpp(1127)|Trace5||??:EndPoint 00000002@ removed from call 1
    23:38:54.490|.\MediaServer.cpp(1082)|Trace5||??:EP 00000002@ joined to call 1
    23:38:54.500|.\MSEndPoint.cpp(1127)|Log2||??:[MS210002] C:1.2:Offer provided. Connection(transcoding mode): 192.168.0.135:7000(7001)
    23:38:54.500|.\MediaServer.cpp(671)|Trace5||??:Get Local SDP. EP:C:1.2SDP:v=0
    o=3cxPS 299137761280 428120997889 IN IP4 192.168.0.135
    s=3cxPS Audio call
    c=IN IP4 192.168.0.135
    t=0 0
    m=audio 7000 RTP/AVP 0 8 3 13 110 99 101
    c=IN IP4 192.168.0.135
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:13 CN/8000
    a=rtpmap:110 iLBC/8000
    a=rtpmap:99 SPEEX/8000
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    
    23:38:54.511|.\MediaServer.cpp(1127)|Trace5||??:EndPoint 00000002@ removed from call 1
    23:38:57.835|.\MediaServer.cpp(733)|Trace5||??:Set Party EP:C:1.2SDP:v=0
    o=- 5 2 IN IP4 192.168.0.45
    s=CounterPath X-Lite 3.0
    c=IN IP4 192.168.0.45
    t=0 0
    m=audio 12812 RTP/AVP 0 8 3 101
    a=fmtp:101 0-15
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    
    23:38:57.835|.\MSEndPoint.cpp(2482)|Trace5||??:New state00000002@ SILENCE
    23:38:57.835|.\MSEndPoint.cpp(2482)|Trace5||??:New state00000002@ ACTIVE
    23:38:57.835|.\MSEndPoint.cpp(1499)|Log2||??:[MS210001] C:1.2:Answer received. RTP connection: 192.168.0.45:12812(12813)
    23:38:57.835|.\MediaServer.cpp(1082)|Trace5||??:EP 00000002@ joined to call 1
    23:38:57.835|.\MSEndPoint.cpp(2482)|Trace5||??:New state00000001@ ACTIVE
    23:38:57.835|.\MSEndPoint.cpp(2482)|Trace5||??:New state00000002@ ACTIVE
    23:38:57.845|.\MSEndPoint.cpp(1132)|Log2||??:[MS210003] C:1.1:Answer provided. Connection(transcoding mode):XXX.XXX.7.140:9000(9001)
    23:38:57.845|.\MediaServer.cpp(671)|Trace5||??:Get Local SDP. EP:C:1.1SDP:v=0
    o=3cxPS 251356250112 231995342849 IN IP4 XXX.XXX.7.140
    s=3cxPS Audio call
    c=IN IP4 XXX.XXX.7.140
    t=0 0
    m=audio 9000 RTP/AVP 8 101
    c=IN IP4 XXX.XXX.7.140
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    
    23:38:57.845|.\MSEndPoint.cpp(1696)|Trace5||??:EP:00000002@ refreshed
    23:39:10.103|.\MediaServer.cpp(1127)|Trace5||??:EndPoint 00000002@ removed from call 1
    23:39:10.113|.\MediaServer.cpp(1127)|Trace5||??:EndPoint 00000001@ removed from call 1
    23:39:10.123|.\MediaServer.cpp(886)|Trace5||??:references to EndPoint 00000002@ were removed
    23:39:10.123|.\MSEndPoint.cpp(686)|Trace5||??:EndPoint 00000002@ destroyed
    23:39:10.123|.\MSEndPoint.cpp(687)|Trace5||??:EndPoint 00000002@ destroyed statistics:
    Begin send:175961589
    LastSent:175973566
    TotalSent:599
    MaxSendInterval:40
    MinSendInterval:0
    AvgSendInterval:19.995
    SendDeviation:1.74164
    Begin recv:175961529
    Last recv:175973506
    TotalRecv:599
    MaxRecvInterval:30
    MinRecvInterval:10
    AvgRecvInterval:19.995
    ReceiveDeviation:1.53805
    
    23:39:10.123|.\RTPReceiver.cpp(127)|Trace5||??:Endpoint for socket 800 is not found!
    23:39:10.233|.\MediaServer.cpp(886)|Trace5||??:references to EndPoint 00000001@ were removed
    23:39:10.233|.\MSEndPoint.cpp(675)|Error1||??:[MS105000] C:1.1: No RTP packets were received:remoteAddr=212.27.52.130:31810,extAddr=XXX.XXX.7.140:9000,localAddr=XXX.XXX.7.140:9000
    23:39:10.233|.\MSEndPoint.cpp(686)|Trace5||??:EndPoint 00000001@ destroyed
    23:39:10.233|.\MSEndPoint.cpp(687)|Trace5||??:EndPoint 00000001@ destroyed statistics:
    Begin send:175961629
    LastSent:175973566
    TotalSent:398
    MaxSendInterval:50
    MinSendInterval:20
    AvgSendInterval:29.9925
    SendDeviation:10.0547
    Begin recv:0
    Last recv:0
    TotalRecv:0
    MaxRecvInterval:0
    MinRecvInterval:0
    AvgRecvInterval:0
    ReceiveDeviation:0
    
    23:39:10.243|.\RTPReceiver.cpp(127)|Trace5||??:Endpoint for socket 0 is not found!
    23:39:10.243|.\MediaServer.cpp(1205)|Trace5||??:references to call 1 were removed
    23:39:10.243|.\MSCallConf.cpp(33)|Trace5||??:Call: 1 destroyed
    
    
     
  13. Discovery Technology

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    The issue looks to be STUN-related or possibly your RTP port forwarding hasn't been setup correctly on your firewall - it's been a while since your last post, have you since resolved this issue?

    what type of firewall are you using?
     
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  14. andrelau

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    all was fine on the routeur / Firewall.
    The proof ? Asterisk is now up and running with sound + video :) on ubuntu server.

    Harder to get in and configure things, but hey it now works fine.
    thanks for the help and thanks to 3CX to publish these free software. I'm sure they work fine in many configurations. Mine was certainly "specific" and ISP related.

    Case closed for me.
     
  15. maker2254

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    I have the same problem , I have asterix I downloaded 3cx installed up and running with sip. I have polycom phone in network works fine but when I put my phone to remote network The phone rings but no sound at all .
    The phone server has 192.168.1.45 which runs under Verizon (ISP ) and my Polycom is in Optimum . I put public Ip address of Phone server to polycom settings My phone rings but no sosund. Also when I call from my cell phone , as if it is not dialing (becuase no dialing tone) but Polycom rings .
    I am very exhausted , either I need to renew my license with fonality or I should solve this problem . ANy recommendation please
     
  16. leejor

    leejor Well-Known Member

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    First off, you should not be resurrecting a thread from almost four years ago. Just post a fresh one.

    Can't help with anything related to asterix, but, no audio, of "remote" extensions, has been a common problem.

    I'm not too clear on your particular set-up, or if you are trying the remote extension at more than one location. You may need to provide more detail.

    One issue is that STUN used to be required in almost all cases, as without it, the extension was seen as registering with a private IP. Now, that isn't always necessary, but you should check to see how the remote extension is "perceived" by the PBX (IP and port).

    In some cases, the remote extension can be reached with SIP signalling, but voice packets don't get through. This may have something to do with the router at the far end being unable to determine where to send the voice "packets". At the PBX end, the router there, is required to have all ports, required for both SIP signalling and voice, forwarded to the IP of the PBX.
     
  17. lneblett

    lneblett Well-Known Member

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    As Leejor, pointed out you provided virtually no detail of your setup, so start with this -

    http://www.3cx.com/blog/voip-howto/remote-extensions/
     
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