Ipphones in remote location

Discussion in '3CX Phone System - General' started by mbs1, Mar 7, 2007.

  1. mbs1

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    Hi,

    I hope someone can help.

    Xlite in a remote location ( behind a nat ) can register to the PBX at a remote location ( also behind a nat). The correct ports are opened on the firewall both ways. The remote soft phone can call a phone ( ipphone or xlite ) on the same subnet as the pbx, ringing is heard but on pickup the trace shows continious icmp messages. The trace also show the remote XLite as having resolved and sent its public IP address in the SIP Invite message. However the OK Contact message subsequently returned to the Xlite from the PBX shows the internal private ip address of the called party.The remote xlite then tries to send RTP packets to the private ip via the internet which fail and the ICMPs.

    Has anyone got remote Extensions working. Would appreciate any help. Have got the RC1 ( 6 march ) version installed.

    thanks

    mbs
     
  2. dekatech

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    Same problem with IP Phone

    I have the same problem with the Polycom IP Phone. It registers and downloads its' sip.id and boot.cfg files and I receive dial tone on remote phone, however when trying to call the remote extension The server status shows as trying to connect to extensionnumber@localipaddress instead of the public ip address.

    Any ideas??
     
  3. mbs1

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    hi,
    just to add to the first post, when i make a call from a remote location to the central site, the phones rings and on pickup, the remote site can hear the central site but not the other way round. I tried this with calling an extension on the central site, a provider for fwd and voipcheap. the same in all three cases. audio only from central to remote.

    calls within the central site are ok

    the trace on all three cases shows the comtact 200 ok from central giving the private ip of the pbx,the remote then trying to send rtp to the private across internet. i think this may be the root of the problem.

    again, if anyone has tried and got this working, would appreciate help.

    maybe this is a feature that will only be available in enterprise?.

    3cx support pls help.

    thanks
     
  4. Anonymous

    Anonymous Guest

    NAT issue, seems to be arround for a while.

    If you can try to get a STUN server on either side. if that does not work port through a VPN.
     
  5. archie

    archie Well-Known Member
    3CX Staff

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    This should be fixed in latest version (RC1). Check it, please.
     
  6. mbs1

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    itfarmer and archie, thanks ever so much for your responses.

    I am currently using rC1 6th March and having these issues. if you need any logs etc let me know.

    thanks again

    mbs
     
  7. archie

    archie Well-Known Member
    3CX Staff

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    In the extension configuration page set the check-box - 'External Phone'
     
  8. mbs1

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    thanks archie,

    i will try that and post back.

    appreciate your help

    mbs
     
  9. mbs1

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    hi all,

    hope all is well.

    I set the external option for external phones but i still cant get audio from remote to central.I have attached the trace as performed on the pbx server.A.B.C.D is the public ip of the remote ip phone. 192.168.201.7 is the private pbx ip and 192.168.201.105 is the private ip of internal phone.

    the call is from A.B.C.D to 192.168.201.105. Sorry its a bit long. From my understanding Packet 43 is from the pbx server to the public ip of remote phone giving the IP address and rtp ports that it would accept to. Lokking at this packet the pbx is giving out the private ip and not the public ip of the central site and i suspect this may be the problem. if i do a trace from the remote site i would see a lot of requests from the remote ip phone to the private ip of the pbx. off course these would never get through.
    the stun settings are set on the pbx to stun.3cx.com

    can i please request asssistance from all experts please


    No. Time Source Destination Protocol Info
    21 54.751316 A.B.C.D 192.168.201.7 SIP/SDP Request: INVITE sip:100@192.168.201.7, with session description

    Frame 21 (722 bytes on wire, 722 bytes captured)

    Session Initiation Protocol
    Request-Line: INVITE sip:100@192.168.201.7 SIP/2.0
    Method: INVITE
    Resent Packet: False
    Message Header
    Via: SIP/2.0/UDP A.B.C.D:5064;branch=z9hG4bKY6U1RHsvl8t1GDf3;rport
    Max-Forwards: 70
    User-Agent: IP PHONE 2 V1.52.007 CFG0
    From: "101" <sip:101@192.168.201.7>;tag=bpsE0S5tofiJvMgC
    SIP Display info: "101"
    SIP from address: sip:101@192.168.201.7
    SIP tag: bpsE0S5tofiJvMgC
    To: "100" <sip:100@192.168.201.7>
    SIP Display info: "100"
    SIP to address: sip:100@192.168.201.7
    Call-ID: tnA1YYe3HhTKX91X@A.B.C.D
    Contact: <sip:101@A.B.C.D:5064>
    Contact Binding: <sip:101@A.B.C.D:5064>
    URI: <sip:101@A.B.C.D:5064>
    SIP contact address: sip:101@A.B.C.D:5064
    CSeq: 1 INVITE
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 243
    Message body
    Session Description Protocol
    Session Description Protocol Version (v): 0
    Owner/Creator, Session Id (o): - 13990265 81066096 IN IP4 A.B.C.D
    Owner Username: -
    Session ID: 13990265
    Session Version: 81066096
    Owner Network Type: IN
    Owner Address Type: IP4
    Owner Address: A.B.C.D
    Session Name (s): SIP CALL
    Connection Information (c): IN IP4 A.B.C.D
    Connection Network Type: IN
    Connection Address Type: IP4
    Connection Address: A.B.C.D
    Time Description, active time (t): 0 0
    Session Start Time: 0
    Session Stop Time: 0
    Media Description, name and address (m): audio 8030 RTP/AVP 3 18 0 101
    Media Type: audio
    Media Port: 8030
    Media Proto: RTP/AVP
    Media Format: GSM 06.10
    Media Format: ITU-T G.729
    Media Format: ITU-T G.711 PCMU
    Media Format: 101
    Media Attribute (a): rtpmap:3 GSM/8000
    Media Attribute Fieldname: rtpmap
    Media Attribute Value: 3 GSM/8000
    Media Attribute (a): rtpmap:18 G729/8000
    Media Attribute Fieldname: rtpmap
    Media Attribute Value: 18 G729/8000
    Media Attribute (a): rtpmap:0 PCMU/8000
    Media Attribute Fieldname: rtpmap
    Media Attribute Value: 0 PCMU/8000
    Media Attribute (a): rtpmap:101 telephone-event/8000
    Media Attribute Fieldname: rtpmap
    Media Attribute Value: 101 telephone-event/8000
    Media Attribute (a): fmtp:101 0-15
    Media Attribute Fieldname: fmtp
    Media Attribute Value: 101 0-15



    No. Time Source Destination Protocol Info
    25 55.222144 192.168.201.7 A.B.C.D SIP Status: 100 Trying

    Frame 25 (323 bytes on wire, 323 bytes captured)

    Session Initiation Protocol
    Status-Line: SIP/2.0 100 Trying
    Status-Code: 100
    Resent Packet: False
    Message Header
    Via: SIP/2.0/UDP A.B.C.D:5064;branch=z9hG4bKQP8mUkfGDhuNaYtQ;rport=5064
    To: "100"<sip:100@192.168.201.7>;tag=a6491607
    SIP Display info: "100"
    SIP to address: sip:100@192.168.201.7
    SIP tag: a6491607
    From: "101"<sip:101@192.168.201.7>;tag=bpsE0S5tofiJvMgC
    SIP Display info: "101"
    SIP from address: sip:101@192.168.201.7
    SIP tag: bpsE0S5tofiJvMgC
    Call-ID: tnA1YYe3HhTKX91X@A.B.C.D
    CSeq: 2 INVITE
    Content-Length: 0

    No. Time Source Destination Protocol Info
    26 55.332017 192.168.201.7 192.168.201.105 SIP/SDP Request: INVITE sip:100@192.168.201.105:5060, with session description

    Frame 26 (787 bytes on wire, 787 bytes captured)

    Session Initiation Protocol
    Request-Line: INVITE sip:100@192.168.201.105:5060 SIP/2.0
    Method: INVITE
    Resent Packet: False
    Message Header
    Via: SIP/2.0/UDP 192.168.201.7:5060;branch=z9hG4bK-d87543-7b33fb266e0cee4f-1--d87543-;rport
    Max-Forwards: 70
    Contact: <sip:101@192.168.201.7:5060>
    Contact Binding: <sip:101@192.168.201.7:5060>
    URI: <sip:101@192.168.201.7:5060>
    SIP contact address: sip:101@192.168.201.7:5060
    To: "100"<sip:100@192.168.201.105:5060>
    SIP Display info: "100"
    SIP to address: sip:100@192.168.201.105:5060
    From: "101"<sip:101@192.168.201.7:5060>;tag=11462903
    SIP Display info: "101"
    SIP from address: sip:101@192.168.201.7:5060
    SIP tag: 11462903
    Call-ID: YWQwNzUzOWJhNWM3MDUxOWIxNmQ5NTg1ZGY5NGJjZGY.
    CSeq: 1 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
    Content-Type: application/sdp
    Content-Length: 246
    Message body
    Session Description Protocol
    Session Description Protocol Version (v): 0
    Owner/Creator, Session Id (o): 3cxPS 12818015825872 12818015825873 IN IP4 192.168.201.7
    Owner Username: 3cxPS
    Session ID: 12818015825872
    Session Version: 12818015825873
    Owner Network Type: IN
    Owner Address Type: IP4
    Owner Address: 192.168.201.7
    Session Name (s): 3cxPS Audio call
    Connection Information (c): IN IP4 192.168.201.7
    Connection Network Type: IN
    Connection Address Type: IP4
    Connection Address: 192.168.201.7
    Time Description, active time (t): 0 0
    Session Start Time: 0
    Session Stop Time: 0
    Media Description, name and address (m): audio 8034 RTP/AVP 0 8 3 98
    Media Type: audio
    Media Port: 8034
    Media Proto: RTP/AVP
    Media Format: ITU-T G.711 PCMU
    Media Format: ITU-T G.711 PCMA
    Media Format: GSM 06.10
    Media Format: 98
    Media Attribute (a): rtpmap:0 PCMU/8000
    Media Attribute Fieldname: rtpmap
    Media Attribute Value: 0 PCMU/8000
    Media Attribute (a): rtpmap:8 PCMA/8000
    Media Attribute Fieldname: rtpmap
    Media Attribute Value: 8 PCMA/8000
    Media Attribute (a): rtpmap:3 GSM/8000
    Media Attribute Fieldname: rtpmap
    Media Attribute Value: 3 GSM/8000
    Media Attribute (a): rtpmap:98 telephone-event/8000
    Media Attribute Fieldname: rtpmap
    Media Attribute Value: 98 telephone-event/8000

    No. Time Source Destination Protocol Info
    27 55.444866 192.168.201.105 192.168.201.7 SIP Status: 180 Ringing

    Frame 27 (472 bytes on wire, 472 bytes captured)

    Session Initiation Protocol
    Status-Line: SIP/2.0 180 Ringing
    Status-Code: 180
    Resent Packet: False
    Message Header
    From: "101"<sip:101@192.168.201.7:5060>;tag=11462903
    SIP Display info: "101"
    SIP from address: sip:101@192.168.201.7:5060
    SIP tag: 11462903
    To: "100"<sip:100@192.168.201.105:5060>;tag=c0a8c969-13c4-45f2d558-269fe9-5798
    SIP Display info: "100"
    SIP to address: sip:100@192.168.201.105:5060
    SIP tag: c0a8c969-13c4-45f2d558-269fe9-5798
    Call-ID: YWQwNzUzOWJhNWM3MDUxOWIxNmQ5NTg1ZGY5NGJjZGY.
    CSeq: 1 INVITE
    Via: SIP/2.0/UDP 192.168.201.7:5060;rport=5060;branch=z9hG4bK-d87543-7b33fb266e0cee4f-1--d87543-
    Supported: replaces
    User-Agent: SIP Phone
    Contact: <sip:100@192.168.201.105:5060>
    Contact Binding: <sip:100@192.168.201.105:5060>
    URI: <sip:100@192.168.201.105:5060>
    SIP contact address: sip:100@192.168.201.105:5060
    Content-Length: 0

    No. Time Source Destination Protocol Info
    28 55.550222 192.168.201.7 A.B.C.D SIP Status: 180 Ringing

    Frame 28 (368 bytes on wire, 368 bytes captured)

    Session Initiation Protocol
    Status-Line: SIP/2.0 180 Ringing
    Status-Code: 180
    Resent Packet: False
    Message Header
    Via: SIP/2.0/UDP A.B.C.D:5064;branch=z9hG4bKQP8mUkfGDhuNaYtQ;rport=5064
    Contact: "101"<sip:101@192.168.201.7:5060>
    Contact Binding: "101"<sip:101@192.168.201.7:5060>
    URI: "101"<sip:101@192.168.201.7:5060>
    SIP Display info: "101"
    SIP contact address: sip:101@192.168.201.7:5060
    To: "100"<sip:100@192.168.201.7>;tag=a6491607
    SIP Display info: "100"
    SIP to address: sip:100@192.168.201.7
    SIP tag: a6491607
    From: "101"<sip:101@192.168.201.7>;tag=bpsE0S5tofiJvMgC
    SIP Display info: "101"
    SIP from address: sip:101@192.168.201.7
    SIP tag: bpsE0S5tofiJvMgC
    Call-ID: tnA1YYe3HhTKX91X@A.B.C.D
    CSeq: 2 INVITE
    Content-Length: 0

    No. Time Source Destination Protocol Info
    29 59.604918 192.168.201.105 192.168.201.7 SIP/SDP Status: 200 OK, with session description

    Frame 29 (727 bytes on wire, 727 bytes captured)

    Session Initiation Protocol
    Status-Line: SIP/2.0 200 OK
    Status-Code: 200
    Resent Packet: False
    Message Header
    From: "101"<sip:101@192.168.201.7:5060>;tag=11462903
    SIP Display info: "101"
    SIP from address: sip:101@192.168.201.7:5060
    SIP tag: 11462903
    To: "100"<sip:100@192.168.201.105:5060>;tag=c0a8c969-13c4-45f2d558-269fe9-5798
    SIP Display info: "100"
    SIP to address: sip:100@192.168.201.105:5060
    SIP tag: c0a8c969-13c4-45f2d558-269fe9-5798
    Call-ID: YWQwNzUzOWJhNWM3MDUxOWIxNmQ5NTg1ZGY5NGJjZGY.
    CSeq: 1 INVITE
    Via: SIP/2.0/UDP 192.168.201.7:5060;rport=5060;branch=z9hG4bK-d87543-7b33fb266e0cee4f-1--d87543-
    Supported: replaces
    User-Agent: SIP Phone
    Contact: <sip:100@192.168.201.105:5060>
    Contact Binding: <sip:100@192.168.201.105:5060>
    URI: <sip:100@192.168.201.105:5060>
    SIP contact address: sip:100@192.168.201.105:5060
    Content-Type: application/sdp
    Content-Length: 227
    Message body
    Session Description Protocol
    Session Description Protocol Version (v): 0
    Owner/Creator, Session Id (o): SIP302 100 0 IN IP4 192.168.201.105
    Owner Username: SIP302
    Session ID: 100
    Session Version: 0
    Owner Network Type: IN
    Owner Address Type: IP4
    Owner Address: 192.168.201.105
    Session Name (s): Audio Session
    Session Information (i): Audio Session
    Connection Information (c): IN IP4 192.168.201.105
    Connection Network Type: IN
    Connection Address Type: IP4
    Connection Address: 192.168.201.105
    Time Description, active time (t): 0 0
    Session Start Time: 0
    Session Stop Time: 0
    Media Description, name and address (m): audio 8000 RTP/AVP 0 101
    Media Type: audio
    Media Port: 8000
    Media Proto: RTP/AVP
    Media Format: ITU-T G.711 PCMU
    Media Format: 101
    Media Attribute (a): ptime:20
    Media Attribute Fieldname: ptime
    Media Attribute Value: 20
    Media Attribute (a): fmtp:101 0-11
    Media Attribute Fieldname: fmtp
    Media Attribute Value: 101 0-11
    Media Attribute (a): rtpmap:0 PCMU/8000/1
    Media Attribute Fieldname: rtpmap
    Media Attribute Value: 0 PCMU/8000/1
    Media Attribute (a): rtpmap:101 telephone-event/8000
    Media Attribute Fieldname: rtpmap
    Media Attribute Value: 101 telephone-event/8000

    No. Time Source Destination Protocol Info
    43 59.707702 192.168.201.7 A.B.C.D SIP/SDP Status: 200 OK, with session description

    Frame 43 (702 bytes on wire, 702 bytes captured)

    Session Initiation Protocol
    Status-Line: SIP/2.0 200 OK
    Status-Code: 200
    Resent Packet: False
    Message Header
    Via: SIP/2.0/UDP A.B.C.D:5064;branch=z9hG4bKQP8mUkfGDhuNaYtQ;rport=5064
    Contact: "101"<sip:101@192.168.201.7:5060>
    Contact Binding: "101"<sip:101@192.168.201.7:5060>
    URI: "101"<sip:101@192.168.201.7:5060>
    SIP Display info: "101"
    SIP contact address: sip:101@192.168.201.7:5060
    To: "100"<sip:100@192.168.201.7>;tag=a6491607
    SIP Display info: "100"
    SIP to address: sip:100@192.168.201.7
    SIP tag: a6491607
    From: "101"<sip:101@192.168.201.7>;tag=bpsE0S5tofiJvMgC
    SIP Display info: "101"
    SIP from address: sip:101@192.168.201.7
    SIP tag: bpsE0S5tofiJvMgC
    Call-ID: tnA1YYe3HhTKX91X@A.B.C.D
    CSeq: 2 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
    Content-Type: application/sdp
    Content-Length: 222
    Message body
    Session Description Protocol
    Session Description Protocol Version (v): 0
    Owner/Creator, Session Id (o): 3cxPS 12818015825826 12818015825828 IN IP4 192.168.201.7
    Owner Username: 3cxPS
    Session ID: 12818015825826
    Session Version: 12818015825828
    Owner Network Type: IN
    Owner Address Type: IP4
    Owner Address: 192.168.201.7
    Session Name (s): 3cxPS Audio call
    Connection Information (c): IN IP4 192.168.201.7
    Connection Network Type: IN
    Connection Address Type: IP4
    Connection Address: 192.168.201.7
    Time Description, active time (t): 0 0
    Session Start Time: 0
    Session Stop Time: 0
    Media Description, name and address (m): audio 9106 RTP/AVP 3 0 98
    Media Type: audio
    Media Port: 9106
    Media Proto: RTP/AVP
    Media Format: GSM 06.10
    Media Format: ITU-T G.711 PCMU
    Media Format: 98
    Media Attribute (a): rtpmap:3 GSM/8000
    Media Attribute Fieldname: rtpmap
    Media Attribute Value: 3 GSM/8000
    Media Attribute (a): rtpmap:0 PCMU/8000
    Media Attribute Fieldname: rtpmap
    Media Attribute Value: 0 PCMU/8000
    Media Attribute (a): rtpmap:98 telephone-event/8000
    Media Attribute Fieldname: rtpmap
    Media Attribute Value: 98 telephone-event/8000


    thanks very much

    mbs
     
  10. mbs1

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    hi all

    Just to add to the above.

    I checked the trace on an outbound call from a ipphone on the same subnet as the pbx to a remote Ipiphone.

    the trace shows the invite going from the pbx to the remote ipiphone with the connection ip address as the local private ip address of the pbx server.
    maybe this could aslo explain why there is no audio from the remote to the central end.

    pls help

    thanks

    mbs
     
  11. archie

    archie Well-Known Member
    3CX Staff

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    Please, check your logs if STUN client was able to resolve your external IP address.
     
  12. mbs1

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    hi archie,

    thanks for getting back.

    I have noticed that on the pbx server status, i only get the External IPaddress resolved correctly if i restart the main service. Otherwise i can see frequent stun client requests but no responses.

    let me get a trace

    thanks again, help appreciated

    mbs
     
  13. mbs1

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    hi

    i have been checking the server status. i can see a message

    stunclient Send Initial Stun request to 194.221.62.209 but there is no response on the log to this request.

    A trace on the pbx pc shows a
    stun binding request to 194.221.62.209
    Stun binding response 194.221.62.209

    the only time i get a response to the stun client request is when the server get booted or i restart the pbx service.

    i have got udp 3478 and 3479 allowed out

    sorry this query is dragging on.

    appreciate your help

    mbs
     
  14. mbs1

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    hi archie,

    i saw in one of the posts that a newer version will have introduced new advanced options, which allows you to bind your device to our Media Server. It should solve one-way audio problem.

    This may cure my problem as well.

    On the general settings, are the stun settings changeable. I can enter a different stun server but if i restart the service, it goes back to stun.3cx.com. if i dont restart, when i logon to the admin screen, it is reset to stun.3cx.com.

    also the log screen shows scheduled intial requests to the ip address of stun.3cx.com but no response. i only get a response if i restart the service.is this normal.

    sorry asking a lot of questions.

    I think i am almost there and if this gets solved many will benefit

    thanks

    mbs
     
  15. mbs1

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    hi

    i saw in one of the posts a reference to a registry key msexternalinterface.

    it is currently set to the private ip address of my pbx server. in this config, a central ipphone can make a call to a voip gateway configured for fwd and as a test the echo test works. the ipphone is configured to use the pbx as a outbound proxy as well. i can see from the trace that the pbx acts as a proxy between the ipphone and the fwd gateway. not phone and pb on same subnet.

    However in this configuration, the internal ipphone can dial a remote ipphone registered to the pbx ( external option set ) but audio is only one way ( central to out not the other way round). i think the reason for this is the pbx giving its contact ip address as the private internal one to the remote end , trace given in previous post. Again i can see the pbx acting as a proxy between the central ipphone and the remote one.

    IF i change the value of msexternalinterface to the public ipaddress of the firewall the the following happens. The remote ipphone can call the fwd gateway and the echo test works. 2 remote ipphone can call each other and there is 2 way audio. Note the remote phones are configured to use stun. from the pbx trace i can see that the pbx is not really acting as a proxy, it tells both ends what rtp ports are in use and the rtp comms is then direct between the endppoints. Success?.......... only partially as this configuration stops the central ipphone on same subnet as the pbx from receiving audio from the fwd gateway. As the pbx is not acting as a proxy it gives the rtp ports to both ends. what is stopping the audio is the config of the firewall which allows inbound rtp to the pbx only. if i allow the relevant rtp port inbound to the ipphone then it works. So i suppose for a site with a small nos of central ipphones the firewall could be configured to allow different inbound rtp to different inbound ports and the phones would have to be configured to use these ports.


    Maybe the new version may fix this.

    can anyone help, please

    credit to the superuser who shared the registry key info as it helped me understand a bit better that there is a real good chance that this may just work.

    thanks

    mbs
     
  16. archie

    archie Well-Known Member
    3CX Staff

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    Hi,

    It is normal behavior. You will get message about External IP resolved, only for first time, or after currently resolved address:port doesn't match newly resolved ones. That is why you see periodic requests, but no message about responses, because they match previous responces.
     
  17. mbs1

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    thanks archie,

    any idea on the contact ip being set to the private ip of the pbx or the msexternalinterface setting determining the mode od operation of the\ proxy.

    am i right in thinking that the pbx should proxy the rtp for all calls ?

    i appreciate your help.

    thanks

    mbs
     
  18. archie

    archie Well-Known Member
    3CX Staff

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    What exactly contact IP do you mean? The address in 'Contact' field of outgoing INVITE should be your external IP resolved by STUN for any endpoint that is marked as external. Though, there's a bug in RC1 due to which, for extensions marked as external, the RTP (receiving) address is local IP or the IP specified in msexternalinterface, but not STUN-resolved one. We've fixed it in upcoming RC2.

    PBX will proxy RTP if one of endpoints is marked as an external. Otherwise, it will try to establish direct connection, unless one of endpoints is IVR, Voice Mailbox, or is placed on-hold.
     
  19. mbs1

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    Hi archie.

    Thanks Very much for our reply.

    by contact ip i mean the connection ip address sent to the external endpoint for it to send rtp packets to. i now understand that RC2 will resolve this issue.

    Thanks very much, with your help i have learnt a lot.

    regards

    mbs
     
  20. dekatech

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    Been following this thread, as I have the exact same issues.

    Can someine calrify for me.

    Does this mean that in RC2 I will be able to have a phone register and communicate with my 3CX server from a remote location outside of my LAN?

    If so,
    Understanding the bug issue in RC1, if I set my msexternalinterface to the external ip (WAN), should'nt I be able to check the external extension box and make phone calls from the remote location through my 3CX server?

    I have been successful at getting the external extension to make calls using my provider to outbound numbers, but extension to extension dialing is not working....

    Travis
     

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