Alcatel OminPCX SIP clients

Discussion in '3CX Phone System - General' started by nbrown, May 9, 2014.

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  1. nbrown

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    Hi, I'm trying to connect our Alcatel OminPCX phone system (192.168.100.235) to 3CX (192.168.100.194) as part of a trial. We don't have SIP trunks on the alcatel system but we do have SIP clients. I've created a new extension and added the connection details as a generic VoIP provider. Registration is OK and when I call this new extension (621) via an alcatel phone it rings a phone on 3CX perfectly OK. But when making a call from the 3CX extension (1000@192.168.100.78) to an alcatel extension (611) nothing happens, just silence. I know the routes and rules are all OK and below is a section of the log file, can anyone help?

    09-May-2014 10:32:53.155 Route to L:5.2[Line:10001>>611] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=7728fa3fcc3ef503 Call-ID=MThiYWM2N2U4NDdmZmRjYjk1NDgxMTM5M2U3YmU1NzM.:
    INVITE sip:611@192.168.100.235:5060 SIP/2.0
    Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-7728fa3fcc3ef503-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:621@192.168.100.194:5061>
    To: <sip:611@192.168.100.235:5060>
    From: "621"<sip:621@192.168.100.235:5060>;tag=93121b39
    Call-ID: MThiYWM2N2U4NDdmZmRjYjk1NDgxMTM5M2U3YmU1NzM.
    CSeq: 1 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    Supported: replaces
    Content-Length: 413

    v=0
    o=3cxPS 255936430080 334386692097 IN IP4 192.168.100.194
    s=3cxPS Audio call
    c=IN IP4 192.168.100.78
    t=0 0
    m=audio 3000 RTP/AVP 0 8 18 9 18 0 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=silenceSupp:eek:ff - - - -
    a=fmtp:18 annexb=no
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv
    09-May-2014 10:32:53.148 Outbound URI is used: sip:192.168.100.235:5060
    09-May-2014 10:32:53.148 SLA is globally disabled
    09-May-2014 10:32:53.146 Added leg L:5.2[Line:10001>>611]
    09-May-2014 10:32:53.108 [Flow] Call(C:5): making call from L:5.1[Extn] to T:Line:10001>>611@[Dev:sip:621@192.168.100.235:5060]
    09-May-2014 10:32:53.108 [CM503027]: Call(C:5): From: Extn:1000 ("Operator Extension" <sip:1000@192.168.100.194:5061>) to T:Line:10001>>611@[Dev:sip:621@192.168.100.235:5060]
    09-May-2014 10:32:53.108 [CM503004]: Call(C:5): Route 1: from L:5.1[Extn] to T:Line:10001>>611@[Dev:sip:621@192.168.100.235:5060]
    09-May-2014 10:32:53.108 Line limit check: Current # of calls for line Lc:10001(@Alcatel[<sip:621@192.168.100.235:5060>]) is 0; limit is 2
    09-May-2014 10:32:53.108 Call(C:5): Call from Extn:1000 to 611 matches outbound rule 'Rule for Sipgate (Nick) (internal 1)'
    09-May-2014 10:32:53.108 [Flow] Call(C:5): has built target endpoint: Out#:>>Rule{Rule for Sipgate (Nick) (internal 1)}>>611 for call from L:5.1[Extn]
    09-May-2014 10:32:53.108 [Flow] Target endpoint for 611 is Out#:>>Rule{Rule for Sipgate (Nick) (internal 1)}>>611
    09-May-2014 10:32:53.108 Selected prefix: 6
    09-May-2014 10:32:53.108 Looking for outbound rule: dialed = [611], processed: [611]; from-ext:
    09-May-2014 10:32:53.108 [Flow] Building target endpoint to 611 from "Operator Extension" <sip:1000@192.168.100.194:5061>
    09-May-2014 10:32:53.108 [CM503010]: Call(C:5): Making route(s) from Extn:1000 to <sip:611@192.168.100.194:5061>
    09-May-2014 10:32:53.108 Remote SDP is set for leg L:5.1[Extn]
    09-May-2014 10:32:53.108 OnOffer from "Operator Extension"<sip:1000@192.168.100.194:5061>;tag=8af1c41f90
    09-May-2014 10:32:53.108 [CM505001]: Endpoint Extn:1000: Device info: Device Identified: [Man: Aastra;Mod: 5x series;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, recvonly] UserAgent: [Aastra 55i/3.3.1.2235] PBX contact: [sip:1000@192.168.100.194:5061]
    09-May-2014 10:32:53.105 Inbound DID: ''; Phonebook Name: ''
    09-May-2014 10:32:53.105 [CM500002]: Call(C:5): Info on incoming INVITE from Extn:1000:
    Invite-IN Recv Req INVITE from 192.168.100.78:5060 tid=67727e5536af0a982.8d0cca12b4a68f69c Call-ID=be2881de189c5934:
    INVITE sip:611@192.168.100.194:5061;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.100.78:5060;branch=z9hG4bK67727e5536af0a982.8d0cca12b4a68f69c
    Max-Forwards: 70
    Contact: "Operator Extension"<sip:1000@192.168.100.78:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1A30B3>"
    To: <sip:611@192.168.100.194:5061;user=phone>
    From: "Operator Extension"<sip:1000@192.168.100.194:5061>;tag=8af1c41f90
    Call-ID: be2881de189c5934
    CSeq: 14044 INVITE
    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
    Content-Type: application/sdp
    Proxy-Authorization: Digest username="1000",realm="3CXPhoneSystem",nonce="414d535c097d31c443:20fff0e9af80367f997982d5161494cf",uri="sip:611@192.168.100.194:5061;user=phone",response="264b6e8bb1645fb4d80b73dbea052bab",algorithm=MD5
    Supported: path, 100rel, replaces
    User-Agent: Aastra 55i/3.3.1.2235
    Allow-Events: aastra-xml, talk, hold, conference, LocalModeStatus
    Content-Length: 382

    v=0
    o=MxSIP 0 1 IN IP4 192.168.100.78
    s=SIP Call
    c=IN IP4 192.168.100.78
    t=0 0
    m=audio 3000 RTP/AVP 0 8 18 9 18 0 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=silenceSupp:eek:ff - - - -
    a=fmtp:18 annexb=no
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv
    09-May-2014 10:32:53.105 [CM503001]: Call(C:5): Incoming call from Extn:1000 to <sip:611@192.168.100.194:5061>
    09-May-2014 10:32:53.096 Outbound URI is used: sip:1000@192.168.100.78:5060;transport=udp
    09-May-2014 10:32:53.096 IncomingCall: C:5 from <sip:1000@192.168.100.194:5061> to <sip:611@192.168.100.194:5061>
    09-May-2014 10:32:53.096 Added leg L:C:5.1[No endpoint yet]
    09-May-2014 10:32:53.096 UasSession 407 started
    09-May-2014 10:32:53.096 Call from "Operator Extension"<sip:1000@192.168.100.194:5061>;tag=8af1c41f90 to <sip:611@192.168.100.194:5061;user=phone>;tag=84379a6f
     
  2. 3CXNP

    3CXNP Support Team
    3CX Support

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    What outbound rules do you have on the 3CX Phone System?
     
    Stop hovering to collapse... Click to collapse... Hover to expand... Click to expand...
  3. leejor

    leejor Well-Known Member

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    The call looks to be routing through OK but what happens later? The start and end of the log shows the same hour, minute, second, so where is the log after a "timeout" period when the call fails?

    Have you used Wireshark to see what is sent, and if there is any response from the Alcatel PBX?
     
  4. nbrown

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    After about 20 seconds the call drops so you get a hang up tone, data below is the wireshark capture and below that is the whole log...

    WIRESHARK
    pPVfHEeddQOaINVITE sip:611@192.168.100.235:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.100.194:5061;branch=z9hG4bK-d8754z-79423e745b451203-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:621@192.168.100.194:5061>
    To: <sip:611@192.168.100.235:5060>
    From: "621"<sip:621@192.168.100.235:5060>;tag=e362662d
    Call-ID: NjExOTdhYTA0OTU1YzRlZTc1Y2FjOTRmYjJiNmMwMWY.
    CSeq: 2 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    Proxy-Authorization: Digest username="621",realm="xnaim",nonce="d661cf04ac69320b25da6482828f2824",uri="sip:611@192.168.100.235:5060",response="85e653f7546025d6588a21126db88449",cnonce="647f3b311c04e32495685586decea460",nc=00000001,qop=auth,algorithm=MD5
    Supported: replaces
    User-Agent: 3CXPhoneSystem 12.0.35528.640 (34969)
    Content-Length: 260

    v=0
    o=3cxPS 444210348032 218523238401 IN IP4 192.168.100.194
    s=3cxPS Audio call
    c=IN IP4 192.168.100.194
    t=0 0
    m=audio 7030 RTP/AVP 8 0 3 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=sendrecv

    WHOLE LOG
    12-May-2014 07:49:18.556 L:8.1[Extn]: Terminating targets, reason:
    12-May-2014 07:49:18.556 Leg L:8.1[Extn] is terminated: Cause: BYE from PBX
    12-May-2014 07:49:18.556 Terminated from "Operator Extension"<sip:1000@192.168.100.194:5061>;tag=5df8454771 to <sip:611@192.168.100.194:5061;user=phone>;tag=8f466075; reason: Rejected
    12-May-2014 07:49:18.556 L:8.1[Extn] Sending: OnSendResp Send 480/INVITE from 0.0.0.0:0 tid=859007c0f31f1f4af.68c3cc52f9fd10929 Call-ID=41a890b05251b9cb:
    SIP/2.0 480 Temporarily Unavailable
    Via: SIP/2.0/UDP 192.168.100.78:5060;branch=z9hG4bK859007c0f31f1f4af.68c3cc52f9fd10929
    To: <sip:611@192.168.100.194:5061;user=phone>;tag=8f466075
    From: "Operator Extension"<sip:1000@192.168.100.194:5061>;tag=5df8454771
    Call-ID: 41a890b05251b9cb
    CSeq: 10288 INVITE
    Warning: 499 CH3CX1.aimaviation.local "No answer"
    Content-Length: 0
    12-May-2014 07:49:18.556 SendMsg from <sip:611@192.168.100.194:5061;user=phone>;tag=8f466075 to "Operator Extension"<sip:1000@192.168.100.194:5061>;tag=5df8454771
    12-May-2014 07:49:18.518 ~Target=VoIPline:611@(Ln.10001@Alcatel)
    12-May-2014 07:49:18.513 Call(C:8) is terminated
    12-May-2014 07:49:18.507 [CM503020]: Call(C:8): Normal call termination. Call originator: Extn:1000. Reason: No answer
    12-May-2014 07:49:18.507 [CM503016]: Call(C:8): Attempt to reach <sip:611@192.168.100.194:5061> from Extn:1000 has failed. Reason: No Answer
    12-May-2014 07:49:18.507 [Flow] Current call diversion path:[]
    12-May-2014 07:49:18.507 L:8.1[Extn] failed to reach Line:10001>>611, reason No Answer
    12-May-2014 07:49:18.507 RerouteReq
    12-May-2014 07:49:18.505 L:8.2[Line:10001>>611]: Terminating targets, reason: SIP ;cause=408 ;text="Request Timeout"
    12-May-2014 07:49:18.505 Leg L:8.2[Line:10001>>611] is terminated: Cause: 408 Request Timeout/INVITE from local
    12-May-2014 07:49:18.505 L:8.2[Line:10001>>611] got Terminated Recv 408/INVITE from 0.0.0.0:0 tid=79423e745b451203 Call-ID=NjExOTdhYTA0OTU1YzRlZTc1Y2FjOTRmYjJiNmMwMWY.:
    SIP/2.0 408 Request Timeout
    Via: SIP/2.0/UDP 192.168.100.194:5061;branch=z9hG4bK-d8754z-79423e745b451203-1---d8754z-;rport
    To: <sip:611@192.168.100.235:5060>;tag=7e096c49
    From: "621"<sip:621@192.168.100.235:5060>;tag=e362662d
    Call-ID: NjExOTdhYTA0OTU1YzRlZTc1Y2FjOTRmYjJiNmMwMWY.
    CSeq: 2 INVITE
    Content-Length: 0
    12-May-2014 07:49:18.505 Terminated from <sip:611@192.168.100.235:5060>;tag=7e096c49 to "621"<sip:621@192.168.100.235:5060>;tag=e362662d; reason: Error
    12-May-2014 07:49:18.505 ~Route=Dev:sip:621@192.168.100.235:5060
    12-May-2014 07:49:18.505 Call to T:Line:10001>>611@[Dev:sip:621@192.168.100.235:5060] from L:8.1[Extn] failed, cause: Cause: 408 Request Timeout/INVITE from local
    12-May-2014 07:49:18.491 L:8.2[Line:10001>>611] got Failure: Failure Recv 408/INVITE from 0.0.0.0:0 tid=79423e745b451203 Call-ID=NjExOTdhYTA0OTU1YzRlZTc1Y2FjOTRmYjJiNmMwMWY.:
    SIP/2.0 408 Request Timeout
    Via: SIP/2.0/UDP 192.168.100.194:5061;branch=z9hG4bK-d8754z-79423e745b451203-1---d8754z-;rport
    To: <sip:611@192.168.100.235:5060>;tag=7e096c49
    From: "621"<sip:621@192.168.100.235:5060>;tag=e362662d
    Call-ID: NjExOTdhYTA0OTU1YzRlZTc1Y2FjOTRmYjJiNmMwMWY.
    CSeq: 2 INVITE
    Content-Length: 0
    12-May-2014 07:49:18.491 [CM503003]: Call(C:8): Call to <sip:611@192.168.100.235:5060> has failed; Cause: 408 Request Timeout/INVITE from local
    12-May-2014 07:49:18.491 Session 69916 has failed in leg L:8.2[Line:10001>>611] ; Cause: 408 Request Timeout/INVITE from local
    12-May-2014 07:49:18.491 Failure from <sip:611@192.168.100.235:5060>;tag=7e096c49 to "621"<sip:621@192.168.100.235:5060>;tag=e362662d
    12-May-2014 07:48:57.820 Currently active calls - 1: [8]
    12-May-2014 07:48:46.244 [CM503025]: Call(C:8): Calling T:Line:10001>>611@[Dev:sip:621@192.168.100.235:5060] for L:8.1[Extn]
    12-May-2014 07:48:46.244 Route to L:8.2[Line:10001>>611] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=c747d776537b5e65 Call-ID=NjExOTdhYTA0OTU1YzRlZTc1Y2FjOTRmYjJiNmMwMWY.:
    INVITE sip:611@192.168.100.235:5060 SIP/2.0
    Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-c747d776537b5e65-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:621@192.168.100.194:5061>
    To: <sip:611@192.168.100.235:5060>
    From: "621"<sip:621@192.168.100.235:5060>;tag=e362662d
    Call-ID: NjExOTdhYTA0OTU1YzRlZTc1Y2FjOTRmYjJiNmMwMWY.
    CSeq: 1 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    Supported: replaces
    Content-Length: 260

    v=0
    o=3cxPS 444210348032 218523238401 IN IP4 192.168.100.194
    s=3cxPS Audio call
    c=IN IP4 192.168.100.194
    t=0 0
    m=audio 7030 RTP/AVP 8 0 3 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    12-May-2014 07:48:46.234 Outbound URI is used: sip:192.168.100.235:5060
    12-May-2014 07:48:46.234 SLA is globally disabled
    12-May-2014 07:48:46.233 Added leg L:8.2[Line:10001>>611]
    12-May-2014 07:48:46.195 [Flow] Call(C:8): making call from L:8.1[Extn] to T:Line:10001>>611@[Dev:sip:621@192.168.100.235:5060]
    12-May-2014 07:48:46.195 [CM503027]: Call(C:8): From: Extn:1000 ("Operator Extension" <sip:1000@192.168.100.194:5061>) to T:Line:10001>>611@[Dev:sip:621@192.168.100.235:5060]
    12-May-2014 07:48:46.195 [CM503004]: Call(C:8): Route 1: from L:8.1[Extn] to T:Line:10001>>611@[Dev:sip:621@192.168.100.235:5060]
    12-May-2014 07:48:46.195 Line limit check: Current # of calls for line Lc:10001(@Alcatel[<sip:621@192.168.100.235:5060>]) is 0; limit is 2
    12-May-2014 07:48:46.195 Call(C:8): Call from Extn:1000 to 611 matches outbound rule 'Rule for Sipgate (Nick) (internal 1)'
    12-May-2014 07:48:46.195 [Flow] Call(C:8): has built target endpoint: Out#:>>Rule{Rule for Sipgate (Nick) (internal 1)}>>611 for call from L:8.1[Extn]
    12-May-2014 07:48:46.195 [Flow] Target endpoint for 611 is Out#:>>Rule{Rule for Sipgate (Nick) (internal 1)}>>611
    12-May-2014 07:48:46.195 Selected prefix: 6
    12-May-2014 07:48:46.195 Looking for outbound rule: dialed = [611], processed: [611]; from-ext:
    12-May-2014 07:48:46.195 [Flow] Building target endpoint to 611 from "Operator Extension" <sip:1000@192.168.100.194:5061>
    12-May-2014 07:48:46.195 [CM503010]: Call(C:8): Making route(s) from Extn:1000 to <sip:611@192.168.100.194:5061>
    12-May-2014 07:48:46.195 Remote SDP is set for leg L:8.1[Extn]
    12-May-2014 07:48:46.195 OnOffer from "Operator Extension"<sip:1000@192.168.100.194:5061>;tag=5df8454771
    12-May-2014 07:48:46.195 [CM505001]: Endpoint Extn:1000: Device info: Device Identified: [Man: Aastra;Mod: 5x series;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, recvonly] UserAgent: [Aastra 55i/3.3.1.2235] PBX contact: [sip:1000@192.168.100.194:5061]
    12-May-2014 07:48:46.193 Inbound DID: ''; Phonebook Name: ''
    12-May-2014 07:48:46.193 [CM500002]: Call(C:8): Info on incoming INVITE from Extn:1000:
    Invite-IN Recv Req INVITE from 192.168.100.78:5060 tid=859007c0f31f1f4af.68c3cc52f9fd10929 Call-ID=41a890b05251b9cb:
    INVITE sip:611@192.168.100.194:5061;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.100.78:5060;branch=z9hG4bK859007c0f31f1f4af.68c3cc52f9fd10929
    Max-Forwards: 70
    Contact: "Operator Extension"<sip:1000@192.168.100.78:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1A30B3>"
    To: <sip:611@192.168.100.194:5061;user=phone>
    From: "Operator Extension"<sip:1000@192.168.100.194:5061>;tag=5df8454771
    Call-ID: 41a890b05251b9cb
    CSeq: 10288 INVITE
    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
    Content-Type: application/sdp
    Proxy-Authorization: Digest username="1000",realm="3CXPhoneSystem",nonce="414d535c0980ffce09:944680bb9c98cea40fafce23b7e06411",uri="sip:611@192.168.100.194:5061;user=phone",response="0f1be8154ede47871416eeb781d3bd05",algorithm=MD5
    Supported: path, 100rel, replaces
    User-Agent: Aastra 55i/3.3.1.2235
    Allow-Events: aastra-xml, talk, hold, conference, LocalModeStatus
    Content-Length: 382

    v=0
    o=MxSIP 0 1 IN IP4 192.168.100.78
    s=SIP Call
    c=IN IP4 192.168.100.78
    t=0 0
    m=audio 3000 RTP/AVP 0 8 18 9 18 0 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=silenceSupp:eek:ff - - - -
    a=fmtp:18 annexb=no
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv
    12-May-2014 07:48:46.193 [CM503001]: Call(C:8): Incoming call from Extn:1000 to <sip:611@192.168.100.194:5061>
    12-May-2014 07:48:46.184 Outbound URI is used: sip:1000@192.168.100.78:5060;transport=udp
    12-May-2014 07:48:46.184 IncomingCall: C:8 from <sip:1000@192.168.100.194:5061> to <sip:611@192.168.100.194:5061>
    12-May-2014 07:48:46.184 Added leg L:C:8.1[No endpoint yet]
    12-May-2014 07:48:46.183 UasSession 69907 started
    12-May-2014 07:48:46.183 Call from "Operator Extension"<sip:1000@192.168.100.194:5061>;tag=5df8454771 to <sip:611@192.168.100.194:5061;user=phone>;tag=8f466075
     
  5. nbrown

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    I've done a port mirror of the network interface for the phone system and performed a wireshark there too, all seems OK and the phone system is responding to the call as far as I can see, there's lots of INVITE's, a REGISTER and more INVITE's, not sure why it isn't working.
     
  6. craigreilly

    craigreilly Well-Known Member

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    Does the alcatel have a firewall blocking any traffic/ports?
     
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  7. leejor

    leejor Well-Known Member

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  8. nbrown

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    The Alcatel has no firewalls between itself and the 3CX and always works fine with any SIP device such as X-lite or Aastra phone. Not sure if there are logs we can get to on the Alcatel PBX but I will find out.

    Nick
     
  9. nbrown

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    There is no obvious logging I can find, unless someone can point me in the right direction!
     
  10. leejor

    leejor Well-Known Member

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    I would use Wireshark and compare a call attempt from 3CX with a call from a set (perhaps an ATA if you have one handy) using the same extension number. Take note of any differences, even in the initial registration.

    This just set-up as a Generic trunk, have you tried using a bridge trunk (slave)?
     
  11. nbrown

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    We did try comparing the Wireshark traces of a device that worked compared to one that didn't and although there were small differences there was nothing obvious that jumped out as being the problem. We'll try this again and maybe from a software client on the 3CX server as the comparison too. Our phone system provider have given us a quote for one of their engineers to visit and help with the diagnostics so we may have to take them up on this.

    Yes this is setup as a generic VoIP provider not a trunk, the trunk just wouldn't work at all in either direction.
     
  12. nbrown

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    We might be getting somewhere, comparing a phone that works to the phone system that doesn't it appears as though 3CX may be trying to connect to the wrong port on the Alcatel system as in the invite request it has this...

    User Datagram Protocol, Src Port: sip-tls (5061), Dst Port: 65471 (65471)

    Is there a way to force it to 5060?
     
  13. nbrown

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    OK, during registration with the Alcatel system this port 65471 appears (maybe NAT?) and I'm sure this is causing the issue. Is there any way to force registration to use port 5060, maybe using custom values in the outbound call? This also comes up in the SIP invites.

    User Datagram Protocol, Src Port: sip (5060), Dst Port: 65471 (65471)
     
  14. leejor

    leejor Well-Known Member

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    In the 3CX settings for the generic trunk you created, there should be a setting to tell 3CX the port to use at the far end (SIP Server Port). While most providers will use 5060 as a default, some don't so the ability to "customize" has to be there.
     
  15. donbru01

    donbru01 New Member

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    Try to set side trunk 3cx
    Contact user part = originator caller id
     
  16. nbrown

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    We've got it working, in the end we have set up Gateways for outgoing calls (Had to un-tick 'Only Authenticated incoming calls' on the Alcatel system to get it to work too). We set the registration to 'Do not require' and we can now make outgoing calls only via these Gateways. We then set up Providers and these were used for incoming calls only.

    It's a bit messy and you have to setup a lot of outbound rules but it's working now it allows us to test the 3CX system alongside our Alcatel Omnipcx which is what we wanted to achieve.

    Thank you all for your help.
     
  17. leejor

    leejor Well-Known Member

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    Glad you got it working. Unfortunately, integrating two PBX's, other than an interconnecting trunk for basic calls, is cumbersome at best, and a logistic nightmare at worst.

    If the situation allows, and, of course, not all do, it's best to set-up a separate system using PSTN "test numbers", with VoIP trunking, or a gateway, that can then be integrated into in the final deployment.
     
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