Incoming call issue with Gradwell Trunk

Discussion in '3CX Phone System - General' started by max1, Nov 23, 2015.

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  1. max1

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    Hi,

    I appreciate that Gradwell is not on your approved and supported VoIP provider list but unfortunately I am forced to work with this provider for the time being.

    I can make outgoing calls via the Gradwell trunk however I am unable to receive calls. I have used Wireshark on the server and see the SIP invite messages however there is no response from 3CX.

    The server is v14 Cloud on Windows 2012 R2 with public IP, therefor no NAT is involved. I suspect the issue is to do with the Inbound Parameters but the combination of things I have tried still fails.

    [​IMG]

    Any and all advice would be of great help. Below is the SIP trace from Wireshark.

    Thank you

    Max

    Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:01234567890@123.123.123.12 SIP/2.0
    Method: INVITE
    Request-URI: sip:01234567890@123.123.123.12
    Request-URI User Part: 01234567890
    Request-URI Host Part: 123.123.123.12
    [Resent Packet: True]
    [Suspected resend of frame: 1093]
    Message Header
    Via: SIP/2.0/UDP 109.224.240.232:5060;branch=z9hG4bK239134b6
    Transport: UDP
    Sent-by Address: 109.224.240.232
    Sent-by port: 5060
    Branch: z9hG4bK239134b6
    Max-Forwards: 70
    From: <sip:07123456789@sip.gradwell.net>;tag=as7d0907fd
    SIP from address: sip:07123456789@sip.gradwell.net
    SIP from address User Part: 07123456789
    SIP from address Host Part: sip.gradwell.net
    SIP from tag: as7d0907fd
    To: <sip:01234567890@123.123.123.12>
    SIP to address: sip:01234567890@123.123.123.12
    SIP to address User Part: 01234567890
    SIP to address Host Part: 123.123.123.12
    Contact: <sip:07123456789@109.224.240.232:5060>
    Contact URI: sip:07123456789@109.224.240.232:5060
    Contact URI User Part: 07123456789
    Contact URI Host Part: 109.224.240.232
    Contact URI Host Port: 5060
    Call-ID: 18ea81bc2bcc48be4c4c89961b27bd01@109.224.240.232:5060
    CSeq: 102 INVITE
    Sequence Number: 102
    Method: INVITE
    User-Agent: Asterisk PBX 11.19.0-gw0.60.1
    Date: Mon, 23 Nov 2015 16:18:41 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Session-ID: 1448295521.793233b375331bf2
    sess-id: 1448295521
    Content-Type: application/sdp
    Content-Length: 322
    Message Body
    Session Description Protocol
    Session Description Protocol Version (v): 0
    Owner/Creator, Session Id (o): root 1928609658 1928609658 IN IP4 109.224.240.232
    Owner Username: root
    Session ID: 1928609658
    Session Version: 1928609658
    Owner Network Type: IN
    Owner Address Type: IP4
    Owner Address: 109.224.240.232
    Session Name (s): Asterisk PBX 11.19.0-gw0.60.1
    Connection Information (c): IN IP4 109.224.240.232
    Connection Network Type: IN
    Connection Address Type: IP4
    Connection Address: 109.224.240.232
    Time Description, active time (t): 0 0
    Session Start Time: 0
    Session Stop Time: 0
    Media Description, name and address (m): audio 10334 RTP/AVP 8 0 18 101
    Media Type: audio
    Media Port: 10334
    Media Protocol: RTP/AVP
    Media Format: ITU-T G.711 PCMA
    Media Format: ITU-T G.711 PCMU
    Media Format: ITU-T G.729
    Media Format: DynamicRTP-Type-101
    Media Attribute (a): rtpmap:8 PCMA/8000
    Media Attribute Fieldname: rtpmap
    Media Format: 8
    MIME Type: PCMA
    Sample Rate: 8000
    Media Attribute (a): rtpmap:0 PCMU/8000
    Media Attribute Fieldname: rtpmap
    Media Format: 0
    MIME Type: PCMU
    Sample Rate: 8000
    Media Attribute (a): rtpmap:18 G729/8000
    Media Attribute Fieldname: rtpmap
    Media Format: 18
    MIME Type: G729
    Sample Rate: 8000
    Media Attribute (a): fmtp:18 annexb=no
    Media Attribute Fieldname: fmtp
    Media Format: 18 [G729]
    Media format specific parameters: annexb=no
    Media Attribute (a): rtpmap:101 telephone-event/8000
    Media Attribute Fieldname: rtpmap
    Media Format: 101
    MIME Type: telephone-event
    Sample Rate: 8000
    Media Attribute (a): fmtp:101 0-16
    Media Attribute Fieldname: fmtp
    Media Format: 101 [telephone-event]
    Media format specific parameters: 0-16
    Media Attribute (a): ptime:20
    Media Attribute Fieldname: ptime
    Media Attribute Value: 20
    Media Attribute (a): sendrecv
     
  2. JonnyM

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    Have you got your phone number listed on the next tab along, under "source identification by DID"? I have that set to To: User Part, and the Gradwell number in the box below in the non-+44 format, e.g. 01xxxxxxxxx
     
  3. jpillow

    jpillow Well-Known Member

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    Did you add the DID under source id---> source identification by DID? Also do you have the SIP trunk set to require registration for in and outbound calls? Whats your trunk set as?
     
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  4. max1

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    Hi,
    Yes I have added it under source ID and tried To:User Part and Request Line URI: User Part.
    The trunk is set to not require registration as Gradwell do not require it. You must specify the IP of the PBX in their Control Panel.
    I'm really stumped!
    Thanks
     
  5. lneblett

    lneblett Well-Known Member

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    You might take a look at the 3CX server log as well as the Security tab to see if the IP might be blocked. The log may provide more insight into the reason.
     
  6. max1

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    I'm such an idiot :roll:

    I forgot that with Cloud server the ports for inbound SIP are incremented from 5060 for instance 1 etc. Turns out all I needed to do was add 13060: to the SIP URL with the provider and boom it works!

    FML :oops:

    Lesson to us all. Read the manual!
     
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