4PSA sip trunk

Discussion in 'General' started by itche, May 3, 2010.

  1. itche

    Joined:
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    Hello all,

    I am trying to set a 4PSA sip trunk.

    I have no problem to call out using this trunk but I can't get it to work for incomming calls.
    below is the settings I used for this trunk

    PEER Details
    host=71.125.xxx.xx
    username=00xx*xxx
    secret=xxxxxxxxx
    type=peer

    USER details
    secret=xxxxxxxxx
    type=user
    context=from-trunk

    Register String
    00xx*xxx:xxxxxxx@71.125.xxx.xxx/00xx*xxx

    when looking in the 4PSA server I can see that elastix has a "register" status.

    Anyone has any ida on how to make this work?

    Thanks
    Itche
     
  2. Patrick_elx

    Joined:
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    log in to your server
    type asterisk -rvvv
    then sip set debug

    and try to receive a call to see what's happening.
     
  3. itche

    Joined:
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    Hi Patrick,
    Here is the output for that session
    I have massked few private details
    Thanks
    Itche


    <--- SIP read from 71.125.60.205:5060 --->
    INVITE sip:15164908048@24.44.140.131 SIP/2.0
    Record-Route: <sip:71.125.60.205;r2=on;lr=on;ftag=as723f857a;did=923.dfa8f802>
    Record-Route: <sip:10.15.1.50;r2=on;lr=on;ftag=as723f857a;did=923.dfa8f802>
    Via: SIP/2.0/UDP 71.125.60.205:5060;branch=z9hG4bKccdd.a75e7661.0
    Max-Forwards: 69
    To: <sip:15164908048@10.15.1.50>
    From: "Cell Phone CT "<sip:12032578412@71.125.60.205>;tag=as723f857a
    Call-ID: 1dbd260b162f39c436668fb90a474df5@10.15.1.50
    Contact: <sip:12032578412@71.125.60.205:5060>
    CSeq: 102 INVITE
    User-Agent: VoipNow PBX
    Date: Tue, 04 May 2010 12:58:17 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces, timer
    X-voipnow-did: 15164908048
    X-voipnow-extension: 00xx*xxx <-- my extn on 4PSA
    X-voipnow-infrastructureid: c8497ed4ff2c
    X-voipnow-did: 15164908048
    Content-Type: application/sdp
    Content-Length: 360

    v=0
    o=root 476229972 476229972 IN IP4 71.125.60.205
    s=Asterisk PBX 1.6.1.4
    c=IN IP4 71.125.60.205
    t=0 0
    m=audio 15956 RTP/AVP 0 8 3 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:eek:ff - - - -
    a=ptime:20
    a=sendrecv
    a=oldmediaip:10.15.1.50
    a=oldmediaip:10.15.1.50

    <------------->
    --- (20 headers 16 lines) ---
    Sending to 71.125.60.205 : 5060 (no NAT)
    Using INVITE request as basis request - 1dbd260b162f39c436668fb90a474df5@10.15.1.50
    Found peer 'xxxxxxxxxxxxx' <-- my peer on 4PSA

    <--- Reliably Transmitting (no NAT) to 71.125.60.205:5060 --->
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 71.125.60.205:5060;branch=z9hG4bKccdd.a75e7661.0;received=71.125.60.205
    From: "Cell Phone CT "<sip:12032578412@71.125.60.205>;tag=as723f857a
    To: <sip:15164908048@10.15.1.50>;tag=as794729fd
    Call-ID: 1dbd260b162f39c436668fb90a474df5@10.15.1.50
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="795641ef"
    Content-Length: 0

    <------------>
    Scheduling destruction of SIP dialog '1dbd260b162f39c436668fb90a474df5@10.15.1.50' in 32000 ms (Method: INVITE)
    tdpbx*CLI>
    <--- SIP read from 71.125.60.205:5060 --->
    ACK sip:15164908048@24.44.140.131 SIP/2.0
    Via: SIP/2.0/UDP 71.125.60.205:5060;branch=z9hG4bKccdd.a75e7661.0
    From: "Cell Phone CT "<sip:12032578412@71.125.60.205>;tag=as723f857a
    Call-ID: 1dbd260b162f39c436668fb90a474df5@10.15.1.50
    To: <sip:15164908048@10.15.1.50>;tag=as794729fd
    CSeq: 102 ACK
    Max-Forwards: 70
    User-Agent: VoipNow PBX
    Content-Length: 0


    <------------->
    --- (9 headers 0 lines) ---
    tdpbx*CLI>
    <--- SIP read from 71.125.60.205:5060 --->
    INVITE sip:15164908048@24.44.140.131 SIP/2.0
    Record-Route: <sip:71.125.60.205;r2=on;lr=on;ftag=as723f857a;did=923.efa8f802>
    Record-Route: <sip:10.15.1.50;r2=on;lr=on;ftag=as723f857a;did=923.efa8f802>
    Via: SIP/2.0/UDP 71.125.60.205:5060;branch=z9hG4bKdcdd.f63a4141.0
    Max-Forwards: 69
    To: <sip:15164908048@10.15.1.50>
    From: "Cell Phone CT "<sip:12032578412@71.125.60.205>;tag=as723f857a
    Call-ID: 1dbd260b162f39c436668fb90a474df5@10.15.1.50
    Contact: <sip:12032578412@71.125.60.205:5060>
    CSeq: 103 INVITE
    User-Agent: VoipNow PBX
    Proxy-Authorization: Digest username="a2dc4316c1", realm="asterisk", algorithm=MD5, uri="sip:0051*001@10.15.1.50", <-- my extn on 4PSA (itche)
    nonce="795641ef", response="927a25c7f9259e8ee811cc8089ccbbcf"
    Date: Tue, 04 May 2010 12:58:17 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces, timer
    X-voipnow-did: 15164908048
    X-voipnow-extension: 00xx*xxx <-- my extn on 4PSA
    X-voipnow-infrastructureid: c8497ed4ff2c
    X-voipnow-did: 15164908048
    Content-Type: application/sdp
    Content-Length: 360

    v=0
    o=root 476229972 476229973 IN IP4 71.125.60.205
    s=Asterisk PBX 1.6.1.4
    c=IN IP4 71.125.60.205
    t=0 0
    m=audio 15956 RTP/AVP 0 8 3 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:eek:ff - - - -
    a=ptime:20
    a=sendrecv
    a=oldmediaip:10.15.1.50
    a=oldmediaip:10.15.1.50

    <------------->
    --- (21 headers 16 lines) ---
    Sending to 71.125.60.205 : 5060 (no NAT)
    Using INVITE request as basis request - 1dbd260b162f39c436668fb90a474df5@10.15.1.50
    Found peer 'xxxxxxxxxxxxxx' <-- my PEER on 4PSA (itche)


    <--- Reliably Transmitting (no NAT) to 71.125.60.205:5060 --->
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP 71.125.60.205:5060;branch=z9hG4bKdcdd.f63a4141.0;received=71.125.60.205
    From: "Cell Phone CT "<sip:12032578412@71.125.60.205>;tag=as723f857a
    To: <sip:15164908048@10.15.1.50>;tag=as794729fd
    Call-ID: 1dbd260b162f39c436668fb90a474df5@10.15.1.50
    CSeq: 103 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog '1dbd260b162f39c436668fb90a474df5@10.15.1.50' in 32000 ms (Method: INVITE)
    Really destroying SIP dialog '23a261473b610b2e10dd17972518b7a0@24.44.140.131' Method: REGISTER
    tdpbx*CLI>
    <--- SIP read from 71.125.60.205:5060 --->
    ACK sip:15164908048@24.44.140.131 SIP/2.0
    Via: SIP/2.0/UDP 71.125.60.205:5060;branch=z9hG4bKdcdd.f63a4141.0
    From: "Cell Phone CT "<sip:12032578412@71.125.60.205>;tag=as723f857a
    Call-ID: 1dbd260b162f39c436668fb90a474df5@10.15.1.50
    To: <sip:15164908048@10.15.1.50>;tag=as794729fd
    CSeq: 103 ACK
    Max-Forwards: 70
    User-Agent: VoipNow PBX
    Content-Length: 0


    <------------->
    --- (9 headers 0 lines) ---
    tdpbx*CLI>
    <--- SIP read from 71.125.60.205:5060 --->
    OPTIONS sip:00xx*xxx@24.44.140.131 SIP/2.0 <-- my extn on 4PSA
    Via: SIP/2.0/UDP 10.15.1.50:5060;branch=z9hG4bK2083.0.0
    From: sip:voipnow@10.15.1.50;tag=032b5f27
    To: sip:00xx*xxx@24.44.140.131 <-- my extn on 4PSA (itche)
    Call-ID: de20f9b6-56878d17-4ba41@10.15.1.50
    CSeq: 1 OPTIONS
    Content-Length: 0


    <------------->
    --- (7 headers 0 lines) ---
    Looking for 00xx*xxx in from-sip-external (domain 24.44.140.131) <-- my extn on 4PSA (itche)

    <--- Transmitting (no NAT) to 71.125.60.205:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.15.1.50:5060;branch=z9hG4bK2083.0.0;received=71.125.60.205
    From: sip:voipnow@10.15.1.50;tag=032b5f27
    To: sip:00xx*xxx@24.44.140.131;tag=as3ab2542b <-- my extn on 4PSA (itche)

    Call-ID: de20f9b6-56878d17-4ba41@10.15.1.50
    CSeq: 1 OPTIONS
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Contact: <sip:24.44.140.131>
    Accept: application/sdp
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog 'de20f9b6-56878d17-4ba41@10.15.1.50' in 32000 ms (Method: OPTIONS)
    Really destroying SIP dialog '10387e40-8401a8c0-13c4-4be019cc-16bdafc6-4be019cc' Method: REGISTER
     
  4. giany

    Joined:
    May 17, 2010
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    What version of VoipNow are you using? If its 2.X then it means VoipNow sip proxy denies connections by giving this message :"SIP/2.0 403 Forbidden". Edit /etc/kamailio/kamailio.cfg and change from
    debug=1
    to
    debug=2

    After that restart kamailio and check /var/log/kamailio.cfg
     
  5. itche

    Joined:
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    Hi all,
    Problem solved.:)
    All I needed to do was to remove the username and secret fields from the PEER and the user's context.

    wireshark helped alot

    Thanks for your help
    Itche
     

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