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Thread: Anyone use a virtual number with Asterisk ?

  1. #1
    Join Date
    Dec 2008
    Location
    Bay Area
    Posts
    31

    Default Anyone use a virtual number with Asterisk ?

    I have a virtual number and when I get an incoming call, I am getting two INVITE messages. Voipo sends a cancel for the first INVITE and is sending a second.

    <--- SIP read from 174.37.45.134:5060 --->
    INVITE sip:4085555555@79.23.170.73 SIP/2.0^M
    Record-Route: <sip:174.37.45.134;lr=on;ftag=VPSF506071629460>^ M
    Via: SIP/2.0/UDP 174.37.45.134;branch=z9hG4bKdbc5.7c6816c4.0^M
    f: "AnonUser" <sip:2015555555@4.68.250.148>;tag=VPSF506071629460 ^M
    t: <sip:+12125555555@67.231.8.93:5060>^M
    i: FOOBAR@209.244.63.25^M
    CSeq: 1 INVITE^M
    m: <sip:+12015555555@4.68.250.148:5060;transport=udp; nat=yes>^M
    Max-Forwards: 64^M
    c: application/sdp^M
    Content-Length: 285^M
    P-Asserted-Identity: "AnonUser" <sip:2015555555@pstn>^

    <--- Transmitting (NAT) to 174.37.45.134:5060 --->
    SIP/2.0 100 Trying^M
    Via: SIP/2.0/UDP 174.37.45.134;branch=z9hG4bKdbc5.7c6816c4.0;receiv ed=174.37.45.134^M
    Record-Route: <sip:174.37.45.134;lr=on;ftag=VPSF506071629460>^ M
    From: "AnonUser" <sip:2015555555@4.68.250.148>;tag=VPSF506071629460 ^M
    To: <sip:+12125555555@67.231.8.93:5060>^M
    Call-ID: FOOBAR@209.244.63.25^M
    CSeq: 1 INVITE^M
    User-Agent: Asterisk PBX^M
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M
    Supported: replaces, timer^M
    Contact: <sip:4085555555@79.23.160.53>^M
    Content-Length: 0^

    Asterisk processes the call

    <--- SIP read from 174.37.45.134:5060 --->
    CANCEL sip:4085555555@79.29.140.37 SIP/2.0^M
    Via: SIP/2.0/UDP 174.37.45.134;branch=z9hG4bKdbc5.7c6816c4.0^M
    From: "AnonUser" <sip:+12015555555@4.68.250.148>;tag=VPSF5060716294 60^M
    Call-ID: FOOBAR@209.244.63.25^M
    To: <sip:+12125555555@67.231.8.93:5060>^M
    CSeq: 1 CANCEL^M
    Max-Forwards: 70^M
    User-Agent: Kamailio (1.4.3-notls (x86_64/linux))^M
    Content-Length: 0

    <--- Reliably Transmitting (NAT) to 174.37.45.134:5060 --->
    SIP/2.0 487 Request Terminated^M
    Via: SIP/2.0/UDP 174.37.45.134;branch=z9hG4bKdbc5.7c6816c4.0;receiv ed=174.37.45.134^M
    From: "AnonUser" <sip:2015555555@4.68.250.148>;tag=VPSF506071629460 ^M
    To: <sip:+12125555555@67.231.8.93:5060>;tag=as47f04e2b ^M
    Call-ID: FOOBAR@209.244.63.25^M
    CSeq: 1 INVITE^M
    User-Agent: Asterisk PBX^M
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M
    Supported: replaces, timer^M
    Content-Length: 0

    <--- SIP read from 174.37.45.134:5060 --->
    ACK sip:4085555555@79.29.150.39 SIP/2.0^M
    Via: SIP/2.0/UDP 174.37.45.134;branch=z9hG4bKdbc5.7c6816c4.0^M
    From: "AnonUser" <sip:+12015555555@4.68.250.148>;tag=VPSF5060716294 60^M
    Call-ID: FOOBAR@209.244.63.25^M
    To: <sip:+12125555555@67.231.8.93:5060>;tag=as47f04e2b ^M
    CSeq: 1 ACK^M
    Max-Forwards: 70^M
    User-Agent: Kamailio (1.4.3-notls (x86_64/linux))^M
    Content-Length: 0

    <--- SIP read from 174.37.45.134:5060 --->
    INVITE sip:4085555555@79.28.129.39 SIP/2.0^M
    Record-Route: <sip:174.37.45.134;lr=on;ftag=VPSF506071629460>^ M
    Allow: INVITE,ACK,CANCEL,BYE^M
    Via: SIP/2.0/UDP 174.37.45.134;branch=z9hG4bKdbc5.9c6816c4.0^M
    f: "AnonUser" <sip:2015555555@4.68.250.148>;tag=VPSF506071629460 ^M
    t: <sip:+12125555555@67.231.8.93:5060>^M
    i: FOOBAR@209.244.63.25^M
    CSeq: 1 INVITE^M
    m: <sip:+12015555555@4.68.250.148:5060;transport=udp; nat=yes>^M
    Max-Forwards: 64^M
    c: application/sdp^M
    Content-Length: 285^M
    P-Asserted-Identity: "AnonUser" <sip:2015555555@pstn>

  2. #2
    Join Date
    Mar 2011
    Posts
    27

    Default Re: Anyone use a virtual number with Asterisk ?

    What's the time between the trying out and the cancel in? Did asterisk ever send a 180 ringing?

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