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Yannick

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Posts posted by Yannick

  1. [7] 2011/10/13 22:11:31:	SIP Rx udp:83.143.188.165:5060:
    OPTIONS sip:62.148.184.174:5060 SIP/2.0
    Via: SIP/2.0/UDP 83.143.188.165:5060;branch=0
    From: sip:pinger@sip1.budgetphone.nl;tag=27c969fc
    To: sip:62.148.184.174:5060
    Call-ID: 8e8757a3-0b35d35c-a61179@83.143.188.165
    CSeq: 1 OPTIONS
    Content-Length: 0
    
    [7] 2011/10/13 22:11:31:	SIP Tx udp:83.143.188.165:5060:
    SIP/2.0 200 Ok
    Via: SIP/2.0/UDP 83.143.188.165:5060;branch=0
    From: <sip:pinger@sip1.budgetphone.nl>;tag=27c969fc
    To: <sip:62.148.184.174:5060>;tag=4b1ca8abcd
    Call-ID: 8e8757a3-0b35d35c-a61179@83.143.188.165
    CSeq: 1 OPTIONS
    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
    Content-Length: 0
    
    [7] 2011/10/13 22:11:56:	Hunt Group 72: Moving to next stage
    [7] 2011/10/13 22:11:56:	Hunt group 72 started 0 calls
    [7] 2011/10/13 22:11:56:	Hunt Group 72: Moving to next stage
    [7] 2011/10/13 22:11:56:	Hunt group 72 started 0 calls
    [7] 2011/10/13 22:11:56:	Hunt Group 72: Moving to next stage
    [7] 2011/10/13 22:11:56:	Hunt group 72 started 0 calls
    [7] 2011/10/13 22:11:56:	Hunt Group 72: Moving to next stage
    [7] 2011/10/13 22:11:59:	SIP Rx udp:83.143.188.165:5060:
    BYE sip:31707113070@62.148.184.174:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 83.143.188.165;branch=z9hG4bK26d1.376ea0c1.0
    Via: SIP/2.0/UDP 83.143.188.161:5060;rport=5060;received=83.143.188.161;branch=z9hG4bK886549626
    From: <sip:+31707113070@83.143.188.161;user=phone>;tag=1473184255
    To: <sip:31707113070@sip1.budgetphone.nl;user=phone>;tag=1e8fcdff21
    Call-ID: 76722836@83.143.188.161
    CSeq: 21 BYE
    Max-Forwards: 12
    Reason: Q.850 ;cause=16 ;text="Normal call clearing"
    Content-Length: 0
    
    [7] 2011/10/13 22:11:59:	SIP Tx udp:83.143.188.165:5060:
    SIP/2.0 200 Ok
    Via: SIP/2.0/UDP 83.143.188.165;branch=z9hG4bK26d1.376ea0c1.0
    Via: SIP/2.0/UDP 83.143.188.161:5060;rport=5060;received=83.143.188.161;branch=z9hG4bK886549626
    From: <sip:+31707113070@83.143.188.161;user=phone>;tag=1473184255
    To: <sip:31707113070@sip1.budgetphone.nl;user=phone>;tag=1e8fcdff21
    Call-ID: 76722836@83.143.188.161
    CSeq: 21 BYE
    Contact: <sip:31707113070@62.148.184.139:5060;transport=udp>
    User-Agent: snom-PBX/2011-4.2.1.4025
    Content-Length: 0
    
    [7] 2011/10/13 22:12:01:	SIP Rx udp:83.143.188.165:5060:
    OPTIONS sip:62.148.184.174:5060 SIP/2.0
    Via: SIP/2.0/UDP 83.143.188.165:5060;branch=0
    From: sip:pinger@sip1.budgetphone.nl;tag=b2fa69fc
    To: sip:62.148.184.174:5060
    Call-ID: 8e8757a3-9666d35c-881179@83.143.188.165
    CSeq: 1 OPTIONS
    Content-Length: 0
    
    [7] 2011/10/13 22:12:01:	SIP Tx udp:83.143.188.165:5060:
    SIP/2.0 200 Ok
    Via: SIP/2.0/UDP 83.143.188.165:5060;branch=0
    From: <sip:pinger@sip1.budgetphone.nl>;tag=b2fa69fc
    To: <sip:62.148.184.174:5060>;tag=14868ba08c
    Call-ID: 8e8757a3-9666d35c-881179@83.143.188.165
    CSeq: 1 OPTIONS
    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
    Content-Length: 0

  2. Ok, just tried that but to no avail. Added mobile number to "call forward when not registered" and to "cell phone number" and also set "when calling the extension in a hunt group" to "include cell phone in the group" but nothing happens, just redirects directly to "final stage".

  3. Try using "Not indication" or "Remote-Party-ID" from the drop-down and place the call again.

    Wow, that did the trick!

    So happy this forum and support (even for free users!) exists. I remember my early days of programming, when you could spend days figuring out 1 simple thing. Now with the help I got a jump start into this great product! thank you

  4. I'm new to this and trying to setup an IVR that will redirect calls to mobile lines on menu selection.

    I'v got the IVR working but when it should redirect to my mobile line it's giving me the default betamax error message: "sorry your call could not be connected.". I'v setup credits for the domain and a dial plan that selects the correct betamax trunk but then when redirecting it gives me this message.

    I don't know why this happens but it seems that either the forwarded number is incorrect (missing country code or something or appending headers to the address) or the redirected caller id is not accepted. I've tried many number formats (with leading 00's, +, etc) without avail.

    I don't know how to debug this and could use some help.

     

    Here is my SIP log: http://pastebin.ca/2086488

     

    Could it be that it is trying to find a user at eu.voxalot.com instead of calling the mobile number?

     

    Any help on how to debug this is very welcome.

    I tried doing more debugging by trying to call out using an iPhone sip client but then the other party could hear me but I couldn't hear anything.

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