Sip trunk not registering

Discussion in 'General' started by telecomtechnician, Aug 19, 2010.

  1. telecomtechnician

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    Hi elastix community

    My VOIP provider send me the following configuration file for my elastix server:

    SIP Configuration for Asterisk
    ;
    [general]
    port = 5060 ; Port to bind to
    bindaddr = 0.0.0.0 ; Address to bind to
    maxexpirey=60 ; Max length of incoming registration we allow
    defaultexpirey=180 ; Default length of incoming/outoing registration

    disallow=all ; Disallow all codecs
    allow=g729

    dtmfmode=rfc2833


    register => 771XXXXXXX:123456789:771XXXXXXX@ssc01.payedsip.com/771XXXXXXX

    [ssc01.payedsip.com]
    type=friend
    disallow=all
    allow=g729
    host=ssc01.payedsip.com
    username=771XXXXXXX
    fromuser=771XXXXXXX
    secret=123456789
    context=sip-in
    canreinvite=no
    nat=yes
    qualify=yes
    insecure=very



    extensions.conf:

    [sip-in]
    exten => 771XXXXXXX,1,Dial(SIP/100) ; donde 100 es una extension interna de la pbx



    How can I actually do this in elastix?,

    I tried to configure this in TRUNK, ADD SIP TRUNK but it does not work.
    My provider says that I have something missing, that is why I can not register.

    Waiting for your comments

    David Medina
     
  2. dicko

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  3. dingoland

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    Hi,

    Try to delete the second 771xxxxx (just before the @) in the string registration (as username and authname are the same) and add fromdomain=ssc01.payedsip.com in the peer details of your trunk.

    Regards
    Greg
     
  4. ramoncio

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    Hi David,

    If you haven't solved it yet, could you paste some sip debug messages?
    I have one provider that when I set qualify=yes doesn't work, but removing this line it works great.
     
  5. telecomtechnician

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    Hi Ramoncio, Here are my sip debug message

    I hope you can help me

    David Medina




    Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK709c0ca5;rport
    Max-Forwards: 70
    From: <sip:7711800883@ssc01.comvoz.com>;tag=as0983d179
    To: <sip:7711800883@ssc01.comvoz.com>
    Call-ID: 3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3
    CSeq: 105 REGISTER
    User-Agent: Elastix
    Expires: 180
    Contact: <sip:7711800883@190.206.213.117>
    Content-Length: 0


    ---
    Retransmitting #5 (NAT) to 216.22.81.36:5060:
    REGISTER sip:ssc01.comvoz.com SIP/2.0
    Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK709c0ca5;rport
    Max-Forwards: 70
    From: <sip:7711800883@ssc01.comvoz.com>;tag=as0983d179
    To: <sip:7711800883@ssc01.comvoz.com>
    Call-ID: 3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3
    CSeq: 105 REGISTER
    User-Agent: Elastix
    Expires: 180
    Contact: <sip:7711800883@190.206.213.117>
    Content-Length: 0


    ---
    Reliably Transmitting (NAT) to 216.22.81.36:5060:
    OPTIONS sip:ssc01.comvoz.com SIP/2.0
    Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK60745335;rport
    Max-Forwards: 70
    From: "Unknown" <sip:Unknown@190.206.213.117>;tag=as3806bbf9
    To: <sip:ssc01.comvoz.com>
    Contact: <sip:Unknown@190.206.213.117>
    Call-ID: 10ddcb5b4ae3a03d561ba8282720ae05@190.206.213.117
    CSeq: 102 OPTIONS
    User-Agent: Elastix
    Date: Fri, 20 Aug 2010 14:26:29 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Length: 0


    ---
    Retransmitting #1 (NAT) to 216.22.81.36:5060:
    OPTIONS sip:ssc01.comvoz.com SIP/2.0
    Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK60745335;rport
    Max-Forwards: 70
    From: "Unknown" <sip:Unknown@190.206.213.117>;tag=as3806bbf9
    To: <sip:ssc01.comvoz.com>
    Contact: <sip:Unknown@190.206.213.117>
    Call-ID: 10ddcb5b4ae3a03d561ba8282720ae05@190.206.213.117
    CSeq: 102 OPTIONS
    User-Agent: Elastix
    Date: Fri, 20 Aug 2010 14:26:29 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Length: 0


    ---
    Retransmitting #2 (NAT) to 216.22.81.36:5060:
    OPTIONS sip:ssc01.comvoz.com SIP/2.0
    Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK60745335;rport
    Max-Forwards: 70
    From: "Unknown" <sip:Unknown@190.206.213.117>;tag=as3806bbf9
    To: <sip:ssc01.comvoz.com>
    Contact: <sip:Unknown@190.206.213.117>
    Call-ID: 10ddcb5b4ae3a03d561ba8282720ae05@190.206.213.117
    CSeq: 102 OPTIONS
    User-Agent: Elastix
    Date: Fri, 20 Aug 2010 14:26:29 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Length: 0


    ---
    Retransmitting #3 (NAT) to 216.22.81.36:5060:
    OPTIONS sip:ssc01.comvoz.com SIP/2.0
    Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK60745335;rport
    Max-Forwards: 70
    From: "Unknown" <sip:Unknown@190.206.213.117>;tag=as3806bbf9
    To: <sip:ssc01.comvoz.com>
    Contact: <sip:Unknown@190.206.213.117>
    Call-ID: 10ddcb5b4ae3a03d561ba8282720ae05@190.206.213.117
    CSeq: 102 OPTIONS
    User-Agent: Elastix
    Date: Fri, 20 Aug 2010 14:26:29 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Length: 0


    ---
    Retransmitting #6 (NAT) to 216.22.81.36:5060:
    REGISTER sip:ssc01.comvoz.com SIP/2.0
    Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK709c0ca5;rport
    Max-Forwards: 70
    From: <sip:7711800883@ssc01.comvoz.com>;tag=as0983d179
    To: <sip:7711800883@ssc01.comvoz.com>
    Call-ID: 3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3
    CSeq: 105 REGISTER
    User-Agent: Elastix
    Expires: 180
    Contact: <sip:7711800883@190.206.213.117>
    Content-Length: 0


    ---
    Retransmitting #4 (NAT) to 216.22.81.36:5060:
    OPTIONS sip:ssc01.comvoz.com SIP/2.0
    Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK60745335;rport
    Max-Forwards: 70
    From: "Unknown" <sip:Unknown@190.206.213.117>;tag=as3806bbf9
    To: <sip:ssc01.comvoz.com>
    Contact: <sip:Unknown@190.206.213.117>
    Call-ID: 10ddcb5b4ae3a03d561ba8282720ae05@190.206.213.117
    CSeq: 102 OPTIONS
    User-Agent: Elastix
    Date: Fri, 20 Aug 2010 14:26:29 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Length: 0


    ---
    Really destroying SIP dialog '10ddcb5b4ae3a03d561ba8282720ae05@190.206.213.117' Method: OPTIONS
    > doing dnsmgr_lookup for 'ssc01.comvoz.com'
    REGISTER 10 headers, 0 lines
    Reliably Transmitting (NAT) to 216.22.81.36:5060:
    REGISTER sip:ssc01.comvoz.com SIP/2.0
    Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK5101745f;rport
    Max-Forwards: 70
    From: <sip:7711800883@ssc01.comvoz.com>;tag=as23b213e2
    To: <sip:7711800883@ssc01.comvoz.com>
    Call-ID: 3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3
    CSeq: 106 REGISTER
    User-Agent: Elastix
    Expires: 180
    Contact: <sip:7711800883@190.206.213.117>
    Content-Length: 0


    ---
    Really destroying SIP dialog '3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3' Method: REGISTER
    Retransmitting #1 (NAT) to 216.22.81.36:5060:
    REGISTER sip:ssc01.comvoz.com SIP/2.0
    Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK5101745f;rport
    Max-Forwards: 70
    From: <sip:7711800883@ssc01.comvoz.com>;tag=as23b213e2
    To: <sip:7711800883@ssc01.comvoz.com>
    Call-ID: 3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3
    CSeq: 106 REGISTER
    User-Agent: Elastix
    Expires: 180
    Contact: <sip:7711800883@190.206.213.117>
    Content-Length: 0


    ---
    Retransmitting #2 (NAT) to 216.22.81.36:5060:
    REGISTER sip:ssc01.comvoz.com SIP/2.0
    Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK5101745f;rport
    Max-Forwards: 70
    From: <sip:7711800883@ssc01.comvoz.com>;tag=as23b213e2
    To: <sip:7711800883@ssc01.comvoz.com>
    Call-ID: 3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3
    CSeq: 106 REGISTER
    User-Agent: Elastix
    Expires: 180
    Contact: <sip:7711800883@190.206.213.117>
    Content-Length: 0


    ---
    Retransmitting #3 (NAT) to 216.22.81.36:5060:
    REGISTER sip:ssc01.comvoz.com SIP/2.0
    Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK5101745f;rport
    Max-Forwards: 70
    From: <sip:7711800883@ssc01.comvoz.com>;tag=as23b213e2
    To: <sip:7711800883@ssc01.comvoz.com>
    Call-ID: 3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3
    CSeq: 106 REGISTER
    User-Agent: Elastix
    Expires: 180
    Contact: <sip:7711800883@190.206.213.117>
    Content-Length: 0


    ---
    Reliably Transmitting (NAT) to 216.22.81.36:5060:
    OPTIONS sip:ssc01.comvoz.com SIP/2.0
    Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK61ffa6db;rport
    Max-Forwards: 70
    From: "Unknown" <sip:Unknown@190.206.213.117>;tag=as4d500399
    To: <sip:ssc01.comvoz.com>
    Contact: <sip:Unknown@190.206.213.117>
    Call-ID: 0337e27052ddc61e060ddaa90a5dddc8@190.206.213.117
    CSeq: 102 OPTIONS
    User-Agent: Elastix
    Date: Fri, 20 Aug 2010 14:26:43 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Length: 0


    ---
    Retransmitting #1 (NAT) to 216.22.81.36:5060:
    OPTIONS sip:ssc01.comvoz.com SIP/2.0
    Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK61ffa6db;rport
    Max-Forwards: 70
    From: "Unknown" <sip:Unknown@190.206.213.117>;tag=as4d500399
    To: <sip:ssc01.comvoz.com>
    Contact: <sip:Unknown@190.206.213.117>
    Call-ID: 0337e27052ddc61e060ddaa90a5dddc8@190.206.213.117
    CSeq: 102 OPTIONS
    User-Agent: Elastix
    Date: Fri, 20 Aug 2010 14:26:43 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Length: 0


    ---
    Retransmitting #4 (NAT) to 216.22.81.36:5060:
    REGISTER sip:ssc01.comvoz.com SIP/2.0
    Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK5101745f;rport
    Max-Forwards: 70
    From: <sip:7711800883@ssc01.comvoz.com>;tag=as23b213e2
    To: <sip:7711800883@ssc01.comvoz.com>
    Call-ID: 3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3
    CSeq: 106 REGISTER
    User-Agent: Elastix
    Expires: 180
    Contact: <sip:7711800883@190.206.213.117>
    Content-Length: 0
     
  6. telecomtechnician

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    Hi Greg

    Thank you for your help, but it did not work. I did not mention (I am doing it now) that there was a moment when the trunks registered with no problem. At nights I turned off the voip server, and the next day I turned it on again, there was days when there was no trunk registration, other days the trunks registered with no problem. But from a month ago, the problem is permanent, no trunk registration.

    Waiting for your comments

    David Medina
     
  7. telecomtechnician

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    Re:Sip trunk not registering (SOLVED)

    Hi elastix community

    The problem was solved in the following way:

    My router´s state tables had to be flushed. Once it was done, (resetting the router to default), the trunks registered with the sip provider with no problem.

    Last night I got into the elastix IRC chat, and somebody gave me the solution. Definetely, network acknowledge is very important.

    I have to be aware of this, because I turn off all my telecom equipment at night. Or maybe I might not turn off the router and internet modem.

    Thanks everybody

    David Medina
     
  8. alfreluis

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    hi David, u know...I have the same debug figure (retransmitting...etc) and my sip trunk cannot registering in to my sip VoIP provider...

    What's the router you have reset? is it the external router, right?

    Thank you very much
    Alfred
     
  9. jmvelazquezmx

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    Trunk not answering

    Hi all:

    I am newbie in elastix, after installing release 1.6.5.2, added my sip extensions (both hardware and sip), was able to comunicate among them; however, when trying to make outbound calls I keep receiving the same message "all our circuitries are busy at this moment. I am using Digium 2 ports. Any body have any ideas?

    Thanks in advance
     
  10. trymes

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    Re: Trunk not answering

    JM: Two things:

    1.) You should create a new topic with your question, ad it is not related to this topic, and you'll likely get a better response if you start your own thread.
    2.) It sounds as if you have not created an outbound route to tell your system how to send calls out to the PSTN. Configure an outbound route or verify the settings on your existing one and see if that helps.

    Tom
     
  11. jmvelazquezmx

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    Re: Trunk not answering

    Thanks for your response. Outbound route is the one that comes by default. No chenges were made to this section. I will open this new topic.
     

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