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Forums » Asterisk Support » rejected because extension not found

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Author Topic: rejected because extension not found 818 Views
  • mogambo
    avatar
    Community Member
    2 posts

    rejected because extension not found Link to this post

    I recently installed asterisk and trying to get it working. My X-Lite phones are registered but i cant make calls between two phones. I only modified sip.conf & extensions.conf do i need to modify any other any other file as well.
    [root@asterisk1 asterisk]# cat /etc/asterisk/extensions.conf
    [globals]
    PHONE1=SIP/1234
    PHONE2=SIP/1111

    [macro-oneline]
    exten => s,1,Dial(${ARG1},20,t)
    exten => s,2,Voicemail(u${MACRO_EXTEN})
    exten => s,3,Hangup
    exten => s,102,Voicemail(b${MACRO_EXTEN})
    exten => s,103,Hangup

    [local]
    exten => 1234,1,Macro(oneline,${PHONE1})
    exten => 1111,1,Macro(oneline,${PHONE2})

    Sip.conf
    [PHONE1]
    type=friend
    context=default
    regexten=1234
    callerid="User1" <1234>
    host=dynamic
    disallow=all
    allow=ulaw

    [PHONE2]
    type=friend
    context=default
    regexten=1111
    callerid="User2" <1111>
    host=dynamic
    disallow=all
    allow=ulaw

    I am getting the following error:
    [Jul 7 16:54:52] NOTICE[27657] chan_sip.c: Call from 'PHONE1' to extension '1111' rejected because extension not found.

    SIP Debug:

    --- SIP read from UDP://10.16.103.244:35672 --->
    INVITE sip:1111@10.16.103.173 SIP/2.0
    Via: SIP/2.0/UDP 10.16.103.244:35672;branch=z9hG4bK-d8754z-443f1d4698659d5a-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:PHONE1@10.16.103.244:35672>
    To: "1111"<sip:1111@10.16.103.173>
    From: "1234"<sip:PHONE1@10.16.103.173>;tag=5c117172
    Call-ID: ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.
    CSeq: 1 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
    Content-Type: application/sdp
    User-Agent: X-Lite release 1103d stamp 53117
    Content-Length: 267

    v=0
    o=- 6 2 IN IP4 10.16.103.244
    s=CounterPath X-Lite 3.0
    c=IN IP4 10.16.103.244
    t=0 0
    m=audio 26956 RTP/AVP 107 0 8 101
    a=alt:1 1 : RbbV/Xv3 90+Yl1m3 10.16.103.244 26956
    a=fmtp:101 0-15
    a=rtpmap:107 BV32/16000
    a=rtpmap:101 telephone-event/8000
    a=sendrecv

    <------------->
    --- (12 headers 11 lines) ---
    == Using SIP RTP CoS mark 5
    Sending to 10.16.103.244 : 35672 (NAT)
    Using INVITE request as basis request - ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.
    Found peer 'PHONE1' for 'PHONE1' from 10.16.103.244:35672
    Found RTP audio format 107
    Found RTP audio format 0
    Found RTP audio format 8
    Found RTP audio format 101
    Peer audio RTP is at port 10.16.103.244:26956
    Found unknown media description format BV32 for ID 107
    Found audio description format telephone-event for ID 101
    Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    Peer audio RTP is at port 10.16.103.244:26956
    Looking for 1111 in default (domain 10.16.103.173)

    <--- Reliably Transmitting (no NAT) to 10.16.103.244:35672 --->
    SIP/2.0 404 Not Found
    Via: SIP/2.0/UDP 10.16.103.244:35672;branch=z9hG4bK-d8754z-443f1d4698659d5a-1---d8754z-;received=10.16.103.244;rport=35672
    From: "1234"<sip:PHONE1@10.16.103.173>;tag=5c117172
    To: "1111"<sip:1111@10.16.103.173>;tag=as1dc2c5a7
    Call-ID: ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.
    CSeq: 1 INVITE
    Server: Asterisk PBX 1.6.1.1
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces, timer
    Content-Length: 0

    <------------>
    [Jul 7 18:31:22] NOTICE[27657]: chan_sip.c:18160 handle_request_invite: Call from 'PHONE1' to extension '1111' rejected because ext
    ension not found.
    Scheduling destruction of SIP dialog 'ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.' in 32000 ms (Method: INVITE)

    <--- SIP read from UDP://10.16.103.244:35672 --->
    ACK sip:1111@10.16.103.173 SIP/2.0
    Via: SIP/2.0/UDP 10.16.103.244:35672;branch=z9hG4bK-d8754z-443f1d4698659d5a-1---d8754z-;rport
    To: "1111"<sip:1111@10.16.103.173>;tag=as1dc2c5a7
    From: "1234"<sip:PHONE1@10.16.103.173>;tag=5c117172
    Call-ID: ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.
    CSeq: 1 ACK
    Content-Length: 0

    <------------->
    --- (7 headers 0 lines) ---
    Really destroying SIP dialog 'ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.' Method: ACK

  • Rais Ali
    avatar
    Community Member
    3 posts

    Re: rejected because extension not found Link to this post

    Dear

    It is easy to start with Web GUI of asterisk server that is FreePBX....using that you dont have to manuly configuration into file but configure it through GUI interface

  • astpanj
    avatar
    Community Member
    2 posts

    Re: rejected because extension not found Link to this post

    Sip.conf
    [PHONE1]
    type=friend
    context=default
    regexten=1234
    callerid="User1" <1234>
    host=dynamic
    disallow=all
    allow=ulaw

    Context default does not exist in your extensions.conf so change it to context=local. If it does not work you may come back here.

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