[asterisk-users] outgoing calls

Josu Lazkano Lete jlazkano at somesi.com
Tue May 8 05:52:05 MST 2007


thank you very much!!!!!

it works!!!!
  ----- Original Message ----- 
  From: Dijkstra, Roelof 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Tuesday, May 08, 2007 1:53 PM
  Subject: RE: [asterisk-users] outgoing calls


  Hello Josu,

  In you're sip.conf you have the 2 phones configured that they are in the SOME context.

  Looking at the SOME contect in extensions.conf you only have the 2 phones defined. If you want to call ouside from the SOME context as well, you need to include the outgoing context there as well.

  Regards, 

  Roelof Dijkstra 
  Network Engineer EMEA 
  Compuware Europe BV 

    -----Original Message-----
    From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Josu Lazkano Lete
    Sent: Tuesday, May 08, 2007 1:36 PM
    To: asterisk-users at lists.digium.com
    Subject: [asterisk-users] outgoing calls


    hello friends, I have a problem when I call to outside (9XXXXXXXX) from IPs Telephones.

    the incomning calls are OK.

    in the console when I put "sip debug peer 101" I have this lines:

    *CLI> sip debug peer 101
    SIP Debugging Enabled for IP: 10.0.0.9:5060
    *CLI>
    <-- SIP read from 10.0.0.9:5060:
    INVITE sip:943833473 at 101 SIP/2.0
    Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport
    From: 101 <sip:101 at 101>;tag=3159122210
    To: "943833473" <sip:943833473 at 101>
    Call-ID: 2591723875-241385386 at 10.0.0.9
    CSeq: 1 INVITE
    Contact: <sip:101 at 10.0.0.9:5060>
    max-forwards: 70
    supported: 100rel
    user-agent:
    Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE
    Content-Type: application/sdp
    Content-Length: 278

    v=0
    o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9
    s=A conversation
    c=IN IP4 10.0.0.9
    t=0 0
    m=audio 10010 RTP/AVP 18 4 4 8 0
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:4 G723/8000
    a=rtpmap:4 G723high/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=sendrecv

    --- (13 headers 13 lines) ---
    Using INVITE request as basis request - 2591723875-241385386 at 10.0.0.9
    Sending to 10.0.0.9 : 5060 (NAT)
    Reliably Transmitting (no NAT) to 10.0.0.9:5060:
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;received=10.0.0.9;rport=5060
    From: 101 <sip:101 at 101>;tag=3159122210
    To: "943833473" <sip:943833473 at 101>;tag=as705b7b72
    Call-ID: 2591723875-241385386 at 10.0.0.9
    CSeq: 1 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5a68d228"
    Content-Length: 0


    ---
    Scheduling destruction of call '2591723875-241385386 at 10.0.0.9' in 15000 ms
    Found user '101'

    <-- SIP read from 10.0.0.9:5060:
    ACK sip:943833473 at 101 SIP/2.0
    Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport
    From: 101 <sip:101 at 101>;tag=3159122210
    To: "943833473" <sip:943833473 at 101>;tag=as705b7b72
    Call-ID: 2591723875-241385386 at 10.0.0.9
    CSeq: 1 ACK
    max-forwards: 70
    Content-Length: 0


    --- (8 headers 0 lines) ---

    <-- SIP read from 10.0.0.9:5060:
    INVITE sip:943833473 at 101 SIP/2.0
    Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport
    From: 101 <sip:101 at 101>;tag=3159122210
    To: "943833473" <sip:943833473 at 101>
    Call-ID: 2591723875-241385386 at 10.0.0.9
    CSeq: 2 INVITE
    Contact: <sip:101 at 10.0.0.9:5060>
    Proxy-Authorization: Digest username="101", realm="asterisk", nonce="5a68d228", uri="sip:943833473 at 101", response="0fa2c9d246b54d137046e23743e1951c", algorithm=MD5
    max-forwards: 70
    supported: 100rel
    user-agent:
    Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE
    Content-Type: application/sdp
    Content-Length: 278

    v=0
    o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9
    s=A conversation
    c=IN IP4 10.0.0.9
    t=0 0
    m=audio 10010 RTP/AVP 18 4 4 8 0
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:4 G723/8000
    a=rtpmap:4 G723high/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=sendrecv

    --- (14 headers 13 lines) ---
    Using INVITE request as basis request - 2591723875-241385386 at 10.0.0.9
    Sending to 10.0.0.9 : 5060 (NAT)
    Found user '101'
    Found RTP audio format 18
    Found RTP audio format 4
    Found RTP audio format 4
    Found RTP audio format 8
    Found RTP audio format 0
    Peer audio RTP is at port 10.0.0.9:10010
    Found description format G729
    Found description format G723
    Found description format G723high
    Found description format PCMA
    Found description format PCMU
    Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
    Looking for 943833473 in SOME (domain 101)
    Reliably Transmitting (no NAT) to 10.0.0.9:5060:
    SIP/2.0 404 Not Found
    Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;received=10.0.0.9;rport=5060
    From: 101 <sip:101 at 101>;tag=3159122210
    To: "943833473" <sip:943833473 at 101>;tag=as705b7b72
    Call-ID: 2591723875-241385386 at 10.0.0.9
    CSeq: 2 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Length: 0


    ---

    <-- SIP read from 10.0.0.9:5060:
    ACK sip:943833473 at 101 SIP/2.0
    Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport
    From: 101 <sip:101 at 101>;tag=3159122210
    To: "943833473" <sip:943833473 at 101>;tag=as705b7b72
    Call-ID: 2591723875-241385386 at 10.0.0.9
    CSeq: 2 ACK
    Proxy-Authorization: Digest username="101", realm="asterisk", nonce="5a68d228", uri="sip:943833473 at 101", response="0fa2c9d246b54d137046e23743e1951c", algorithm=MD5
    max-forwards: 70
    Content-Length: 0


    --- (9 headers 0 lines) ---
    Destroying call '2591723875-241385386 at 10.0.0.9'
    Destroying call '44721582-00111639 at 10.0.0.9'
    12 headers, 0 lines
    Reliably Transmitting (no NAT) to 10.0.0.9:5060:
    OPTIONS sip:101 at 10.0.0.9:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK65c2f229;rport
    From: "asterisk" <sip:asterisk at 10.0.0.7>;tag=as0cc11f28
    To: <sip:101 at 10.0.0.9:5060>
    Contact: <sip:asterisk at 10.0.0.7>
    Call-ID: 5e0885a1360422511624720e14422af0 at 10.0.0.7
    CSeq: 102 OPTIONS
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Date: Tue, 08 May 2007 11:34:55 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Length: 0


    ---

    <-- SIP read from 10.0.0.9:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK65c2f229;rport
    From: "asterisk" <sip:asterisk at 10.0.0.7>;tag=as0cc11f28
    To: <sip:101 at 10.0.0.9:5060>
    Call-ID: 5e0885a1360422511624720e14422af0 at 10.0.0.7
    CSeq: 102 OPTIONS
    Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
    Content-Length: 0


    --- (8 headers 0 lines) ---
    Destroying call '5e0885a1360422511624720e14422af0 at 10.0.0.7'

    I attached my configuration files.

    Thanks for all

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  Compuware Europe B.V. (Registration number: 33245192) is a company registered in The Netherlands whose registered office is at Hoogoorddreef 5, 1101 BA Amsterdam, The Netherlands. Compuware B.V. (Registration number: 33227492) is a company registered in the Netherlands whose registered office is at Hoogoorddreef 5, 1101 BA Amsterdam, The Netherlands 


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