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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Josu Lazkano 
Lete
    Sent: Tuesday, May 08, 2007 1:36 PM
    To: asterisk-users@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:[EMAIL PROTECTED] SIP/2.0
    Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport
    From: 101 <sip:[EMAIL PROTECTED]>;tag=3159122210
    To: "943833473" <sip:[EMAIL PROTECTED]>
    Call-ID: [EMAIL PROTECTED]
    CSeq: 1 INVITE
    Contact: <sip:[EMAIL PROTECTED]: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 - [EMAIL PROTECTED]
    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:[EMAIL PROTECTED]>;tag=3159122210
    To: "943833473" <sip:[EMAIL PROTECTED]>;tag=as705b7b72
    Call-ID: [EMAIL PROTECTED]
    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 '[EMAIL PROTECTED]' in 15000 ms
    Found user '101'

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


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

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


    --- (9 headers 0 lines) ---
    Destroying call '[EMAIL PROTECTED]'
    Destroying call '[EMAIL PROTECTED]'
    12 headers, 0 lines
    Reliably Transmitting (no NAT) to 10.0.0.9:5060:
    OPTIONS sip:[EMAIL PROTECTED]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK65c2f229;rport
    From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as0cc11f28
    To: <sip:[EMAIL PROTECTED]:5060>
    Contact: <sip:[EMAIL PROTECTED]>
    Call-ID: [EMAIL PROTECTED]
    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:[EMAIL PROTECTED]>;tag=as0cc11f28
    To: <sip:[EMAIL PROTECTED]:5060>
    Call-ID: [EMAIL PROTECTED]
    CSeq: 102 OPTIONS
    Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, 
UPDATE, MESSAGE
    Content-Length: 0


    --- (8 headers 0 lines) ---
    Destroying call '[EMAIL PROTECTED]'

    I attached my configuration files.

    Thanks for all

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