app_dial.c no sale llamada con provvedor sip

Discussion in 'Elastix 2.x' started by nancy, Sep 27, 2009.

  1. nancy

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    Hola;
    Espero me puedan ayudar configure mi proveedor sip y si se registro, pero no me salen las llamadas me marca lo siguiente en el CLI

    Reloading SIP
    == Using SIP RTP CoS mark 5
    == Using SIP VRTP CoS mark 6
    -- Executing [8524929982034@internas:1] Dial("SIP/400-08a683e0", "SIP/ProveedorSip/524929982034,30,Tt" ) in new stack
    == Using SIP RTP CoS mark 5
    == Using SIP VRTP CoS mark 6
    [Sep 26 19:01:44] WARNING[5086]: app_dial.c:1528 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
    == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [8524929982034@internas:2] Hangup("SIP/400-08a683e0", "" ) in new stack
    == Spawn extension (internas, 8524929982034, 2) exited non-zero on 'SIP/400-08a683e0'
    [Sep 26 19:01:57] NOTICE[3982]: dnsmgr.c:172 dnsmgr_refresh: dnssrv: host 'sip.12voip.com' changed from 194.120.0.198:0 to 77.72.169.129:0
    [Sep 26 19:01:57] NOTICE[3982]: chan_sip.c:10169 sip_reg_timeout: -- Registration for 'geasa@sip.12voip.com' timed out, trying again (Attempt #1)
    -- Refreshing DNS lookups.
    [Sep 26 19:03:08] NOTICE[3673]: dnsmgr.c:172 dnsmgr_refresh: dnssrv: host 'sip.12voip.com' changed from 77.72.169.129:0 to 194.120.0.198:0
    [Sep 26 19:03:08] NOTICE[3673]: dnsmgr.c:172 dnsmgr_refresh: dnssrv: host 'sip.12voip.com' changed from 0.0.0.0:5060 to 77.72.169.129:5060
    -- Refreshing DNS lookups.


    == Using SIP RTP CoS mark 5
    == Using SIP VRTP CoS mark 6
    -- Executing [8524929982034@internas:1] Dial("SIP/400-08a62cb0", "SIP/ProveedorSip/524929982034,30,Tt" ) in new stack
    == Using SIP RTP CoS mark 5
    == Using SIP VRTP CoS mark 6
    [Sep 26 19:14:01] WARNING[6778]: chan_sip.c:2921 __sip_xmit: sip_xmit of 0xb6c98040 (len 924) to 77.72.169.129:5060 returned -1: Address family not supported by protocol
    -- Called ProveedorSip/524929982034
    [Sep 26 19:14:02] WARNING[3982]: chan_sip.c:2921 __sip_xmit: sip_xmit of 0xb6c98040 (len 924) to 77.72.169.129:5060 returned -1: Address family not supported by protocol
    [Sep 26 19:14:03] WARNING[3982]: chan_sip.c:2921 __sip_xmit: sip_xmit of 0xb6c98040 (len 924) to 77.72.169.129:5060 returned -1: Address family not supported by protocol
    [Sep 26 19:14:05] WARNING[3982]: chan_sip.c:2921 __sip_xmit: sip_xmit of 0xb6c98040 (len 924) to 77.72.169.129:5060 returned -1: Address family not supported by protocol
    [Sep 26 19:14:09] WARNING[3982]: chan_sip.c:2921 __sip_xmit: sip_xmit of 0xb6c98040 (len 924) to 77.72.169.129:5060 returned -1: Address family not supported by protocol
    [Sep 26 19:14:17] WARNING[3982]: chan_sip.c:2921 __sip_xmit: sip_xmit of 0xb6c98040 (len 924) to 77.72.169.129:5060 returned -1: Address family not supported by protocol
    -- Nobody picked up in 30000 ms
    -- Executing [8524929982034@internas:2] Hangup("SIP/400-08a62cb0", "" ) in new stack
    == Spawn extension (internas, 8524929982034, 2) exited non-zero on 'SIP/400-08a62cb0'


    Les agradeceria mucho su ayuda version de asterisk 1.6.1.6

    De antemano Gracias!
     

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