Estimados, estoy configurando una trunk de Skype en asterisk, el problema que tenemos es que se queda en Request Sent, no se si esta algo mal puesto o que. les dejo los datos de los archivos y lo que me arroja el CLI
parte del archivo sip.conf
848 register => xxxxxxx:xxxxxxxxxx@sip.skype.com:5060/xxxxxxxxxxx
849 ;
850 ;
851 ;register => 2345assword@sip_proxy/1234
852 ;
853 ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
854 ; connect to local extension 1234 in extensions.conf, default context,
855 ; unless you configure a [sip_proxy] section below, and configure a
856 ; context.
857 ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
858 ; Tip 2: Use separate inbound and outbound sections for SIP providers
859 ; (instead of type=friend) if you have calls in both directions
860 ;
861 ;register => 3456@mydomain:5082::@mysipprovider.com
862 ;
863 ; Note that in this example, the optional authuser and secret portions have
864 ; been left blank because we have specified a port in the user section
865 ;
866 ;register => tls://username:xxxxxx@sip-tls-proxy.example.org
867 ;
1633 ;[skype]
1634 ;host=sip.skype.com
1635 ;username=xxxxxxxxxxxxxxxx
1636 ;fromuserxxxxxxxxxxxxx
1637 ;secret=xxxxxxxxxxx
1638 ;type=friend host= sip.skype.com
1639 ;fromdomain=sip.skype.com
1640 ;insecure=port,invite
CLI:
[Jan 6 14:04:21] NOTICE[2039]: chan_sip.c:15175 sip_reregister: -- Re-registration for xxxxxxxxxxxxxxx@sip.skype.com
[Jan 6 14:04:21] NOTICE[2039]: chan_sip.c:15290 transmit_register: Strange, trying to register xxxxxxxxxxxxxxxxx8@sip.skype.com when registration already pending
Retransmitting #3 (NAT) to 204.9.161.164:5060:
REGISTER sip:sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 192.168.6.173:5060;branch=z9hG4bK2c3b43a5;rport
Max-Forwards: 70
From: <sip:xxxxxxxxxxxxxxx@sip.skype.com>;tag=as669a3a 1 2
To: <sip:xxxxxxxxxx@sip.skype.com>
Call-ID: 14fd0b05725c5f3954e5413a3ffdfb5c@[::1]
CSeq: 189 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:xxxxxxxxxxxx@192.168.6.173:5060>
Content-Length: 0
no se que puede ser, hemos configurado varias veces y no funciona, en google muchas guias y nada desde ya agradezco su ayuda
parte del archivo sip.conf
848 register => xxxxxxx:xxxxxxxxxx@sip.skype.com:5060/xxxxxxxxxxx
849 ;
850 ;
851 ;register => 2345assword@sip_proxy/1234
852 ;
853 ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
854 ; connect to local extension 1234 in extensions.conf, default context,
855 ; unless you configure a [sip_proxy] section below, and configure a
856 ; context.
857 ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
858 ; Tip 2: Use separate inbound and outbound sections for SIP providers
859 ; (instead of type=friend) if you have calls in both directions
860 ;
861 ;register => 3456@mydomain:5082::@mysipprovider.com
862 ;
863 ; Note that in this example, the optional authuser and secret portions have
864 ; been left blank because we have specified a port in the user section
865 ;
866 ;register => tls://username:xxxxxx@sip-tls-proxy.example.org
867 ;
1633 ;[skype]
1634 ;host=sip.skype.com
1635 ;username=xxxxxxxxxxxxxxxx
1636 ;fromuserxxxxxxxxxxxxx
1637 ;secret=xxxxxxxxxxx
1638 ;type=friend host= sip.skype.com
1639 ;fromdomain=sip.skype.com
1640 ;insecure=port,invite
CLI:
[Jan 6 14:04:21] NOTICE[2039]: chan_sip.c:15175 sip_reregister: -- Re-registration for xxxxxxxxxxxxxxx@sip.skype.com
[Jan 6 14:04:21] NOTICE[2039]: chan_sip.c:15290 transmit_register: Strange, trying to register xxxxxxxxxxxxxxxxx8@sip.skype.com when registration already pending
Retransmitting #3 (NAT) to 204.9.161.164:5060:
REGISTER sip:sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 192.168.6.173:5060;branch=z9hG4bK2c3b43a5;rport
Max-Forwards: 70
From: <sip:xxxxxxxxxxxxxxx@sip.skype.com>;tag=as669a3a 1 2
To: <sip:xxxxxxxxxx@sip.skype.com>
Call-ID: 14fd0b05725c5f3954e5413a3ffdfb5c@[::1]
CSeq: 189 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:xxxxxxxxxxxx@192.168.6.173:5060>
Content-Length: 0
no se que puede ser, hemos configurado varias veces y no funciona, en google muchas guias y nada desde ya agradezco su ayuda