voice-class sip bind and bind under sip

[Rafael Jimenez]

Hi,

I have adoubt about the bind interface.

Im having problems setting a SIP TRUNK,

Im guessing everything because the SIP SP is not much helpfull.

My question is because when I put

sip

  bind all source-interface Loopback0  /* lo 0 is the same source-address in voice register global and telephony-service.

The ip phones register fine.

If I change the bind all source-interface to gi0/0  (to the SIP SP) the ip phones remains "registering".

Im not sure what if I setting the bind to the lo0 in the sip section and bind ....gi0/0 in the dial-peer going to the SIP SP will help me to solve the problem.

I need some ideas.

Thanks.

[Chris Deren · Accepted solution]

Leave the global binding to loop back interface, change the binding to provider interface on the dial peers pointing to/from provider.

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