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[asterisk-users] keep incoming codec same as outcoming on sip proxy

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jnod99 at gmail.com
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Posted: Thu Apr 17, 2008 2:40 am Post subject: [asterisk-users] keep incoming codec same as outcoming on si

Hi, i have two computer with asterisk. One is a SIP proxy that Dial() the other. It is possible to be sure that the proxy does not make transcoding in any case and Hangup() the call if the Second asterisk does not support the codec ? Thanks

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