Tag : RTP

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Ive upgraded my asterisk 1.4 to the version 1.8.11. After making some adjustments to the configuration files to port it to the new version, calls between registered phones in asterisk, work fine, but inbound calls coming from the SIP trunk I have w..

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We use a obfuscation software to encrypt/mangle both SIP/RTP which sits before asterisk. What happens is sometimes we dont get any voice. after some rtp set debug we found out that when received ip of the rtp stream is routers public ip, everything wo..

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