I am unable to figure out why I am not able to blind transfer when I am the caller and I call the extension defined by gosub. When running asterisk -rvvv I can see: — Channel PJSIP/-00000009: Dialed @gosub-stdexten does not exist. It is evident th..

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how can I separate individual calls in my asterisk log? I noticed that a call always starts with following line: VERBOSE[C-0000000c]: Using SIP RTP CoS mark 5 How reliable is this as a separator? It seems it appears twice during every call. And, w..

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