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Hi, I am using asterisk 1.8.22 and have a problem when calling in parallel several SIP endpoints and I am not sure how to resolve it. In this case Asterisk will not bridge any audio to the caller before the 200 OK. Which means any progress announcements, including remotely generated ringback, are not passed back to the caller. This behavior is completely correct, because there is no way to know

Hi, I am using Asterisk 1.8 and trying to pack some custom data in a CEL HANGUP event. In a master (inbound) channel I can set the CHANENL(userfield) to pass custom information to a CEL event. In the outbound channel created by Dial() I can also possibly use a macro/gosub on answer and set the CHANENL(userfield) from there. The problem is how to set it in an outbound channel created by Queue()