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Hello

I have noticed something annoying. When calls are routed through A2B, they are immediately picked up - ie, the calling party gets a SIP 200 / OK immediately.

details
- one sip client, attached to a card by caller id
- one outgoing sip trunk

Client dials, calls to asterisk, extensions passes to deadagi/a2b, trunk configured as sip trunk.

Through ngrep, it appears that a message SIP 200 /OK is sent to the client before the INVITE goes out on the oubound trunk.

when I alter my extensions.conf not to use A2B, I get a SIP 183 Session in progress, until the remote trunk is connected.

I've tried to remove from extensions.conf the initial Answer indicated by the install instructions, no luck

I could find this topic on the forum either...


Any suggestions ?

Many thanks,

J.