We have Asterisk 13.22.0 (FreePBX 14) as a PBX, Cisco2911 with FXO modules as a gateway to the city telephone network. Outgoing routing is configured, calls go.

The task is to play the call recording messages to the called subscribers. Implemented as adding option A (custom / DialogIsRecorded) to a trunk.

The problem is that the playback of the recording begins after the Asterisk is connected to the ciscus, and not to the subscriber. As a result, the subscriber, at best, hears a part of the phrase being played, at worst (if he picked up the phone at all at once) - nothing at all.

It seems to me that the situation looks like this: for an outgoing call, the Asterisk is connected via SIP to cis, the tsiska responds with SIP 200 OK status, and the Asterisk considers that the connection is established and starts playing the message. But Tsisk at this moment is just starting to dial the subscriber number, so the message goes into the void.

How can I get Asterisk to play a message after picking up the handset by the called subscriber, and not after connecting to the gateway?

    1 answer 1

    There are two options - correct and cheap / non-working.

    The correct option is to get rid of FXO in principle. Digital PBX and analog lines are a bad combination.

    The second option is to try using TALK_DETECT, for example, like here .