Right now I have an Asterisk 1.4.18ish server and a Wildcard POTS interface. As it is now, when the zap line gets a call, Asterisk answers it and waits for the analog CID to be presented, then rings the SIP phones with the call and the CID. There's a significant latency involved in doing this.
I'm wondering if the SIP lines can start ringing as soon as the zap line gets a call and when the zap line finally gets the CID, that is passed down to the already ringing SIP phones. That way if a SIP phone user wants to wait for the CID, they can, but if they just want to answer the phone without waiting for the CID, they can do that too. One might suggest that everyone wants to see the CID anyway, so why bother? Because in some situations, the phone is not at an arms reach and the person only starts making their way towards it when they start to hear the ringing, so if the ringing starts before the CID is available it is likely that by the time they have gotten to the phone, the CID is available and yet the latency between the availability of the call on the zap line and it being picked up at a ringing phone has been reduced a ring or two. b.
signature.asc
Description: This is a digitally signed message part
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users