I've currently configured incoming calls to simultaneously ring an analog phone (via TDM400P) and two SIP phones. I'd like to have it also simultaneously dial out the TDM400P on a PSTN to ring my cell phone, and have the first one to answer win the battle.
In my digging I've figured out that I can add the Zap channel to the dial list, such as Dial(SIP/7001&SIP/7002&ZAP/3/5551212,20), however when I include the PSTN line in this command (ZAP/3/....) I get an interesting thing happening.
All SIP phones start ringing.
Asterisk connects ZAP/3 to dial out and dials out
Asterisk then says to the effect of "ZAP/3 has answered the call" (since the line has now gone off hook) and stops ringing all the SIP phones immediately, leaving me with only the cell phone ringing. It then fails to go to Voicemail and just keeps ringing the cell phone, because as far as Asterisk is concerned the line has been bridged and is connected.
Any suggestions?
Analog FXO ports ae considered "answered" as soon as the dialing is finished. Nothing you can do about this because there is no way for Asterisk to know when the far end answers. This is not a problem with (most) Channelized Voice T-1, it's not a problem with PRI and not a problem with VoIP telephone companies, since they all use PRI.
You can sort of work around this problem by using the poorly documented "c" option to the Zap dial command. Something like Zap/1c or something like that. I've never used it. That option requires the callee press # to accept the call. No sound file is played. See the mailing list archives. It's been discussed off and on.
--Eric --Eric _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
