Just getting started with Asterisk. I have home office PABX using ATCOM IP04 running uclinux O/S and Asterisk 1.4 as provided by ATCOM. In addition to some analog phone lines I also have a SIP trunk set up to my VoIP Provider VIATALK. I have configured Asterisk using dynamic features to recognize a DTMF key sequence from my phones to generate a hookflash on the analog phone lines using zapflash however I don’t know how to configure Asterisk to send a hook flash signal to VIATALK so I can access their call waiting and 2 party calling features. Normally VIATALK provides a LINKSYS PAP2 MTA and the LINKSYS MTA responds to a hook flash from an attached analog telephone set to send some sort of signal to VIATALK but I don’t know what that signal is or how to cause Asterisk to send the same signal. I have some limited skills with LINUX and Asterisk so providing me or pointing me to an answer suitable for a newbee would be most appreciated. My online research on the LINKSYS PAP2 seems to indicate that it signals a hook flash by sending a SIP INFO message with a MIME content-type:application/hook-flash however I see other information online that describes sending the DTMF value “16” using SIP INFO method of sending DTMF. I am not sure if either of these are correct and if correct how to cause Asterisk to generate these signals. I have looked at the sendDTMF command and see that is does not support the value “16” so that does not see to be productive. I also don’t see a native method in Asterisk 1.4 to send a SIP INFO message with the correct MIME type. I have taken notice of the addsipheader command but that seems to only work when the dial command is executed. What I need to do is send the hookflash to VIATALK after the initial dial command and then send a new phone number for the 2nd party in the conference and then a final hookflash to bridge the calls together.
Any help would be appreciated.