I can think of a couple of ways to achieve testing of a PSTN line but this would seem to be the easiest.
Attempt to call an incoming PSTN/SIP/IAX line from your outgoing PSTN trunk, answer the call at a vmail box and notify you of a message via email. insert a delay of x minutes and do it again. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada > yes.. actualy use 1 did for each proxy to check.. > > then inbound for each use the method he described.. > > > On 10/12/06, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> > wrote: >> >> on an analog Zap PSTN channel, you have no real way of determining if >> the remote side answered, because, as you discerned, it IS considered >> answered as soon as asterisk opens the channel. >> >> How about you contact another asterisk server through the PSTN, and dial >> through to an extension on that remote asterisk server that, in turn, >> notifies the first asterisk server maybe via the internet that it was >> received? >> >> for example, consider the following php script accessupdate.php on >> primary asterisk box: >> >> <?php >> if (!strcmp($_GET['update'], 'true')) >> { >> touch("/etc/asterisk/secondary_server_last_access"); >> } >> ?> >> >> then primary calls secondary box through PSTN, and through the magic of >> DISA or CID or what-have-you, dials through to an extension that >> executes >> System(wget -q -O /dev/null >> http://primary-server/access_update.php?update=true) >> >> then hangs up. then primary server checks the last-access time of >> /etc/asterisk/secondary_server_last_access to make its decision, via >> cron script or bash script triggered through the dialplan subsequent to >> the initial dial-out. >> >> This is of course a very rudimentary on-the-fly thing I came up with, >> but think outside the box and this may be the easiest way for you to do >> what you want. >> >> Moj >> >> >> John Kane wrote: >> > I am trying to write a script to attempt to make a call on a Zap >> > channel, and if it fails, send an alarm. I can generate the call, but >> > because the Zap channel accepts the call, even though the other end >> > never answers, it sees it as a successful call, which it isn't. >> > >> > >> > >> > Anyone have any ideas on this? Thanks. >> > >> > !DSPAM:500,452d7fa8199221504517840! >> > >> > >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > --Bandwidth and Colocation provided by Easynews.com -- >> > >> > asterisk-users mailing list >> > To UNSUBSCRIBE or update options visit: >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> > >> > !DSPAM:500,452d7fa8199221504517840! >> >> -- >> Mojo <[EMAIL PROTECTED]> >> Office Manager, Horan & Company, LLC >> (907) 747-6666 x112 >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > Mike > Sales Manager > http://www.theclubvoip.com > Making it happen > 1.877.807.VOIP (8647) > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users