Tried that, and no go. There's something wrong with Zapteller. It works fine on internal calls, but the only way I can get it to work on external calls (through a SIP/PSTN gateway, no Zap hw necessary) is to first play a message. For instance, this works:
exten => 2200,1,Playback(ss-noservice) exten => 2200,2,Zapateller exten => 2200,3,Dial(SIP/2205) -brian > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Andrew Thompson > Sent: Friday, April 09, 2004 12:48 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Problems with Zpateller on > incoming external calls > > Brian Cuthie wrote: > > I've setup the following in extensions.con: > > exten => 2200,1,Ringing > > exten => 2200,2,Wait(2) > > exten => 2200,3,Answer > > exten => 2200,4,Zapateller > > exten => 2200,5,Macro(stdexten,2205,SIP/2205) > > This works as expected if I dial from a SIP phone on my desk. > > However, if I dial in from the PSTN (through a SIP > provider) it fails > > while trying to play ths SIT with: Apr 8 18:53:12 > > WARNING[1209269552]: rtp.c:407 ast_rtp_read: RTP Read > error: Resource > > temporarily unavailable > > Any idea what's going on? My suspicion is that the PSTN gateway > > hasn't setup an audio path yet, although I thought Answer would do > > that. > > Cheers, > > Brian > > I don't have a zap device to test on, but can you do Ringing > before you Answer? > > ----- > Andrew Thompson > http://aktzero.com/ > > > _______________________________________________ > 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 > _______________________________________________ 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