[Asterisk-Users] Send HookFlash after answering a ZAP (analog) channel
My hardware configuration looks like this: public network ---other PBX ---analog line (ext 100) --- FXO,Asterisk | |--- extension (200) by calling extension 100 from public network I can call Asterisk. Now I would like to answer the call, SayNumber and the transer back to the extension 200 in the other pbx. When I put a telephone instead of Asterisk on the line 100 I can do the transfer by pressing Hookflash and the dial 200 and then hookon. The call is no at 200. extension.conf --- exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,SayNumber(101) exten = s,4,Wait(2) exten = s,5,Transfer(**51) exten = s,6,Wait(40) -- I trie something, but it is not working. kinds regards. HJB ___ --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
Re: [Asterisk-Users] Send HookFlash after answering a ZAP (analog) channel
Found solution: exten = s,1,Wait(10) exten = s,2,Answer() exten = s,3,Wait(5) exten = s,4,Flash() exten = s,5,Wait(2) exten = s,6,SendDTMF(200) exten = s,7,Wait(5) exten = s,8,Hangup() --- Ursprüngliche Nachricht --- Von: Hans-Juergen Brand [EMAIL PROTECTED] An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] Send HookFlash after answering a ZAP (analog) channel Datum: Mon, 13 Feb 2006 19:16:23 +0100 (MET) My hardware configuration looks like this: public network ---other PBX ---analog line (ext 100) --- FXO,Asterisk | |--- extension (200) by calling extension 100 from public network I can call Asterisk. Now I would like to answer the call, SayNumber and the transer back to the extension 200 in the other pbx. When I put a telephone instead of Asterisk on the line 100 I can do the transfer by pressing Hookflash and the dial 200 and then hookon. The call is no at 200. extension.conf --- exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,SayNumber(101) exten = s,4,Wait(2) exten = s,5,Transfer(**51) exten = s,6,Wait(40) -- I trie something, but it is not working. kinds regards. HJB ___ --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
[Asterisk-Users] Asterisk on VMWare 4.5, Error Ouch ... error while writing audio data
I' m using Asterisk 1.09 on an virtual pc (VMWare 4.5) for testing. I can make calls from a Softphone to softphone, Hardphone to Softphone and so on. I can hear both RTP Streams. But when I call prompst on Asterisk I can hear nothing. RTP Stream goning from Phone to Asterisk but not the other way. I I start the PBX for console I got an error [EMAIL PROTECTED] root]# Ouch ... error while writing audio data: : Broken pipe ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
