[asterisk-users] voicemail in mp3 format
Hi! Does anyone know about saving the voicemail messages in mp3 format? Thanks Ever Zalazar ___ --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] app-callforward
Hi People. Well I need help in app-callforward. I have a electronic recepcionist, the message she say is: Press 1 to enter your voice mail menuPress 2 to call forwardPress 3 to cancell call forward. If the user press 2, It will hear a message : Please enter the target attendant: The user dial a number(example 202020), then he hear a message again, call forward unconditional, blablablabla activated. But, when someone call to that customer, the call still going to his number, and not to the 202020(this number is supposed to ring).I saw in the database the CF/202021 : 202020But the asterisk is not doing that. Today the option 2, go to a extension 200, who dial to *72 to do the call forward.If I press directly *72, I'm not hear the message, directly put the number to be forwarded, in that case work. But I can't see it in the database. Someone please can give me some help? Best REgards Ever ___ --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] call forwarding
Hi people. I want to know about call forwarding. I dial *72, and a message say me to dial the extension , I did, then the message said is forward is UNCONDITIONLA . But when I call , it doesn't work the forwarding. Who can help me please. Best Regards Ever ___ --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] zapRAS
Hi people! Anyone have used zapRAS? It's possible to mount an asterisk with zapRAS and give internet service to some customers? Best Regards Ever ___ --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] frame.c:128 ast_smoother_feed
hello, anyone that know about this asterisk's message: frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Best REgards Ever ___ --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] frame.c:128 ast_smoother_feed
hello, anyone that know about this asterisk's message: frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Best REgards Ever ___ --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] frame.c:128 ast_smoother_feed
hello, anyone that know about this asterisk's message: frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Best REgards Ever ___ --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] multiple calls using IAX
Hi. I have a aplication for web, when u press on the link, the application log into an asterisk(user, password), and call to one extension(ex 201). How can I do to that call go to 201, if busy, go to 202, and so on? I want to implement in a call center. Best Regards Thanks Ever ___ --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] voicemailmain()
Hi, in the menu of voicemailmain, appear a lot of options, there is a way to leave only some of them? Also I want to know if there is a option that erase all message in a user box. Best REgards Ever Zalazar ___ --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] voicemailmain()
Hi, in the menu of voicemailmain, appear a lot of options, there is a way to leave only some of them? Also I want to know if there is a option that erase all message in a user box. Best REgards Ever Zalazar ___ --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] voice mail notification
Hello, there is a way to send notification(not email) when it's received an voice mail? Maybe a SIP message to inform? Best REgards Ever Zalazar ___ --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] vmail access problem
Hi everybody..I have the follow problem with my vmail access: http://voicemail.cheaphone.com/cgi-bin/vmail.cgi?action=""> For example this is the address to access the voice mail of one customer. If that customer change the number for : http://voicemail.cheaphone.com/cgi-bin/vmail.cgi?action=""> He will access that user account and see the messages.HOw can I protect this? Thanks Ever ___ --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 and S.E.R.
Hi, I have some questions : If a client connected to a S.E.R. use as codecs only G729, and I want to call and give him a message in gsm or wav format using the manager API from asterisk server? This will work directly or it's necesary a codec converter? My asterisk has the codec g729 as well. I try several times, but I didn't here the message in the client sip conected to S.E.R. Best Regards Ever Zalazar ___ --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] php+agi
Hello, I want to know if someone made a script in php(with agi) to call some voip number, and when the user answer the call, he hears a message with an advertisement. I want to input the number directly from cli or read the numbers from a file(ex.8021,8022,8023). Thanks in advantage Ever Zalazar ___ --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] new in asterisk world
Hi, I'm new in asterisk world. I have questions. For example I have my server with public IP address, but two customer with softphone in a private network. How can I do to make them work with the asterisk server? Best Regards -- Ever Zalazar ___ --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
