[Asterisk-Users] SIP Hard Disconnect Detection
Title: SIP Hard Disconnect Detection Hello. I have a question regarding Asterisk internal API. I am developing a new asterisk module application using asterisk internal c API. I am having problem detecting hard hangups when the SIP clients disconnect (suppose power goes off in the phones). I am not receiving any disconnect control frames and don't know how to check if the clients are really connected. Can anyone help? Thank you, Pedro Goncalves
[Asterisk-Users] Record Application Problem
Title: Record Application Problem Hi everybody, I am having a problem with * Record Application. The thing is I don't want the beep before recording, so I removed the instructions: ast_streamfile(chan, null, chan-language); ast_waitstream(chan, ); ast_stopstream(chan); Now I am having a strange problem. After I record the sound, the recorded file gets a 3 second of silence before the actual recorded sound. Can anyone solve this?? I can workaround this by playing a silent sound file of about 0.25s before start recording... but I would prefer to get the problem properly solved. Thank you, Pedro Goncalves
[Asterisk-Users] H263 SIP Video Playback
Title: H263 SIP Video Playback Hi. Was anyone able to send an H263 to SIP clients through any Asterisk play function? If so, which h263 test files did you use? Thank you, Pedro Goncalves
[Asterisk-Users] indications.conf for Portugal
Title: indications.conf for Portugal Does someone have the settings for 'indications.conf' in Portugal? Thank you, Pedro Goncalves -- Pedro Goncalves PT Inovação SA - Pólo do Porto Largo de Mompilher, 22 - 4º 4050-392 Porto - Portugal Phone: +351 222079329 Email: [EMAIL PROTECTED] --
RE: [Asterisk-Users] Need a list of asterisk built-in variables
Title: RE: [Asterisk-Users] Need a list of asterisk built-in variables The ${CALLERIDNUM} variable has the calling extens number. Regards, Pedro Goncalves -Original Message- From: Justin Carlson [mailto:[EMAIL PROTECTED]] Sent: terça-feira, 6 de Abril de 2004 17:32 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Need a list of asterisk built-in variables I need to be able to use a variable that has the calling extension number rather than the called. thanks. ___ 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
RE: [Asterisk-Users] Need a list of asterisk built-in variables
Title: RE: [Asterisk-Users] Need a list of asterisk built-in variables Suppose EXT1 makes call to EXT2. Then the ${CALLERIDNUM} is the number of EXT1 while ${EXT} is the number of EXT2. Any doubts? Regards, Pedro Goncalves From: Justin Carlson [mailto:[EMAIL PROTECTED] Sent: terça-feira, 6 de Abril de 2004 18:01 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Need a list of asterisk built-in variables it puts the callerid number I have in the sip.conf instead. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Pedro Bessa Goncalves Sent: Tuesday, April 06, 2004 11:37 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Need a list of asterisk built-in variables The ${CALLERIDNUM} variable has the calling extens number. Regards, Pedro Goncalves -Original Message- From: Justin Carlson [mailto:[EMAIL PROTECTED]] Sent: terça-feira, 6 de Abril de 2004 17:32 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Need a list of asterisk built-in variables I need to be able to use a variable that has the calling extension number rather than the called. thanks. ___ 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
RE: [Asterisk-Users] Play file from an offset
Hi. I am also doing same thing as you. Currently my company has a PBX platform which uses Dialogic hardware to control channels. Now they want their software to work over Asterisk. I still havent started implementing modifications, as I am still studying Asterisk. I think it would be interesting and profitable if we could collaborate on this. How far have you done in your project? Did you test h263 video over SIP in Asterisk? I couldnt get it to work with various SIP video clients. Looking forward to hear from you. Best regards, Pedro Goncalves From: Yves Chouinard [mailto:[EMAIL PROTECTED] Sent: quarta-feira, 31 de Março de 2004 17:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Play file from an offset I am projecting to migrate applications written with the Dialogic API to Asterisk. There are a few things that I do with Dialogic that I am still not sure are possible with Asterisk : - play a file from an offset (so a user can press a key to rewind 3 sec., pause, etc.) - dynamic volume control (the user can press a key to increase or decrease the volume) Thanks if anyone has a clue... Yves