Re: [asterisk-users] some questions about atxfer usage
Hi I just press * to retrieve the caller again - Have you tried that? No, I haven't. Thanks, it's perfect for me. Conrad ___ --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
Re: [asterisk-users] some questions about atxfer usage
OK Thank you very much. On 11/16/06, Alberto Pastore [EMAIL PROTECTED] wrote: Antonio Almodóvar ha scritto: Hi all. I have enabled the attended transfer feature in features.conf. I'm using it and I want to resolve some questions, I hope someone can help me :) When I transfer a call to an extension: - The extension rings during 15 seconds and the call returns to the transferer. Is there any possibility to recover the call before the timeout of 15 seconds expires? I mean, I would like to personalize the way of making transfers using the feature of atxfer. How can I do that? Thanks in advance. ___ --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 Hi Antonio. Taking a look at the following code line from res_features.c: newchan = ast_feature_request_and_dial(transferer, Local, ast_best_codec(transferer-nativeformats), dialstr, 15000, // --- outstate, cid_num, cid_name); I assume that 15000 msecs is a hardcoded value... You might want to replace it with some variable taken from pbx_builtin_getvar_helper() results but it involves recompiling at least the res_features.c module; something more or less like this (I haven't tested it!!!): //these two lines go at the beginning of the if {} block char *transfer_timeout_str; int transfer_timeout = 15; //default value //these lines replace the newchan = ast_feature_request_and_dial(...) one //read the value (if any) from TRANSFER_TIMEOUT //can be set in extensions.conf's [globals] (TRANSFER_TIMEOUT = 30) transfer_timeout_str = pbx_builtin_getvar_helper(transferer, TRANSFER_TIMEOUT); if (transfer_timeout_str) { transfer_timeout = atoi(transfer_timeout_str); //sanity check if (transfer_timeout = 0) transfer_timeout = 15; } newchan = ast_feature_request_and_dial(transferer, Local, ast_best_codec(transferer-nativeformats), dialstr, transfer_timeout * 1000, // --- outstate, cid_num, cid_name); Bye, Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] some questions about atxfer usage
Hi all. I have enabled the attended transfer feature in features.conf. I'm using it and I want to resolve some questions, I hope someone can help me :) When I transfer a call to an extension: - The extension rings during 15 seconds and the call returns to the transferer. Is there any possibility to recover the call before the timeout of 15 seconds expires? I mean, I would like to personalize the way of making transfers using the feature of atxfer. How can I do that? Thanks in advance. ___ --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] some questions about atxfer usage
On Wed, 2006-11-15 at 13:54 +0100, Antonio Almodóvar wrote: Hi all. I have enabled the attended transfer feature in features.conf. I'm using it and I want to resolve some questions, I hope someone can help me :) When I transfer a call to an extension: - The extension rings during 15 seconds and the call returns to the transferer. Is there any possibility to recover the call before the timeout of 15 seconds expires? I just press * to retrieve the caller again - Have you tried that? I mean, I would like to personalize the way of making transfers using the feature of atxfer. How can I do that? anything in particular? Conrad ___ --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] some questions about atxfer usage
Antonio Almodóvar ha scritto: Hi all. I have enabled the attended transfer feature in features.conf. I'm using it and I want to resolve some questions, I hope someone can help me :) When I transfer a call to an extension: - The extension rings during 15 seconds and the call returns to the transferer. Is there any possibility to recover the call before the timeout of 15 seconds expires? I mean, I would like to personalize the way of making transfers using the feature of atxfer. How can I do that? Thanks in advance. ___ --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 Hi Antonio. Taking a look at the following code line from res_features.c: newchan = ast_feature_request_and_dial(transferer, Local, ast_best_codec(transferer-nativeformats), dialstr, 15000, // --- outstate, cid_num, cid_name); I assume that 15000 msecs is a hardcoded value... You might want to replace it with some variable taken from pbx_builtin_getvar_helper() results but it involves recompiling at least the res_features.c module; something more or less like this (I haven't tested it!!!): //these two lines go at the beginning of the if {} block char *transfer_timeout_str; int transfer_timeout = 15; //default value //these lines replace the newchan = ast_feature_request_and_dial(...) one //read the value (if any) from TRANSFER_TIMEOUT //can be set in extensions.conf's [globals] (TRANSFER_TIMEOUT = 30) transfer_timeout_str = pbx_builtin_getvar_helper(transferer, TRANSFER_TIMEOUT); if (transfer_timeout_str) { transfer_timeout = atoi(transfer_timeout_str); //sanity check if (transfer_timeout = 0) transfer_timeout = 15; } newchan = ast_feature_request_and_dial(transferer, Local, ast_best_codec(transferer-nativeformats), dialstr, transfer_timeout * 1000, // --- outstate, cid_num, cid_name); Bye, Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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