❤Asterisk meetme timing source ❤ Click here: http://hosetmalown.fastdownloadcloud.ru/dt?s=YToyOntzOjc6InJlZmVyZXIiO3M6MjE6Imh0dHA6Ly9iaXRiaW4uaXQyX2R0LyI7czozOiJrZXkiO3M6Mjk6IkFzdGVyaXNrIG1lZXRtZSB0aW1pbmcgc291cmNlIjt9 Perhaps you should enable some sort of flag in the Page application to control which application is used for mixing the channels? But the last one is destroying it! Zorro azzoumarou at hotmail dot com 30 August 2006 13:43:31 I have no audio Carte, and i want to configure conférence with Asterisk ,but i can't. Starting with Asterisk 1. I've dug around the mailing lists and through the bug reports, and it appears that the problem is known at least, and hopefully will be fixed in a future release, or when MeetMe2 is finally ready for prime time. Certified Asterisk 13 has an end-of-life date EOL of October 24, 2019, and Ubuntu 14. Maybe there is something I have to do to instantiate the conference room that I'm skipping over. The general idea is that audio has to be played at exactly the fub time so that it sounds correct and isn't choppy or distorted. Or are there other tools for having conference in asterisk with actual stuff. The best option would be to check for the page application and if it is available go ahead and use that option. Now i con to make conference call acording to call center process. I've been having some latency issues with MeetMe and either SIP or IAX clients and an IAX trunk to my PSTN provider. I have a SIP truck connected from our Metaswitch and can make any inbound or solo calls as needed. Even though internal timing is not a requirement for most Asterisk functionality, it may be advantageous to use it asterisk meetme timing source the alternative is to use timing based on incoming frames of audio. DTMF digits used to enable conference features will not be passed through. I met my extensions. Asterisk Forums - If the conference number is omitted, the user will be prompted to enter one. I have the feeling it is something obvious. Most of the features now are independent of dahdi for a timing source now that there is kernel timing available in 1. It would be better to remove that dependency. I've attached a diff which changes the application to confbridge rather than meetme. If you're removing functionality from Page because ConfBridge doesn't have recording, etc... Perhaps you should enable some sort of flag in the Page application to control which application is used for mixing the channels? I would not support removing the ability to use Page as it currently stands. Perhaps you should enable some sort of flag in the Page application to control which application is used for mixing the channels? I would not support removing the ability to use Page as it currently stands. That is a good idea. The best option would be to check for the page application and if it is available go ahead and use that option. If not, then write a warning the log that confbridge is being used instead and recording isn't available. Then use confbridge to mix the channels. I've been testing confbridge a little more and am having a few issues getting things to work. I'll see what I can do but I don't have any license on file to submit a patch. FYI, this is the same approach that patches I've supplied to freepbx do as well, check for an application that may require dahdi and if it is available use it e. Does that make sense? The best option would be to check for the page application and if it is available go ahead and use that option. If not, then write a warning the log that confbridge is being used instead and recording isn't available. Then use confbridge to mix the channels. I've been testing confbridge a little more and am having a few issues getting things to work. I'll see what I can do but I don't have any license on file to submit a patch. FYI, this is the same approach that patches I've supplied to freepbx do as well, check for an application that may require dahdi and if it is available use it e. Does that make sense?