[asterisk-users] Using asterisk as voicemail for cisco call manager
Hello; Instead of using Cisco Unity as voicemail for Cisco Call Manager, I need to use asterisk to be the voicemail for the Cisco Call Manager version 7 which supports SIP. Did anyone try this? Was it a successful implementation? If yes, I hope that someone gives a steps to help me. Regards Bilal-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outgoing call queues
Hello, Is there any way to make Asterisk not drop calls that are overwhelming to the outbound trunk capacity and have it queue that call till the time the trunk is free to take a new call? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change time between DTMF
You could use Timeout function. Like this: 1)Timeout(digit): max interval time between digits in DTMF. 2)Timeout(response): max time wait for autofallthrough. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)
My scenario is (2) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Realtime
Hello everyone, I'm wondering if someone could help me. I would like to configure Voicemail users in realtime, by to not realtime voicemail storages. So i would like to have all voicemail accounts in database, but all voice message to be stored on disk. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)
On Wednesday 04 Jun 2014, Mojtaba wrote: > Thank you for your replying. > Is there any way so that i could found the far end user pick up phone? I > could use Wait() function in dialplan but i dont how long (secend) > should be wait! > Thanks with Regards.Mojtaba I'm confused now. Please describe which scenario applies to you: (1) Your Asterisk box shares an analogue line with an ordinary telephone. You want the person sitting at that telephone to be able to answer it and prevent the Asterisk box friom answering it. This is where you need a Wait() in your dialplan. (2) Outgoing calls from your Asterisk box are showing up as "answered" even whether or not the person at the far end picked up their telephone. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
