search for: rollovers

Hello, I've got 2 analogue lines (from SBC) coming into a TDM22B. SBC have put rollover from the first to the second line. The rollover works fine when handsets are connected directly to the lines (ie when Asterisk is not involved), but when the lines are connected to Asterisk, the rollover fails: the caller just hears the line ringing, and the person on the first (busy) line hears call

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8...

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8...

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8...

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8...

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, I have a simple question about incoming phone line rollovers. How are these usually done? Is this done at the phone company usually, or is this something that Asterisk or channel bank is capable of? I just need someone to give me a brief explanation how it usually works, and if someone was implementing an Asterisk system, how they would go about providing...

I am trying to perform a "rollover" when the primary number is busy. This is coming from a POTS line. Apparently I need call waiting on the POTS line as I get immediate busy from the FXS if I don't have it. So my question is this. I have an Aastra 480I CT. The call forward when busy here seems pretty straight forward. Choose the mode as busy enter the extension in the forward number

https://bugzilla.samba.org/show_bug.cgi?id=1529 Summary: 32bit rollover problem rsyncing files greater than 4GB in size Product: rsync Version: 2.6.2 Platform: x86 OS/Version: Linux Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned@samba.org

...ing like the big boys do and dial 9 first? I'm guessing a custom dial plan might do that but I haven't figured out how to do it. I'm running asterisk@home version 2.8 sounds right. (maybe that's asterisk 2.8) Can someone shed some light on a work around until I can figure out rollovers? Thanks, Jim.

Have a question about implementing Call Rollover with my current extensions.conf configuration. [macro-stdexten] exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Voicemail2(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce exten => s,3,Goto(default,s,1) ; If they press #, return to start exten =>

My goal is to have only 1 primary phone number that can seamlessly "pool" multiple VoicePulse accounts. Let's say I have 3 accounts with VoicePulse Connect 212-555-1000 (primary) 212-555-1001 212-555-1002 When I receive inbound calls on 212-555-1000, I want to "forward" or "roll over" the connection to 212-555-1001 and 212-555-1002 so that the 212-555-1000

I have about 5 incoming USA SIP lines, but my provider does not have any sort of roll-over or huntgroup feature. Does anybody have an idea on how I can create a general number that will ring to the next available, non-busy SIP line that I have? Is there a provider out there that would do this? Any suggestions would be greatly welcome. Thank you. -------------- next part --------------

It appears that part of the problem could be that when logger rolls over the mongrel log file. There is a mutex synchronization issue where the various mongrel instances all try to write to the new log file and block each other. This keeps them in a continual Application 500 Error state. Has anyone run into this problem? With log rotation turned off the problem does not appear. - Jared Brown

I want to have a link_to (<a>) that when the mouse rolls over it (hover) a little if possible semitranspartent "box" (maybe 80px*80px) is displayed with some help/further information included. Is it possible to do this? -- Posted via http://www.ruby-forum.com/.

I have the Polycom 500 and 600 phones. Rather than put an entry in for each line appearance, I would like to use the feature that shares one extension for the lines, so that I will get the call on the enxt available button. How do I configure that? Chris Mason

Hello, I've got an Asterisk system with 3 SIP trunks configured. Each SIP trunk is actually a 4 port Mediatrix PSTN gateway. The current outbound call routing (via AMP 1.10.007a) uses the 3 trunks in descending order, all set with max channels to 4. Unfortunately, when the first trunk reports a "480 Service Unavailable" (all ports in use), Asterisk reports congestion without

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I've been using tinc to do some high bandwidth VPNs between nodes in Amazon's EC2 environment (to work around some limitations there for effectively loadbalancing raw TCP connections while preserving the sources addresses, using Linux IPVS in NAT mode). The raw amount of traffic involved is probably making this a bit of a corner case for tinc. In the overall it has held up remarkably

Hi folks. I attempting to set up a Cisco 7960 (SIP) so that if the user is on a call, other incoming calls will ring through to her phone and can be answered. So far I have only been able to get this working by using the call-waiting function, which is cumbersome and does not properly allow the first call to be retreived. Is there a better way to do this? Thanks in advance. John Mensel