similar to: Cisco 7960 Line rollover for secretary's phone.

Displaying 20 results from an estimated 2000 matches similar to: "Cisco 7960 Line rollover for secretary's phone."

2005 May 12
2
Cisco 7960 Can't be unlocked
Odd problem here--I just got a couple of Cisco 7960s from Ebay that are not functioning as expected...... These 7960s can't seem to be unlocked for manual configuration via any mechanism that I can find. If you go to settings, there is no option 9 (unlock). Available options stop at 4 (Status). **# has no effect. The Phones report that thier current firmware version is 3.1 MF.G2.
2005 May 18
5
SIP Phone Recommendations?
Hi all. I'm in the process of putting together a new Asterisk system as a proof-of-concept, and wanted to see which SIP phones all of you had the best luck using with Asterisk. ?I've just come off a very trying experience with some Cisco 7960s, and am looking for something else to round out the phones on our network. This is a small setup, for no more than 20 users total. ?We need at
2005 Jun 14
2
Features.conf for secretary function
Hi, I am trying to use the attended transfer. So I put this in my feature.conf: [general] [featuremap] atxfer => *0 blindxfer => #0 I completly restart asterik, and not just make a RELOAD. But during a call, when I press # it runs a blind transfer and if I press * I am disconnected. I am using the CVS version of * get as explain here
2004 Jul 08
1
Rollover oddity
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
2004 Aug 06
3
Mailing list rollover
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 ---- List archives:
2004 Aug 06
3
Mailing list rollover
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 ---- List archives:
2004 Aug 06
3
Mailing list rollover
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 ---- List archives:
2004 Aug 06
3
Mailing list rollover
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 ---- List archives:
2004 Aug 06
3
Mailing list rollover
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 ---- List archives:
2003 Sep 21
2
Incoming phone line rollover / hunt?
-----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
2004 Jul 16
0
[Bug 1529] New: 32bit rollover problem rsyncing files greater than 4GB in size
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
2007 Dec 31
2
USB HID - interrupt reports
Since reports received over the interrupt pipeline are a recurring problem for various types of UPS'es, I propose to simply ignore the data we receive there and only flush the respective report buffer. By doing so, at the time the interrupt reports are processed the first variable that is retreived will trigger a poll for the corresponding feature report and we should be fine. The impact on
2006 Jun 15
2
rollover simulation
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
2005 Jan 05
2
Allowing "pooling" or "rollover" for inbound calls on VoicePulse
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
2009 Jan 19
1
Suggestions on how to create a hunt or hunt like (rollover, multi-line) group or where to get one?
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 --------------
2005 Mar 20
2
IPSwitchBoard-BETA Update
Release 0.66 of IPSwitchBoard is now available for FREE download at: http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA Enhancements: Support for Call Parking and retrieve/forward them again. Last Call on the Queues Page now displays a date-time in human readable format. Added CallerID on the Queue Members listing on the Queue page. New page with Agent information. Minor bug
2006 Oct 30
2
Possible log rollover mutex problem
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
2006 Jun 12
0
how to get link_to mouse rollover to pop up a "help" box.
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/.
2005 May 06
1
Polycom 600 rollover
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
2005 Aug 24
0
SIP trunk rollover problem
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