similar to: Limit // incoming calls to Queue Agents

Hi, I seem to have a problem with chanisavail and the call limits on sip phones(incoming and outgoing) The problem seems to be that chanisavail when trying create to create channels and hanging them up afterwards screw up the current usage limit on the phones. Example with chanisavail: Phone A calls voicemail (usage now 1) Phone B tries to call Phone A and uses ChanIsAvail in the dialplan.

Hello. I was hoping someone might be able to help me with the following problem: When an incoming call comes into to * I would like it to attempt to find the first extension in a group of extensions that isn't busy and to send the call to that extension. Should that extension not be picked up or if the phone at that extension is on DND I would then like * to send the call to voice mail

I got a Uniden UIP200 and started to configure it and I am lost.... I have a tftp server setup on my * server and have the files unidencom.txt and uniden<mac>.txt there. But it doesn't quite work yet. It registers as a sip phone (sip show peers), but I cann't dial it and the display shows #1 disconnected all the time. It has firmware version BS4.59a in it. I have no idea if I

We're running a large cluster, and are leery of upgrading them all to 5.5, but would like to find the latest security patches. Is there a repo for this? I can't seem to find a 5.4 specific update repo with anything since last March. Is 5.4 EOL'd? Thanks, -- Mitch -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I curious to know if other UIP200 users have this same issue: You flash (XFER button) to consult-transfer a caller to another extension. If the transfer target party is unavailable (ie: voicemail), there appears to be no way to get the original caller back. If it's a known limitation, has anyone come up with a functional work around? Thank -- ..................................

Hi, Here are some top reminders for the 3.12 release: 1) When 3.12 is released 3.8 will be EOL'd, hence users are encouraged to prepare for the same as per the calendar posted here. 2) 3.12 is a long term maintenance (LTM) release, and potentially the last in the 3.x line of Gluster! 3) From this release onward, the feature freeze date is moved ~45 days in advance, before the release.

Hello, I''m struggling to handle the amount of memory that dom0 gets on some system. It''s a system with 8GB of RAM, running 2.6.38-6 with pv_ops and Xen 4.1 If I want to set the amount of memory for dom0, I get very bogus results. See here: http://pastebin.com/gRgY9ERN If I use the M notation (xl mem-set 0 4096M) the outcome is similarily unpredictable, and sometimes I get the

Yeah, about that... Back in '12, we got a nice HP poster printer. They don't support Linux, but a co-worker got the .ppd out of the Mac support file. I tried it, then I looked at the file in vi... and found it *only* supported the 24" printer, *not* the 44" printer that we had. Well, we just bought a replacement, since the old printer was EOL'd. This time, I went to hplips,

Hello, Maybe it''s a silly question, but I cannot find anything in Rails changelog or google it out. There is a Rails::Version module (since version 0.14.something) that gives me a version number of Rails, similarily to Rails::Info module. Apparently in Rails version 1.0 (or maybe sooner) the name of the module was changed from ''Version'' to

I couldn't be happier with the simplicity of this - but it's a hack! Hi all, I'm currently using a SIP client (BT101) to connect via DSL to a remote instance of Asterisk. - Asterisk has a private IP behind my OFFICE router. - The SIP client has a private IP behind my HOME router. I'm doing this _without_ the use of STUN or proxy servers. Here's how it works: -

The jitter buffer would "freeze" under the following condition: - The sender and receiver are in sync. - The sender machine freezes for a few seconds while the receiver does not. - This causes all the packets sent by the sender to have a timestamp that is too low to be considered, meaning jitter_buffer_put ditches it before it checks if lost_count > 20 to reset. - The jitter buffer

Dear R helpers I have following dataset rate_number = matrix(c(5, 15, 60, 15, 5, 0, 20, 60, 20,0, 10, 20, 40, 20, 10), nrow = 5, ncol = 3) range_mat = matrix(c(6.25, 6.75, 7.25, 8.75, 9.25, 9.75, 8.5, 9, 9.5, 10.5, 11, 11.5, 4.25, 4.75, 5.25, 5.75, 6.25, 6.75), nrow = 6, ncol = 3) > rate_number        [,1]   [,2]   [,3] [1,]    5     0      10 [2,]   15   20      20 [3,]   60   60      40

Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the "very inexpensive" Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to

Another day, another apparant unexplained hardware incompatibility. I have a TE412P and a TDM400B living quite happily in a whitebox using an Intel motherboard: http://www.intel.com/design/servers/boards/se7230nh1-e/index.htm I tried to move to an IBM x3650 system. It uses a slightly newer chipset, but apparantly it's in the same family. The SE-7230 board has been EOL'd and the

On 01/06/17 22:47, Shyam wrote: > Hi, > > Here are some top reminders for the 3.12 release: > > 1) When 3.12 is released 3.8 will be EOL'd, hence users are encouraged > to prepare for the same as per the calendar posted here. > > 2) 3.12 is a long term maintenance (LTM) release, and potentially the > last in the 3.x line of Gluster! > > 3) From this release

Shanmuhka wrote: > I looked at the sanjoys patch for SIV Test. And i figured out that this is exactly what i need. > as the comments said, check if subscript A can possibly have the same value as B in analyseSIV(A,B) > but i didn't get How to use this information ? > > lets just say in the above program > When i use depends function it shows the dependency from load of x to

Hello all, The "Secret Agent" final release of the Asterisk Management Portal is now available for download: http://amp.coalescentsystems.ca/ This exciting new release adds a great deal of functionality and flexibility. Thank you for all the contributions and feedback! 1.10.007 - Added AMP Users (multi-department, basic multi-tenant) - Added incremental upgrade script

On Tue, 4 Jan 2011, Andy Graybeal wrote: >> The Polycom 321 has not been EOL'd and supports VLAN. It is, however, >> lacking a 2nd ethernet port if you were to go that route. >> >> -M >> > Thanks for the response Mark. I see the 331 has two ports and the same > features as the 321. > > I'm wondering what phone would be best being used as an

Is there a way to trigger an action when a certain number of zap channels are in use, or is there a variable that stores max used channels that can be read? I use PRI for inbound calls, but outbound goes out via SIP, so the simple solution does not work. I need to know when the potential exists for inbound calls via PRI/Wildcard to be blocked because there are no more channels. Obviously

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Tim, *! Thank you for posting your nice apache2 recipe to the wiki! It''s really good to see how others make things happen and a good example for others. One little thing though caught my notice because I''m currently making the same naughty thing on Simple Text Recipes. There is no ensure attribute. Since you already have done