similar to: Timeframe for QueueStatus values

There are 6 different ${QUEUESTATUS} variable values defined in asterisk 1.2, I am attempting to make sure I have a full understanting of when they would be set; If someone could correct errors with these definitions ot would be appreciated; TIMEOUT - the max time specified in the queue command elapsed, only checked between retries so may not be 100% accurate. FULL - the number of callers

I went to put in the new sound files over the weekend, but forgot to backup the custom folder and lost my custom digital receptionist files. I then had to copy the old files back from a duplicate machine. The problem is now though that voicemail just hangs up when I dial it. Other apps work - *70 for example gives me 'call waiting...activated' so I know it's accessing the files

Hello, I'm trying to write a php script that issues the QueueStatus and Queues manager API commands to Asterisk and records the returned event data into a MySQL table. On voip-info.org, I see that QueueStatus returns such things as Queue, Max, Calls, Holdtime, Completed, Abandoned, ServiceLevel, and ServiceLevelPerf. However, there are no descriptions for these values. I've tried

Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Hello everyone. I got a little problem in here: I want to set up a queue so that if anything of these happens: a) No agents logged in b) All agents busy Then the user gets diverted somewhere. I used this (for testing purposes only, of course): exten => 7080,1,Answer() exten => 7080,n,Queue(teste) exten => 7080,n,Goto(${QUEUESTATUS}) exten => 7080,n(ERROR),NoOp(${QUEUESTATUS}) exten

I see it's been in the wishlist a while. Any possible timeframe for getting it working, or maybe a patch (or is it too big a change) to support it? I'm building a new server to move an existing userbase to. So far I have working: Postfix (with Maildir quota patches that work fine) Amavis-new with ClamAV scanning for viruses (will add SpamAssassin soon) SquirrelMail IMP (Horde)

Is there a specific file with the version number in it that I would modify, or do you just mean the file names? Sincerely, Rob Groner From: Charles Lepple [mailto:clepple at gmail.com] Sent: Monday, March 09, 2015 9:23 AM To: Rob Groner Cc: nut-upsdev at lists.alioth.debian.org Subject: Re: [Nut-upsdev] New sub-driver submission process timeframe? On Mar 9, 2015, at 9:09 AM, Rob Groner

I recollect that there used to be a fixed, finite limit to the number of call groups that could exist. Does anyone know if that limitation still exists in 1.2.1, or maybe where I could look in the code to find out if it's a fixed length array or not? Thanks. Doug.

Using 1.0.9: If I have: exten => s,1,Dial(SIP/5555&SIP/12345@192.168.1.1) How can I return the DIALSTATUS variable for the second SIP channel ONLY if the second SIP channel is busy, regardless of the dialstatus of the first SIP channel? What I want is, if the second SIP channel is busy go to n+1 or n+101 regardless of the status of the first SIP channel. tia

Hello all, I think Lee has given me a head start, but I'm still running in a circle (at least i'm in the lead). The problem is with my queues. The phones go to their own voicemail after 5 rings. That's about the same time I allow the phone to ring before trying another phone in the queue. Is there a way to tell asterisk....? If this call is coming from a queue, do not follow a

I'm pretty sure I already know the answer to this, but... Is there a way to copy/transfer/replicate sip subscriptions from one asterisk system to another, for the purposes of HA? You coudln't even write a script to do it I don't think. You can do an 'asterisk -rx sip show subscriptions' but there'd be no way to repopulate it on a second system. Yes/No? Doug.

Hi all, I know I'm probably stirring up a hornet's nest with this question/comment but I've spent the last few days working on a PHP-based class for the manager interface as we're preparing for a pretty big upgrade at our call center and I'm revamping all of the management apps I've written. I can connect to the manager interface and send query strings back and forth all

Hi, I have several Polycom 501 connected to asterisk. The phone has an ACD-login function that I'd like to use. But I can't find find much information about this. I've read a post on bugs@digium (http://bugs.digium.com/view.php?id=6119) about this function but I'm not really clear on if this is actually working or not? Has anyone actually used the Polycom ACD-login function

Hi, Is there any way that I can store my manager API output that is: My question is that is there any why using that I can get the QueueStatus and store the result in some text file for further processing. <?php $strHost = "127.0.0.1"; $strUser = "cron"; $strSecret = "1234";

Thanks! Do you have any suggestions on what I might do next. I have the phones, I have asterisk, and I have everything setup. But i can't get the login to work with the Polycom function. Nothing happens...and I can't find any readmes' or manuals. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com

I don't have any problem with pointing customers to a branch they can clone via git and compile. But until that's available, I will do what you suggested (and what my boss has been suggesting for a while) which is to take 2.7.2 and put our driver in it nice and happy and then package that up for our customer to use. That will be a more than sufficient means to get our UPS working until

Is there an app or softphone for meetings that displays the hosts screen like webex or intercall. Jordan Novak -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060321/df90d527/attachment.htm

Does anyone know where I can get replacement handsets for the Polycom SoundPoint IP phones? Or does anyone have any they want to sell? From the looks of it you have to buy a whole new phone to get a new handset. My vendor, TriaTechCOA, told me I had to buy a whole new phone to get a handset, which is pretty ridiculous. Maybe there is a more sane vendor I should be buying from? Thanks, -Ryan

I've noticed that when app_queue.so is reloaded(or just a reload command is used) that all queue members that were paused are automatically unpaused. Is there a workaround for this? (Note, I use statically defined callback agents). --johann

This doesn't seem to be my day for getting things to work right.... The most obvious place I could find to change the version number was near the top: AC_INIT(nut, 2.7.2) I changed to AC_INIT(nut, 2.7.2_RTD) However, I cannot find "autogren.sh" anywhere, whether in the nut dir or as a system executable. I also tried "autogen.sh" in case that was a typo...no luck.