search for: broyles

...ce file is played from my dialplan, much like setting the monitor-filename. Something like this: exten => s,n, Set(PERIODIC_ANNOUNCE=foo) This would be a great feature if it doesn't already exist. Or perhaps there's a better way to do this. Thanks for your time. -- Regards, Robert Broyles

We love our Digium phones and DPMA - but we really need it to work on our Realtime Platform. Otherwise we lose all the cool features and they are just standard SIP phones. Anyone working on a solution for this? Or anyone from Digium see this on the roadmap?

Is there a way to set "na.rm=TRUE" as a default, so that this does not have to be re-specified for all of the functions requiring this option? Thanks in advance, Stephanie Broyles sbroyl@lsuhsc.edu [[alternative HTML version deleted]]

...I'm in need of getting this bug fixed. Bug has all of the details, but basically 1.4.22 broke it all. I've waited as long as I can - hoping the bug would 'resolve itself' - but now I'm putting a bounty out on it. http://bugs.digium.com/view.php?id=13691 -- Regards, Robert Broyles DISCLAIMER : This email and any files transmitted with it are property of Poornam Info Vision Pvt. Ltd. This email contains confidential information intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the se...

...om an outside line, and entering the digits during the read() part of my dialplan, it's accepting some of the digits twice, though it's only keyed in once. When testing the dialplan internally, it accepts only the digits that I key in. Anyone else experienced this? -- Regards, Robert Broyles DISCLAIMER : This email and any files transmitted with it are property of Poornam Info Vision Pvt. Ltd. This email contains confidential information intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the s...

...erride the queue wrapup time on the fly? I would like to allow a longer wrapup time for my agents, but if they are already done with closing up the call ticket, I would like them to be able to dial an extension or something to override the wrapup. Is there a way to do that? -- Regards, Robert Broyles

On Fri, May 1, 2015 at 10:43 AM, Robert Broyles <robert at webservicesaz.com> wrote: > We love our Digium phones and DPMA - but we really need it to work on our > Realtime Platform. Otherwise we lose all the cool features and they are > just standard SIP phones. > > Anyone working on a solution for this? Or anyone from Digi...

On 12/17/08 I updated to 1.4.22 from 1.4.21... Now the CDR data isn't recording calls where the caller hung up while waiting on the Queue. Sample CDR data BEFORE the upgrade: "2008-10-30 12:46:47";"\"John\"

Hi, My queue used to work fine until I upgraded to 1.6. I am getting the message: No application 'AgentCallBackLogin' for extension (default, 31001, 1) After some rearch I learnt that AgentCallBackLogin is removed in 1.6. Any one has a configuration that works in place of AgentCallBackLogin in 1.6. -- ond -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Sorry, I'm a newbee in Asterisk, and I want to call from one SIP trunk of Asterisk B (registered in Asterisk A as extension) to incoming call across another trunk of Asterisk B to extension of Asterisk C What the dial plan should be? Thanks -- We never did too much talking anyway So don't think twice, it's all right -------------- next part -------------- An HTML attachment was

It's my understanding that the backport is available now in 1.4. However, seem to be having some issues with it. Just wondering if I have everything setup right. I'm running 1.4.26.2 realtime. queue_members: `uniqueid` int(10) unsigned NOT NULL auto_increment, `membername` varchar(40) default NULL, `queue_name` varchar(128) default NULL, `interface` varchar(128) default NULL,

Is anybody using checkbox.cc to make iax2 calls? They have recently did some changes and my calls no no longer go through. They don't have a best service either, not replying to emails.. -- Joseph

Hi, So I'm using Asterisk Realtime Queues and Queue members on 1.4.28. I've noticed if there are no people in the queue when a call enters, even after a queue member enters, the call is never rang to him. From the debug, it seems that Asterisk is only grabbing the queue member list upon entering the queue. And not again until another call enters the queue. As a result, a caller will

...aller talking to themselves. The logs also show asterisk bridging the incoming SIP connection with the Phone, but when the agent hangs up, it shows the agent as hanging up, but it's not closing the bridge. Any ideas? Where should I begin in troubleshooting this? Thanks! -- Regards, Robert Broyles Team Lead - Customer Support Rep Poornam Inc aka Bobcares Phoenix, Arizona, USA 602.288.9145 DISCLAIMER : This email and any files transmitted with it are property of Poornam Info Vision Pvt. Ltd. This email contains confidential information intended solely for the use of the individual or ent...

So I'm noticing from the docs/ on Asterisk Addons 1.4.10 that the database structure for cdr_mysql is: CREATE TABLE cdr ( calldate datetime NOT NULL default '0000-00-00 00:00:00', clid varchar(80) NOT NULL default '', src varchar(80) NOT NULL default '', dst varchar(80) NOT NULL default '', dcontext varchar(80) NOT NULL default '', channel

Can anyone who has used both comment on the pros and cons ? Need to buy about 30 of these, for a small company with limited IT support. Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email

...DR Bug Fix To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <200903091100.46627.tilghman at mail.jeffandtilghman.com> Content-Type: text/plain; charset="utf-8" On Friday 06 March 2009 06:26:13 am Robert Broyles wrote: > Great backports! :-) > > This should really be merged into 1.4. This would violate the release policy for 1.4. -- Tilghman ------------------------------ Message: 8 Date: Mon, 9 Mar 2009 09:08:07 -0700 (PDT) From: Steve Edwards <asterisk.org at sedwards.com> Subject:...

Hi You all, Does anyone know any SIP client for BlackBerry? thank you -- TianLun Song We care your day to day business operation CCVP, CCNP, M.Eng Cell:1-647-868-2950 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090106/a278cbbd/attachment.htm

Can anyone point me to a working how to on state_interface? I found this little example: http://www.freepbx.org/v2/ticket/3496 But it didn't work with the backport on v1.4.x, so I have v1.6.2 installed on my test box, and it's still not working. Are there any special settings that need to be set for this to work? queues.conf: [test] ringinuse=no eventmemberstatus=yes

I'm trying to solve a problem I have with agents hanging up on callers before they even talk to them (caused by agents dropping their handset or something.) What I want is something like AgentLogin() where the agent has to press '1' to accept the call. Does anyone know how to get this to work with chan_local ? Thanks! Robert