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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

...with asterisk? > How did you do it? > It is indeed functional. It's in production at the present time at one of our call center clients. I believe there is a README in the docs of the branch that outlines setup of the functionality. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

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.

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

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

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, 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

...privacy concerns, but at least with this approach the barrier to get it working again is generally in the hands of the user or their administrator.=20 I fully agree Maybe simply asking when a user logs in is an acceptable compromise. =20 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ =09 =09 =09 =09 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- =09 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users=20 =20 ------_=_NextPar...

...fter |> restart. Unfortunately, unlike registrations, there's no |> guick/tactical way for us to keep track and "reseed" a |> subscription immediately after a restart as we do with registrations. |> |> -- |> Bird's The Word Technologies, Inc. |> http://www.btwtech.com/ |> |> |_______________________________________________ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users |

Is there any technical difference between a T1 cable and a cat5e patch cable as far as using them with Digium T1/E1 cards? Can PRI circuits terminating at a smart jack connect successfully to Digium cards using straight through CAT5e cables? If so, are they using all of the pins in the cable? Thanks in advance

Has anyone had any success using the MixMonitor() plus "command" as nothing I have tried works. I am using 1.2.1 I did google the archive but couldn't see any mention of anyone using this. What I am hoping to do is run a macro on hangup, current method I am using seems to miss some calls 5% of calls fail to mix / convert to mp3 etc. Was hoping that MixMonitor would fix this.

On Oct 22, 2014, at 11:47 AM, BJ Weschke <bweschke at btwtech.com> wrote: > On 10/22/14, 12:14 PM, Paul Albrecht wrote: >> On Oct 22, 2014, at 10:33 AM, Joshua Colp <jcolp at digium.com> wrote: >> >>> Paul Albrecht wrote: >>>> Really? Shouldn?t something this major affecting the entire Asterisk >>>> c...

Hello, I recently switched over to using Mixmonitor versus Monitor to see if it would clear up some warble that I was getting in my recordings. It did indeed clear that up, but a new problem was introduced. The recordings for no reason will just end abnormally. There is no rhyme or reason as to when they will end, but usually after a minute or so. Here is my current setup. Asterisk v. 1-2-0rc2

Hi, I want to setup a pause/break between calls coming from a queue to an agent. If an agent is logged in only in one queue the wrapuptime parameter works. But what can I do if the same agent is in more then one queue? I found no parameter to configure a pause after a (incomeing) call and the next call (asterisk CVS-HEAD-04 / polycom ip500 sip phones) thanks a lot for help ciao, morel

I upgraded from 1.0.9 to 1.2.1. In 1.0.9 everything worked perfect. Now, I call in my IVR, and after navigating in menus when I get dialtone for dialing extension, Sound is choppy and I get bunch of messagess: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence

According to this page: http://www.asterisk.org/doxygen/Config_sip.html canreinvite=yes redirects just the RTP. I was under the impression that the entire SIP connection got redirected, therefore losing accounting ability. Could someone clarify this? --Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, this is probably a stupid question but can someone point me to where I can find the pinouts for the connector at the back of the TDM2400P card from Digium. I am assuming this is some kind of telecomms standard but I need to connect to 8 existing analogue lines and therefor need to wire this connector to a patch panel or R11 jacks. Also most of the docs do not mention this card - am I

I need to send calls to a bank of servers in a round robin fashion. Has anybody implemented this in dialplan language? If not, in some other fashion. </edg>

Have any providers started to offer T.38 yet? I am anxious to find a solution for faxing. -- Chris Mason -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.