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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

...rything 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 [mailto:asterisk-users-bounces@lists.digium.com] F?r BJ Weschke Skickat: den 23 februari 2006 13:44 Till: Asterisk Users Mailing List - Non-Commercial Discussion ?mne: Re: [Asterisk-Users] Polycom 501 ACDlogin On 2/23/06, jan.sarin@securia.se <jan.sarin@securia.se> wrote: > Hi, > > I have several Polycom 501 connected to asterisk. The phone has...

Has anybody got the polycom acd function to work? I have the following setup: Debian 3.1 - 2.6.8 linux zlib-1.1.4 libpri-1.2.3 zaptel- 1.2.6 Asterisk - the bweschke/polycom_acd_funtions branch version - I get one error when doing a make install about needing a newer version of libpri and zaptel, I got the above versions from asterisk.org, are there newer version anywhere else? In the sip.conf file I have set the agentlogin=yes and agentcbcontext=demo (demo as...

...Behalf Of Pisac Sent: Saturday, January 14, 2006 9:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 1.2.1 "Silence suppression is disabled" whatthehell? Asterisk 1.2.1-BRIstuffed-0.3.0-PRE-1f ----- Original Message ----- From: "BJ Weschke" <bweschke@gmail.com> Where did you download this 1.2.1 version of Asterisk from? These messages are coming from a patch to Asterisk that should not be in any version of the 1.2 branch. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com...

On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp <jcolp at digium.com> wrote: > On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote: > > I get the desired use case to run app_amd from within a Stasis > > application, but I’m not sure about app_queue. You have everything at > > your disposal within ARI itself to replicate all of the functionality > > of app_queue and beyond. > > Yes, there are...

Hi, I have callcenter running with v 1.2 with AgentCallbackLogin and now trying to move to 1.4 using the example doc, doc/queues-with-callback-members.txt. From what I understand the basic idea in the example is to 1. Authenticate a caller with VMAuthenticate 2. Get his SIP Channel number 3. Use

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

Where are user's voicemail passwords stored and how does the asterisk administrator change them? TIA, Jeff Heath

I want to replace a custom PBX, that is infront on a IVR system based on OLD NMS AG-E1 Card. The Cards is configurated with CAS Digitalmode, someone can give me some info about Digim Cards CAS configuration i need a conversion Table? I wanto to don't touch configuration on winbox, i want only replace HWPBX box with asterisk. Diagram Telco E1 ===>Proprietary PBX========(CAS)===>IVR

BJ, >BJ Weschke <bweschke@gmail.com> >Subject: Re: [Asterisk-Users] Do Both! :) Re: Telecom >SIP termination vs. DS3 >To: Asterisk Users Mailing List - Non-Commercial >Discussion <asterisk-users@lists.digium.com> >Message-ID: <79cf63305051908056c284cc9@mail.gmail.com> >Content-T...

Anyone have an opinion about these guys and their recent performance? I need some local DIDs and they provide for my area code.... Thanks, Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050511/365bc7b0/attachment.htm

...tand binary and those who don't. > >Gerry Gilmore >Field Applications Engineer >Intel Corporation >(<http://www.intel.com>http://www.intel.com) > > >From: asterisk-dev-bounces@lists.digium.com >[mailto:asterisk-dev-bounces@lists.digium.com] >On Behalf Of BJ Weschke >Sent: Friday, September 23, 2005 12:29 PM >To: Asterisk Developers Mailing List >Subject: Re: [Asterisk-Dev] Open source time card application for Asterisk > > From an infrastructure perspective, you're right. > > From an ASP perspective, you're wrong. > > htt...

Hi Asterisk users, I'm one of Asterisk ARI users, and trying to designing the new ARI for application execution in Stasis(). This will be made possible for executing the applications in the Stasis() application. But, before going further, I would like to know which application needs to be considered. Because this feature will introduce new Stasis behavior, I would like to test the

...t settings to try make it |happen unless I am doing something wrong or not waiting long |enough for the |phones to re-subscribe. I must have tested it for at least 3 |hours and BLF |never came back. I confirmed it with the Asterisk CLI as well. | |> -----Original Message----- |> From: BJ Weschke [mailto:bweschke@gmail.com] |> Sent: Thursday, March 23, 2006 5:34 PM |> To: Asterisk Users Mailing List - Non-Commercial Discussion |> Subject: Re: [Asterisk-Users] Re: Fw: anybody has SIP |> realtime working ? |> |> On 3/23/06, mustardman29 <mustardman29@hotmail.com> wr...

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 Ast...

Anyone know how to fix this? gcc -shared -Xlinker -x -o app_rxfax.so app_rxfax.c -lspandsp -ltiff In file included from app_rxfax.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:302: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/include/asterisk/lock.h:302: error: (Each undeclared identifier is reported

...th the 1.0.x releases). We want to extend our thanks to all the community members whose contributions have made this release possible; without their support, testing and other involvement we would not have reached this milestone so soon! I want to extend special thanks to Russell Bryant and BJ Weschke who put in many hours in the last two weeks doing 'janitorial' code updates that nobody else wanted to do. Thanks to you both!

I'm trying to make H.323 trunk between AVAYA&Asterisk. But call from AVAYA is terminated inmediatelly when apps DIAL on Asterisk is started. Does any one use AVAYA and h.323 channel? Thanks Bob.

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.

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