similar to: Queue cmd option 'i'

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

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

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

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

Can anyone explain why when I hit * (as in *2 to transfer) a call that has come to me in a queue asterisk disconnects the call? All I have to do is hit "*" and the call drops.

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

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 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

James Fromm wrote on 4/15/15 12:45 PM: > Odd, we've used a FQDN for our netbios domain for 15 years. I'll give > it shot. > > On 04/15/2015 12:40 PM, Robert Moulton wrote: >> Sonic wrote on 4/15/15 12:34 PM: >>> On Wed, Apr 15, 2015 at 3:23 PM, James Fromm <fromm at omnis.com> wrote: >>>> workgroup = CUST.OMNIS.COM >>> >>> My

Sonic wrote on 4/15/15 12:34 PM: > On Wed, Apr 15, 2015 at 3:23 PM, James Fromm <fromm at omnis.com> wrote: >> workgroup = CUST.OMNIS.COM > > My understanding is that workgroup names are NetBIOS names and are > therefore not hierarchical like domain names (no dots, 15 characters, > etc.). > > Chris > We encountered this too, when trying to join Win 8 and Win

Wow, this is great. However, we need to use DNS. If we enable it AFTER the computer has joined the domain, would it keep working, or DNS is no avail in W8.1? D?a 15.04.2015 o 22:03 James Fromm nap?sal(a): > That worked. Server joined just fine. Thanks Robert! > > Thanks, > James > > On 04/15/2015 12:48 PM, Robert Moulton wrote: >> James Fromm wrote on 4/15/15 12:45 PM:

I don't have anything but Server 2003, 2008 and 2012 to test with. 2003 joins the domain without issue. 2008 and 2012 will not. The registry has been updated on both: Windows Registry Editor Version 5.00 [HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\services\LanmanWorkstation\Parameters] "DNSNameResolutionRequired"=dword:00000000

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

Is it still possible to join a Windows 2012 Server R2 system as a member to a 'pre-NT5' Samba (3.6.23) domain controller? The Windows 'Domain Change' GUI errors immediately after failing the SRV lookup for the AD server. Even with the SRV record in place, the GUI fails trying to connect to the non-existent LDAP port. Netdom on the command line tries to work if the DC name

Actually, I have tested this here with an Aastra 9133i and an Asterisk@Home server, and the 9133i will re-subscribe on its own after an Asterisk reboot, if you wait long enough. It took on the order of an hour to do so. Of course, a phone reboot will get it done faster, if necessary, but it _will_ eventually re-subscribe on its own. In another thread, I've seen a response that the GXP2000

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 >>>> community get discussed on the lists?

We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James

Hi, My setup: Solaris 9, mbox format, mailboxes NFS mounted from another S9 system, imap. I got the following assert yesterday: Jan 30 19:57:15 emerald dovecot: [ID 107833 mail.error] imap(user): file index-mail-headers.c: line 258 (index_mail_parse_header): assertion failed: (part != NULL) Jan 30 19:57:16 emerald dovecot: [ID 107833 mail.error] imap(user): file index-mail-headers.c: line

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