similar to: SIP interface status and calllimit

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

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

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:

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

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

Asterisk 11.1.0 One queue with strategy=leastrecent. (Full queues.conf below.) Occasionally (several times today), a caller will get "stuck" in the queue - there are operators available to take the call, but the caller stays in the queue for a long time. Any idea what might cause this, or where I can start looking to debug it? I'm going to start digging through the queue log

On Mon, May 11, 2015 at 06:18:04PM +0000, Marc Fromm wrote: > Marc Fromm: > > Will the current RPM packages work on a RHEL7 system? I am upgrading > > from a RHEL5 system that uses the omega search tool. > > Since there have been no replies, I take it to mean that there is no > compatible package for RHEL7, thus I must remove the omega search > tool. I don't use

Using Asterisk 1.4, on the console 'show application queue' mentions an option 'i' that should "ignore call forward requests from queue members and do nothing when they are requested." Does this work? My assumption is that the member whose next according to the queue strategy should get the call even if they have forwarding enabled on their SIP device. The forwarding

Hi... I just had a real quick and simple question... I have a asterisk implementation setup w/ real time off of a mySQL database for SIP peers and queues, voicemail, agents etc... I after the upgrade to asterisk 1.4.3 there are some new configuration features i would like to use. I was wondering if i could just add to the database table a column for the new config option? if this will work or

Dear all on a debian amd64 i've installed (from source) asterisk 1.4.30 On the system we have in average 50 concurrent calls in queue and 40 sip members. I'm experiencing an apparently random problem: sometimes some users receive 2 calls from asterisk, apparently ignoring the ringinuse=no settings. It appears on users that are members of many queues As you can see from the log, the

My settings for the queue.log are in the [general] section of logger.conf I'm running 13, I didn't see what version you said you were running. If I wanted to add a LOCAL channel to my queue I'd do it as member => LOCAL/7124 at kiniston-intern,0,John,hint:7124 at kiniston-intern On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch <idemkovitch at yahoo.com> wrote: > John,

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

I am confused on the various account creation methods in samba. I am running Fedora Core 8 and Samba 3.0.26a-6. 1.) I can create a user with pdbedit -a I can create a user with smbpasswd -a I can list the users with the pdbedit -L command Where is the file with the users that the above commands create and pdbedit -L lists? 2.) In SWAT under "server password management" I created a use

When agents are on the phone, and the CLI queue show command shows their status as busy, the queue still tries to send them calls. Running Asterisk 1.6.0.17 and using AddQueueMember to dynamically add agents. ringinuse is set to no for queue. Agents are using Polycom 430s. dialplan: exten => s,n,Queue(orders,itT,,,80) queue definition in queues.conf: [orders] maxlen=20 queue-thankyou=

Just a silly question that I'm not sure. Ringinuse is working with IAX in 1.6??? like in sip?? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090306/fa56ebfc/attachment.htm

Hi, I am using asterisk-1.4.15, My sip configs is like [2501] type=friend username=2501 secret=2501 canreinvite=no host=dynamic dtmfmode=rfc2833 context = sip disallow=all allow=ulaw incominglimit=1 nat=1 queue.conf is like [gen-enq] joinempty = yes musiconhold = default strategy = rrmemory servicelevel = 60 timeout = 60 retry = 5 wrapuptime=5 announce-frequency = 90 announce-holdtime = yes

Hi, I use asterisk 1.8. My issue is : I have the same SIP members added to two queues. I use realtime configuration and has set the field ringinuse=0 for both the queues. But if an extension is answering the call in one queue, and some new call comes in the second queue, still that extension is ringed. In the queue_log table I am getting RINGNOANSWER events each second for the extension until

I've an Asterisk server running in my office, which forwards all long distance calls to a third party SIP service using an extension rule: exten => _1XX0.,1,Dial(SIP/{EXTEN:4}@external_sip_server.com) (1XX0 is the international calls rule for Chile) Also, in my sip.conf, I've defined canreinvite=yes to decrease the network load to the server caused by the RTP. However, the external

I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred agents operating simultaneously in peak hours. The system uses

Nothing much to be said.. I backported ringinuse, autofill and the QueueLog application from 1.4.0 to 1.2.14. Any or all may be applied - order doesn't matter. They have received minimal testing but appear to function correctly. As always with these things, don't blame me if they connect your callers to a phonesex line, etc. http://bum.net/patches/ Cheers, Gavin.