similar to: 1.4.22 CALLERID(num)

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

Hi guys, i just came to know that CDR(dst) field is set to current extension instead of the dialed no. i need to set it to DNID because our every user has 5 dids and i want to show the caller at the end of the month which numbers he dialed for every call, along with other cdr info. Our rating depends on the dialed number also. here is my extensions.conf exten=> 1212,1,Dial(SIP/rizwan)

All; I have implemented an "autodialer" solution where I create .call files for each number to be dialed. The .call file is very simple in design: Channel: SIP/2405551212 at broadvoice-outbound Context: autodial MaxRetries: 5 RetryTime: 600 WaitTime: 60 CallerID: Hildas Cleaning <4105551111> Extension: 2405551212 Priority: 1 Account: 999 Archive: yes A problem recently

Im trying to figure out the base way to check the callerID being sent to my Asterisk box and use it if it is a valid NANP number, but replace it with a static NANP number if it is not. (Why? I have a few carriers that require this, and a few international users - if it happens to take one of the carriers that require it, I want it to set a static number that is valid). I'm playing

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

Hi, Since we're moving from a legacy PABX that has been serving one of our customers for more than 15 years, we'd like this process to require no "human habits" change among the users. Software: Asterisk 1.4.22 Hardware: Polycom phones (mainly 430/601) Here are the "problems": We did configure call groups, pickup groups, ... - When someone picks up a call from

I thought that I would try this on iax.conf as well however I still get asterisk asterisk as the callerid name and num. I have the latest CVS as of 8/17/05 Has anyone have this working with iax incoming? Thanks Ben > That worked. The following line also got rid of "asterisk" without > entering any custom info: > > callerid= > > Thank you, > Hugh > > On

I'm testing GotoIf($["${CALLERID(num) but I'm missing something as it is not working: [office-open] exten => s,1,Wait(1) exten => s,2,Answer() ; for Caller ID is 471-5665, always signal congestion: exten => s,3,GotoIf($["${CALLERID(num)}" = "4715665"]?4:6) exten => s,4,Playtones(congestion) exten => s,5,Congestion(5) exten =>

HI There is a question about the fxo of the zaptel card which is set a number to use as common analog phone. When I use ${CALLERID(num)}to get it's number, it could'n be done. But ${CALLERID(num)} could get the other number of the SIP or IAX . Could you tell me the reason, and how I could get the number of the fxo which is used as a common analog phone? Thanks

I sent a question yesterday about having problems setting the caller ID. I turned on pri debug for both a good and bad call and I see this in the good call [2010-03-05 05:58:20.743] > [6c 0c 21 80 30 30 30 30 30 30 30 30 30 30] [2010-03-05 05:58:20.744] > Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) [2010-03-05

Hello, A question on channel originating (call files and AMI Originate): How can I change the CALLERID(num) var (because of the E1 provider needs), but having another n?mber (the original one) stored on the "clid" CDR field on the database? A channel agnostic solution would be the best one, without having to deal with the problem based on what type of Tech used for the outgoing

I'm having trouble setting callerid with teliax. I use a simple dial-out subroutine to set the callerid depending on the calling extension, and then dial out. Teliax is saying they're not seeing any callerid info. [DialOut] ; subroutine for dialing out. exten => s,1,NoOp(Context: DialOut called with outgoing number ${ARG1} ) exten => s,n,NoOp(${CALLERID(num)}XXXX) exten =>

Hi, I am new to asterisk , i am getting the following error,& the /etc/zaptel.conf file entry is as follows defaultzone=us loadzone=us span=1,1,0,esf,b8zs,yellow bchan=1-23 dchan=24 Parsing '/etc/asterisk/zapata.conf': Found Jul 1 18:33:35 WARNING[16384]: chan_zap.c:664 zt_open: Unable to specify channel 1: No such device or address Jul 1 18:33:35 ERROR[16384]: chan_zap.c:5296

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

Hello, I am running Asterisk 1.6 with Realtime enabled for my SIP users and peers. The backend is a MySQL database running through the ODBC backend in Asterisk. At this point everything works in terms of phones registering, placing calls between them, etc. However, I am having a problem setting the Caller ID number whenever I am using the Realtime database for the SIP users/peers. If I use a

via google, I found the reference regarding digit maps for the Polycom phones: http://lists.digium.com/pipermail/asterisk-users/2005-January/082884.html But I don't see any instruction for prepending digits to the number dialed. Does anyone know how to prepend a digit to the number dialed (from the Polycom side, not Asterisk)? I can do this pretty easily on a Sipura. i.e. Say I want to

Thanks for trying, what asterisk version do you use? вт, 25 июн. 2019 г. в 17:50, Doug Lytle <support at drdos.info>: > We have Polycom phones (I'm using a VVX601, the destination is a VVX301). > We're also on Asterisk 13. > > I forwarded my call to the VVX301 and then dialed my phones DID. The > forwarded call showed my cell phone number, so I cannot reproduce.