similar to: Need help with implementing prepaid in asterisk

Hey guys, It's me again. Today I moved our NAS from our old 2000 domain to a new domain presided over by two Samba 4.7.2 domain controllers. After the move I cant access the NAS at all from my Windows 7 test pc. I keep getting an error that "The group name could not be found" I am at the end of my troubleshooting skills. I also moved the NAS' samba from sernet-samba 4.1 to

Hi Roland On Wed, 10 Apr 2019 at 12:00, Rowland Penny <rpenny at samba.org> wrote: > On Wed, 10 Apr 2019 11:41:52 +0200 > Ian Coetzee <samba at iancoetzee.za.net> wrote: > > > Ho Roland, > > > > Replies inline > > > > > > > > > The only user I have is the jeadmin user which is the domain > > > > admin as well as a

On 16 November 2017 at 10:29, Rowland Penny <rpenny at samba.org> wrote: > On Thu, 16 Nov 2017 08:40:01 +0200 > Ian Coetzee via samba <samba at lists.samba.org> wrote: > >> Hi Rowland, >> >> > How did you upgrade ? >> >> I upgraded by merging the latest code from the samba git repo, doing a >> git clean to remove old local files. Then

Hi all, when a user dials any number, asterisk automatically generates ringing which caller can hear, and after 2 - 3 rings asterisk detects that the called user is busy, then caller hears busy tone. for example user hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing at the start so that user hears only beep beep beep if the called user is busy. I have used the R

Hi Rowland, Sure, I have pastebinned the configs (and done some public ip masking) DC1 - bind config https://www.jacklin.co.za/privatebin/?be125b7e578c53d4#q9nGwU3f9Tz7wtHLTf3UIcFhz/GIJjryq6/cN2rip1k= DC2 - bind config https://www.jacklin.co.za/privatebin/?c1c921a4289a4e91#URHcPgK0B1fgeoTCeWXL6QDKdUxR6YpHZ1dcwXR44Iw= DC1 - Samba Config

hi all, i just finished developing some incoming call features in a macro. that macro gets executed everytime an incoming call is received and a new mysql connection is made using the MYSQL cmd in dialplan. i want to use a single mysql connection for every incoming call. my idea of doing it is like this, i want to get a mysql connection in a global variable, just to share the connection with

Hi all, Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? -- Best Regards

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)

Hi all, Is there anyway i can prevent multiple sip registrations from different IPs using single username in asterisk. Does asterisk provide any aid in this respect? As far as my knowledge is concerned i dont think there is any support for this in asterisk, so i think i'll have to makeup a script which sniffs sip packets coming for asterisk and detect for multiple register requests coming from

Hi all, I need a fully developed web based remote provisioning system. I cant find anything reliable on the internet. Have already checked ataconfig.com and voxilla-ays.com. have tried to contact them but got no response. So if anybody knows a good provisioning system then plz tell me about it. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part

Hi all, There is a problem with my dialplan. here is the dialplan: exten=> 123,1,Dial(SIP/U1,,Ttg) exten=> 123,2,Hangup exten=> h,1,AGI(onhangup.pl) The problem is whenever U1 is called or calls someone, if U1 hangsup the call then the h extension is NOT executed. but if the other person hangsup the call, then the h extension is executed (assuming that the other person is calling

Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Ho Roland, Replies inline On Wed, 10 Apr 2019 at 11:03, Rowland Penny <rpenny at samba.org> wrote: > On Wed, 10 Apr 2019 10:25:25 +0200 > Ian Coetzee <samba at iancoetzee.za.net> wrote: > > > Hi Rowland, > > > > Please see my replies inline. > > > > > > Possibly, but it could just be down to you not having this line > > > in

Hi all, We are planning to shift our sip users from one platform to another. (basically from one asterisk server to another). the problem we are facing is that both asterisk servers are using different ports to listen for sip. and both have live customers on them. provisioning their ata's is not a good option for us coz of our settup. we cant just ask the customers to change their ports for

Hi im trying access the www.voip-info.org website since yesterday but i cant open it. google search diaplay correct search results but it doesnt open when i click the link. it displays a message about tcp error which says -->"There was a problem communicating with the server". I dont know what the problem is. I just want to ask whether their server is down or not and is everybody

Gid 4294967295 = Nobody (32-bit) Thats your problem. A problem in the nss mappings. https://lists.samba.org/archive/samba/2017-January/205672.html Is the a simular thread to your problem, ready it. Greetz, Louis > -----Oorspronkelijk bericht----- > Van: samba [mailto:samba-bounces at lists.samba.org] Namens Ian > Coetzee via samba > Verzonden: woensdag 10 april 2019 15:37

Hi all, i need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was

Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost

Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to

Hi Rowland, Please see my replies inline. On Wed, 10 Apr 2019 at 09:58, Rowland Penny via samba <samba at lists.samba.org> wrote: > On Wed, 10 Apr 2019 09:04:06 +0200 > Ian Coetzee via samba <samba at lists.samba.org> wrote: > > > Hi All, > > > > I have a very weird issue on one of my servers. I think I might just > > be missing something quite