similar to: Dahdi dial plan

I'm running FreeBSD 5.4-STABLE with Samba 3.0.20 configured as a PDC serving a small-ish network (10 XP hosts). My problem concerns roaming profiles - some XP clients take ages to logon and logoff. On doing some Googling I identified the 'profile acls' parameter as being a likely fix. However, I seem to have got myself into one of two scenarios: a) With 'profile acls'

There is a description of all the possible linkage types at http://llvm.org/docs/LangRef.html#linkage – does this answer your question? (Basically, an extern_weak resp. ExternalWeakLinkage symbol becomes null instead of being an undefined reference) David On 3/16/11 7:06 PM, Junjie Gu wrote: > What is the difference between WeakAnyLinkage and ExternalWeakLinkage > ? They are defined

What is the difference between WeakAnyLinkage and ExternalWeakLinkage ? They are defined in GlobalValue.h. Thanks Junjie

https://bugzilla.samba.org/show_bug.cgi?id=2628 wayned@samba.org changed: What |Removed |Added ---------------------------------------------------------------------------- Summary|hangs indefinitely (while |hangs indefinitely at start |listing files) |of phase 1 ------- Additional Comments From wayned@samba.org

I am porting over the portable version of openssh to our uCLinux implementation. Everything has worked with minimal effort and I appreciate all the work. But, I am having a problem whereby the sshd executable is crashing and I really could use some help on where to look at this in more details. Here is how I start up the sshd for testing. /usr/sbin/sshd -D -ddd -f /etc/ssh/sshd_config -p 65

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Hola, compañeros. Pido ayuda con algo que sé que tiene que ser simple, pero la presión de tener que sacarmelo de encima me simplifica a mí demasiado y no me doy cuenta. Tengo una matriz de datos en la que tengo características tipo factor, necesito trasponer esa información a una matriz de datos binarios, en función de algunas combinaciones de esas variables. Unas combinaciones tienen que pasar a

Hi, We are running a samba fileserver, access controlled using posix acl (right 770, with users/groups on the filesystem level. Therefore samba shares look like this: [share] path = /srv/academic read only = no writable = yes create mask = 0770 directory mask = 0770 Now certain users complain that they cannot access certain folders, but looking at the folders from the linux fileystem, their

Hi, I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now having a problem with Originate and chan_local. I'm using the following Manager API action to originate a call: Action: originate Priority: 1 Context: trunk Callerid: 100 Channel: Local/100 at callback/n Exten: 123456789 Variable: USERFIELD=127.0.0.1|USEREXT=123456789 WaitTime: 30 This is intended to first call

Back to the RE-provision workbench. :-)

hi! anyone has a idea how to send caller id (incoming number) to shell script like: exten => s,1,wait,2 exten => s,2,System(/bin/script callerid-number) exten =>s,3,answer i need to run script in shell with command line like "/bin/script -a 123456789 " this 123456789 must be callerid (incoming call) number. tnx, Thomas

Hello, I have an internal extension, e.g. 1005 which is being called from an external/public number like 123456789. Now when it comes to the spoken voicemail information it says something like "number 1000 not available", however it should say "number 123456789 not available". How can I configure this? I already googled and I guess this is really easy, but I just couldn't

Hi, Try this: lines1<- readLines(textConnection("gene1 or1|1234 or3|56 or4|793 gene4 or2|347 gene5 or3|23 or7|123456789")) lines2<-readLines(textConnection(">or1|1234 ATCGGATTCAGG >or2|347 GAACCTATCGGGGGGGGAATTTATATATTTTA >or3|56 ATCGGAGATATAACCAATC >or3|23 AAAATTAACAAGAGAATAGACAAAAAAA >or4|793 ATCTCTCTCCTCTCTCTCTAAAAA >or7|123456789

Hi, I've migrated from an old samba installation (Samba as PDC) that used TDB backend for password. I've setup a box with ubuntu and samba 3 + ldap and I imported the old users. Old users works fine. I have problems with new users and machines. Old users works but they don't show up with smbldap-usershow command and I've problem in changing their passwords. If I check the ldap

Not sure what's up with the mailing list here. For some reason mails are not coming through. Try again... I am trying to link an asterisk box to my provider's asterisk server via SIP. (I know I could use IAX, but the provider does not allow that, so I can't). When an inbound call happens I get this: Jan 31 13:09:14 NOTICE[3716] chan_sip.c: Failed to authenticate user

Hi, I've been getting complaints lately that callers to my IVR are pressing a digit once but the system is responding as if they pressed it twice (once for each of two consecutive menus). I'm using an AGI script and logging all DTMF entries - and to the script, at least, it looks like the digit is being pressed twice. The TN being called is a VOIP number (provided by Flowroute) and being

I am using /set.seed()/ before the /sample/ function. How does the length of the argument of /set.seed()/ and order of the digits affect how the sampling is carried out? Specifically, I have used set.seed(123456789). Will this configuration give me a genuinely random sampling?? Thank you in anticipation. Penny. [[alternative HTML version deleted]]

Hello all, I would like to use Asterisk to add/modify SIP headers in the INVITE message, to include Privacy information, if the INVITE includes a *67 prefix (or another predefined prefix). That's an example of the INVITE I get: /INVITE sip:*6700112233445 at 192.168.1.100 SIP/2.0 From: "123456789"<sip:*123456789*@192.168.1.100>;tag=333333333 To: <sip:*6700112233445 at

Asterisk 13.8 Is CALLERID(all) supposed to wok for pjsip? When I do this: exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) same => n,Dial(PJSIP/phone123, 30) I expect the callerid to be as set, but is always seems to be "phone123", the name of the endpoint. Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I want to enable call forwarding for asterisk 1.6.0.6 I couldnt seen any config or option on gui or extensions.conf about it. I found some dialing plans to enable it on web as follows: [apps] ; Unconditional Call Forward exten => _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4}) exten => _*21*X.,2,Hangup exten => #21#,1,DBdel(CFIM/${CALLERIDNUM}) exten => #21#,2,Hangup ;