similar to: dummy Technology/resource for Dial

Does anyone have a better ENUM lookup handler than the built-in ENUMLOOKUP() function? The built-in function does not properly handle multiple return values such as: 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 "u" "E2U+SIP" "!^\\+1866(.*)$!sip:1866\\1 at tollfree.sip-happens.com!" . 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 "u"

> -----Original Message----- > From: Brian J. Murrell [mailto:brian@interlinx.bc.ca] > > - Single Sign-On via Kerberos > > OK. Actually I understood this feature. I am just wondering how it > applies in an MS network. SSO to all of what? If my DCs are my > file/printer server(s) (let's say I mirror the data contents > of my PDC to > my BDC as well --

Is there any way (short of not patching it into the kernel) to turn KDB off on a given machine? There are some machines I do not want dropping into kdb when/if it oopses. Thanx, b.

Slightly OT, but I'm wondering if anyone here has come across a "soft ATA". That is, software that will perform the functions of a basic POTS line ATA on Linux with a zaptel driven card. I have a Linux machine with a zaptel card in it and I want to have another Linux machine running Asterisk utilize the zaptel card in the first Linux machine to make outgoing and receive

Greetings all. I have some controller code that uses win32ole (only available on windows). This code is now solid, and I''d now like to resume development on (any) other OS(grin). But alas, the controller bails because the OS specific library can''t be found. Can I conditionally specify action code compilation (and a require ''win32ole'') based on OS or

Greetings railsers - I''m trying to add to a collection through HABTM, but the sql insert is trying to insert a PK rather than letting mysql produce the auto_increment''ed PK. ## @medication_dose holds a validated, saved model @medication_dose.medication_frequencies << MedicationFrequency.find (:all) The above bails with, Mysql::Error: #23000Duplicate

https://bugs.freedesktop.org/show_bug.cgi?id=47306 Bug #: 47306 Summary: segfault in nouveau_fence_update Classification: Unclassified Product: xorg Version: unspecified Platform: x86 (IA32) OS/Version: Linux (All) Status: NEW Severity: blocker Priority: medium Component: Driver/nouveau

Greetings again, all! Just in case it's useful, here is my smb.conf file (it's very simple right now, as it is only for testing pusposes. Once I get it working right, then it'll be filled in!) <-----START-----> [global] workgroup = HOME encrypt passwords = yes netbios name = TAMIRA [homes] comment = Home Directory path = /home/

I think I understand what Active Directory is all about. I understand LDAP and I understand Kerberos. I can see how AD (well, Kerberos actually) enables single-sign-on (I assume it deals in tickets with the Windows clients as standard Kerberos clients do) and can make life easy in a large network (which, IIRC was one of the design goals of Kerberos in the first place). But lets say I have a

Hi All After a Dial() I get: WARNING[7964][C-000075a8]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) if the subscriber is not registered. Is there a way from dialplan to know, *before* Dial(), if a destination Subscriber is a) not registered or b) busy ? I need to redirect a call to some other Subscriber if (s)he is not there

if i have this setup: [sip users] -- [asterisk] --- [as5300] --- [pstn] asterisk will talk to as5300 using sip. i will use as5300 as a trunk on the asterisk so sip users can call out to pstn. what i would like to is do prepaid on those trunks, not on the sip users. sip users can call any other sip users . i want to do it that way coz i'm trying to build a multi-tenant pbx, and i will use

I want to use Asterisk as a general message delivery system here. That is, I want to be able to have a (shell, perl, etc.) script on my Asterisk server dial an extension, wait for it to be answered and then play a sound file and then hang up, or even wait for a response or reactions to some IVR. Certainly if I had a SIP library, I could have the script simply look like a SIP extension but that

I seem to be running into a problem where I am trying to rsync from a source directory that lacks write permissions (i.e. r-xr-xr-x). Presumably this is because rsync creates the directory on the destination, then sets the permissions to match the source and then tries to sync the contents of the directory, which it cannot of course lacking write permission in the directory. Is there a way to

Hi all I'm looking for a solution to this problem. *box--------internet-----------nat-----------softphone We have potential customers who will be travelling the world with laptops/pda's. They need to be able to connect to the asterisk box via ip wherever they are and will have no control over nat whatsoever. I have read that STUN offers this service, but cannot picture in my mind how

Are there any configuration options to allow certain sip/iax accounts to dial out over specific trunks, and also to stop them dialing out over other trunks. Thanks in advance Bails

Morning list, Was curious if it is possible to decrease the time asterisk takes to answer an incoming call to a zaptel interface. Example: [Jun 11 09:33:06] VERBOSE[4489] logger.c: -- Starting simple switch on 'Zap/2-1' [Jun 11 09:33:10] NOTICE[4489] chan_zap.c: Got event 18 (Ring Begin)... [Jun 11 09:33:12] NOTICE[4489] chan_zap.c: Got event 2 (Ring/Answered)... [Jun 11 09:33:12]

Is there any less cumbersome way of doing conditionalized/branching in extensions.conf other than something like: exten => s,n,GotoIf($["${SIP}" = "PJSIP" ]?pjsip) exten => s,n,Dial(${ARG2},20,TtWw) exten => s,n,Goto(afterdial) exten => s,n(pjsip),Dial(${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,"PJSIP/","")})},20,TtWw) exten =>

So, I have been testing ekiga 3.0 with Asterisk, and sadly, it don't work. I am told by the ekiga devs in http://bugzilla.gnome.org/show_bug.cgi?id=553595 and http://bugzilla.gnome.org/show_bug.cgi?id=553810 that the problem is that Asterisk does not support SIP forking. The issue is that I have multiple addresses on my workstation: 2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500

Hi everybody, after upgrading a few CentOS machines from 5.0 to 5.2, and during that upgrading rsync from 3.0.0-1.el5.rf to 3.0.3-1.el5.rf (both from rpmforge) rsync stopped working and bails out with a segfault after some undeterministic time (zero to a few seconds) during a sync. Manually going back to 3.0.2-1.el5.rf seems to work. Has anybody experienced similar problems? I'm not sure if

I am using asterisk 1.4.10 and seem to be having a problem with blind transfer. This could very well be a pebkac problem but I'm not sure. A call comes in on a Zap channel and answered just find by a context that does a Goto which calls a macro (seems convoluted now that I look at it) to do some CID bookkeeping but that ultimately dials all of the phones interested in calls from the Zap