similar to: allpage issu on asterisk 1.8.3.x

Hello Everyone, I've been playing with an agi script for paging sip phones. page.agi will take all available sip extensions and assign them to the global variable PAGE_GROUP. Allowing the phones to be paged from the dialplan with the new Page cmd. Extensions to be excluded are presented as arguments to the agi. Each time a page is made this agi refreshes the global variable. This works with

Hello, I have a question concerning llvm-c. I have set up a function that needs to invoke an external method, in a other library. It has the following signature: void* NSFullUserName(void); The void* can be replaced with a i8*, that far I was able to get, but when my call is invoked, the engine gives me the following message: LLVM ERROR: Tried to execute an unknown external function: i8* ()*

Is there a means to split a vector into its individual elements without going the brute-force route for arguments to a predefined function call? offred.rgb <- c(1, 0, 0) * 0.60; ## Brute force style offred.col <- rgb(offred.rgb[1], offred.rgb[2], offred.rgb[3], names = "offred") ## Desired style

Hello, I'm testing with asterisk 1.8.3.2 and come across this: Call from one extension to another with: [macro-internal-call] ;ARG1=extension to call exten => s,1,Set(TOCALL=${DB(SIP/${ARG1})}) exten => s,2,Dial(SIP/${TOCALL},60,tT) ... As I had no entry in the asteriskdb, so the SIP uri was empty, and asterisk core dumped with: gdb output: #0 0xb7c7db33 in strchr () from

On Aug 12, 2010, at 10:43 AM, F van der Meeren wrote: > Where am I going wrong here? > Did you link against the library that contains the function? -eric

Hey Guys, We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't working We have allpage.agi script for paging system on all polycom 501 but after upgrade it broke. Any idea what is this error ? extension.conf exten => 7770,1,agi(allpage.agi) exten => 7770,2,meetme(7770,dq) exten => 7770,3,playback(beep) exten => 7770,4,hangup following is agi debug....

Hey Guys, I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi stopped working look like asterisk 1.8 did some changes in manager apps i am doing following.. my phone is ringing but not auto answer could you give me some issue what i am doing wrong ? root at ubuntu-test:~# telnet 127.0.0.1 5038 Trying 127.0.0.1... Connected to 127.0.0.1. Escape character is '^]'. Asterisk

I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems

First off, A lot of thanks to this list. I have learned ton from reading through the posts this past year. I need some advise. I have two group of phones connected to a single server. Group1= SIP/2503&SIP/2504 Group2=SIP/3501&SIP/3502 I'd like to be able to dial an extension and page a certain group of phones only if ChanIsAvail returns 1. I am not sure how to go about

I have just finished implementing an Asterisk system for my place of business (first one), and after three days of flawless usage, Asterisk seems to have crashed. I wasn't running with '-g', so I don't have a core dump. Here's the sequence of events leading up to the crash: 1. call comes in on our TDM2400P 2. all of our phones (about 26 Polycoms) ring. (it's after

Hi All, I am trying to compile R from source on a 64 bit Windows. I have downloaded and installed all the third party software as per the R - documentation. The compilation starts fine and after a while it stops with the following error message: D:/Rtools/mingw_64/bin/../lib/gcc/x86_64-w64-mingw32/4.9.3/../../../../x86_64-w64-mingw32/bin/ld.exe: cannot find -lRgraphapp collect2.exe: error: ld

I am having what I think is a strange problem with applying TukeyHSD to an aov fit with error strata. TukeyHSD is supposed to take "A fitted model object, usually an 'aov' fit." aov (with error strata) is supposed to generate an object of type aovlist, which is a list of objects of type aov. But I can't seem to feed components of my aovlist to TukeyHSD. I guess I

Abwesenheitsnotiz: [Asterisk-Users] oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6Hi All I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2 gb ram, with g729 for i686 , (fedora 1). my problem is sip - oh323 - h323 (quintum) - pstn , sip party can listen otherparty realtime voice , but other party geting sip party's voice 1 sec later (not

I''ve attached the patch I had to make to get open3 to work under 1.8.3. It doesn''t take backwards compatibility in to account, so that''ll probably have to be addressed. Thanks, -- Nathaniel Talbott <:((>< -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: win32-open3-0.2.0.txt Url:

I''ve seen some reports of problems with running Rails (version 1.0 here) on Ruby 1.8.3. Is this true across the board or just Mac OS? I''m setting up a Linux server and want to know if 1.8.3 is ok on that. Thanks, Jamie

I am running 1.8.3 and my BLF lights have stopped working. The hints appear to be intact when I use core show hints. But none of the phones are getting the BLF updates. This has happend in the past and I have had to restart my server. What could be causing this to occur. It did not do this with the 1.6.x builds. Is there a way to reload the hints or force a refresh without re-starting

Dears; Actually, the needed file name to be ooh323.conf and not chan_ooh323.conf, so I copied the file from chan_ooh323.conf to a new file name ooh323.conf and it is working fine.

In console.conf.sample it says run the command "console list available" CLI command. It does not seem to be present: console list available No such command 'console list available' These are the only console commands I see: console answer Answer an incoming console call console autoanswer Sets/displays autoanswer console dial

Hi all, I've got something strange, that got me searching for quite awhile. Configuration as followed: Linphone on a laptop, that is connected via openvpn to a proxy. That proxy is connected with iax to another asterisk. On the second one i have several hard and softphones. Behaviour at first glance: >From the softphone i can allways set up a connection, But the otherway round fails 9

When did Dial() with a custom ring cadence replace the default from indications.conf for subsequent calls? indications.conf: ringcadence = 2000,4000 asterisk -rx "dahdi show cadences" r1: 667,1333 extensions.conf: exten=> 201, 1, Dial(DAHDI/1) exten=> 202, 1, Dial(DAHDI/1r1) exten=> 203, 1, Dial(DAHDI/1r0) Dialing, in sequence: 201 -> 2000,4000 ; Good 202