similar to: Outgoing Calls Only -- Firewall Rules

I'm trying to figure out how to listen in to a channel that I specify. I have the impression I've seen this done via Flash web controls, but I'm trying to write something myself and I can't figure out what command would be used. ChanSpy looks great, but I don't see how to specify the channel. I have a channel identifier like "SIP/provider-08748db0" which is what I

hello, please any IAX2 ActiveX control that wrap libiax2 or libiaxclient? i want to develope my softphone in delphi thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4345 (20090818) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

Hi All, I am relatively new to Asterisk. I have CDR enabled and successfully writing to MS SQL server. In my cdr table I am setting the userfield value with a line in my dialplan. If a call is placed to an invalid number (e.g. 12125551212), I see a cdr record created, however, my userfield value never gets set since the call never made it into the context of my dialplan. I am using AMI with the

Hello, I'm running several asterisks in a carrier environment. The asterisks do mainly gateway business between E1 cards and IAX with some routing logic. On one key server I see issues of "Auto-congesting call due to slow response" coming every number of calls. The IAX peer is in the same subnet, the servers are not really loaded. Versions in use are 1.2.2 and 1.4.23-rc3, with rsa

Hello, I have a time-series of standards measured for Refractive index. They are daily standards, however, I didn't run one everyday so some days have no data. I can plot the values, but the x-axis does not represent the correct time series (i.e. it's just an evenly spaced 1,2,3 type axis). I want to plot the points with some form of representitive date line on the x-axis. I don't

Has anyone else had issues with Originate returning the wrong error code? According to the docs, the following errors are supposed to be returned: 0 = no such extension or number 1 = no answer 4 = answered 8 = congested or not available Now in Asterisk 1.4.23 I get some error code 5's but since they're so few I tend not to worry. But what is concerning is the number of Error 0's I

On Aug 7, 2007 'Mojo' wrote: Nicholas Blasgen wrote: > I've got 4 SIP phone lines with a call-limit of 2 for each. I've > written a handy macro to allow my users to dial a phone number and the > macro will figure out the next available line to use by first checking > if the GROUP() is over 2 and then checking to see if ChanIsAvail() as a > backup, and if it

I've set up plenty of Asterisk boxes but never one that had to deal with a proxy server to be able to use a line. Using "X-Lite" I have no issue with settings as follows: Display Name: Any Name User name: 00575000010XXXX Password: 00575000010XXXX Authorization user name: <blank> Domain: directnationalloan.com Checked "Register with domain" and "Send outbound

I'm running Asterisk without FreePBX or any of the other managers. I'm trying to figure out how to set the maximum number of channels allowed on a single line? I'd just rather not have Asterisk try the line when I know I'll recieve a CONGESTION back from the SIP phone provider (ViaTalk in this case). Is there a configuration option I can't find that sets the maximum number

I've got a very nice PHP AGI script but I want to be able to do some database cleanup when the user hangs up the phone. I wish everyone would hang up when they were suposed to, but some people don't. So what does Asterisk send to an AGI file when the line has been disconnected? If I remember reading somewhere correctly, I don't need to use DeadAGI. Instead I'm able to use

Hello, I am a student considering participating in Google Summer of Code program. I came across Stephen Hemminger's idea of implementing a RSTP posted on http://developer.osdl.org/dev/soc/ and found it interesting. In attachment there's my proposal, which I just submitted to google. Please share your comments. -- tadeusz andrzej kad?ubowski -------------- next part --------------

Greetings All- I'm about to embark on some remote management testing and need a way to login to a remote system running CentOS 4.x/5.x via SSH, su to root (using a password), then execute a command. I currently login to the boxes using key based SSH like this: ssh -i ~/remote_key admin@$REMOTEIP Then, I SU to root. However, if I try to do this automatically like this: ssh -i ~/remote_key

Hi, yess its running on my local ip address. I've loged in to the router, andf orwarded port 8000. On 12/1/2015 2:17 AM, Marius Flage wrote: > Hi! > > Are you even sure that your icecast server is exposed on the internet? > Is your icecast server running on a public or private ip address? If > it's running on a private ip address (192.168.0.0/16, 10.0.0.0/8 or >

Hi all, I am testing different setups for Samba home share mounts via the CIFS protocol on Linux clients with and without Keberos security (both krb5 and krb5i). I am experiencing some strange behaviour in case of Kerberos authentication: In case of mounts (by root or the user itself) without Kerberos security (only NTLMv2 authentication), local root and the owning user on the Linux client

I've got 4 SIP phone lines with a call-limit of 2 for each. I've written a handy macro to allow my users to dial a phone number and the macro will figure out the next available line to use by first checking if the GROUP() is over 2 and then checking to see if ChanIsAvail() as a backup, and if it can't use the line for either reason it goes to the next line. The problem is that there

I have 10 SIP trunks that I'd really like to round-robin load balance. Currently I have a macro that switches between available lines, but there really must be a function in Asterisk to do this on its own. So my question is just that, are there any easy ways for Asterisk to either balance between SIP trunks or even just a built in function to find the next available SIP trunk. I think using

I've heard about this, but I really can't seem to find anything on it. I've got a strange setup that exists only because of firewall issues, and everything about it seems fine. The setup: SIP clients -> Asterisk (office) -> IAX -> Asterisk (colocation) -> SIP PSTN Termination All the extensions I want to be able to dial are on the colocation box. What I'd really

Hi there, I have set up a server to act as a PDC, file server and print server for a record store I occasionally work at. Everything is working fine, except browsing the printer share is painfully slow. It's not really a problem, but if something is wrong I'd like to know what it is. smb.conf contains ---------------------------------------- [global] ... printing = cups printcap =

Hi, today i wrote a function in R of the type: index.refraction <- function(Temp,Press, RH, CO2) When i try to plug a negative number in Temp, i got this type of error: " n <- index.refraction(Temp= -40,100,80,CO2) Messages d'avis : 1: In Ops.ordered(left, right) : '-' is not meaningful for ordered factors 2: In Ops.factor(left, right) : - not meaningful for factors 3:

Hello, i wrote a simple but useful (i hope) bash script featuring a easly and fastly configuration method for implementing a Linux Gateway feauturing Incoming/Outgoing MultiHomed/LoadBalanced Subsystem Management. I want share this little script. you can download a copy from the following address: http://www.initzero.it/products/opensource/izbalancing/download/izbalancing just save it in your