similar to: Forcing Dial application to skip if called server is unreachable

Hi, I write the CDR of my Asterisk 1.2.17 server in MySQL database using cdr_addon_mysql.so. Now I'm trying to write some custom values to userfield column by the SET(CDR(USERFILED)=SOME_TEXT) sintax, but nothing gets writeen in MySQL cdr table!! Why? I'm I skeeping something or what? Taking a look at the URL:

Hi, Can Asterisk write to multiple MySQL databases in different machines, at the same time, as a backup scheme? If it does, where can that be configured? In res_mysql.conf file? Does anyone ever made it? Regards, Ricardo. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I am trying to do : Exten =>_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:10000)) But it return that error: [Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid timeout specified: 'L(10208400:61000:10000)' Why? I forgot something ? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que

Hi, I use Asterisk 1.2.17 in my site, and now I'm trying to configure ENUM lookup in my server. When someone calls a number that has multiple ENUM entries, randomly Asterisk seems to fail to return a correct answer, and dial by ENUM fails. I've goggled a bit on this, but didn't get any good conclusion. There is some RFC Compliant ENUM Macro that can be used that is announced

How do I manually set up a ring group? All the info I've Googled tells me how to do this using Trixbox or FreePBX. I am using standard Asterisk 1.4 configuring at the CLI. Michael

Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem

Dear all, I'm trying to enable Asterisk to work with FAX using T38. I've tried Asterisk 1.2.4 with the available patch found at URL http://bugs.digium.com/view.php?id=5090 and also with the new 1.4 Beta3 that is announced to support it too. With both Asterisk versions, I've sent with success FAXes between two FAX machines each one attached to an ATA interface, both registered in

Hi all, Does ENUMLOOKUP can query multiple DNS servers without having to replicate the same code in which the only thing replaced is the server? If I use ENUMLOOKUP(${exten}), Asterisk will parse enum.conf file to find the list of DNS servers in order of preference to be queried, but, I pretend to use something like this: ${ENUMLOOKUP(+${ARG1:2:},sip,c) which does not seam to care about

Hi I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by outbound SIP. Now i want to detect DTMF-Tone Code coming from the called party to trigger a signal. Can this be done with asterisk? I read that the codec with DTMF detection are ulaw and alaw. But i couldn't find a command to detect dtmf's within a normal call. thanks and mani greetings Christian

Hi all, I have a question about finalizers... I have a package that manages state for a few connections, and I'd like to ensure that these connections are 'cleanly' closed upon either (i) R quitting or (ii) an unloading of the package. So, in a pared-down example package with a single R file, it looks something like: ##### BEGIN PACKAGE CODE ##### .CONNS <- new.env(parent =

Hi, I currently have deadtime = 15 in my smb.conf. This featured worked good to disconnect clients that have been idle for > 15 minutes in the past. With the 3.0.13 version, it seems to do nothing. As in, it just keeps the user connected indefinitely. Here is what I see when no one is currently at the office and everyone is logged off their computer. Thanks [root@spicy p]#

Hi everybody, Firstly I'm going to describe the scenario where I'm working. I use a E400P with Asterisk CVS-05/22/03-11:14:50, and I'm working with asterisk trow AGI scripts (Perl). The configuration of extension.conf is: exten =>_s,1,Answer exten =>_s,2,AGI,script.agi Inside the AGI script is call Dial application as follows: print "EXEC Dial

Is it possible to load phone registration information stored in sipfriends MySQL DB, so that Asterisk "thinks" those phones are already registered? This would be very usefull for a redundant server... Regards, Ricardo Carvalho. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have done a cluster analysis doing: 1-clusNorth <-hclust(dist(Artorious)^2, method="ward") 2-clusNorth$labels <-Artorious$Name ## to show the case names and not numbers 3-dend1 <- as.dendrogram(clusNorth) 4-plot(dend1) My Dendrogram is now showing the names of my cases in the dataframe on the x axis 1 OMNICELL INC COM 2 GETTY IMAGES INC

Package: xen-hypervisor-4.0-amd64 Version: 4.0.1-5.4 Severity: important When using the the intel igb driver from the 3.2.0-0.bpo.4-amd64 kernel and debian squeeze hypervisor with a Intel 82576 quad port nic the first nic fails to get an IRQ mapping: relevant lines from dmesg ----8<---- [ 24.264857] Intel(R) Gigabit Ethernet Network Driver - version 3.2.10-k [ 24.264929] Copyright (c)

Hi, I'd like to have a SIP phone at home and at the office and have them both ring when my extension is dialed. Right now I used the same config for the phones (Cisco 7960's). So they both register with the same login & pw. This doesn't seem to work quiet right, where only the last phone to register seems to get the calls. What is the proper way to set this up? Thanks, Justin

Is there a way to differentiate between a SIP address which hasn't registered (but is within sip.conf) and one that's not there at all (i.e. not in sip.conf) using a straight dialplan. I'd like to differentiate actions depending the state of a SIP device and whether it's in my config or not (if that makes sense, basic automap of dial-in lines to sip phones, but if they've

Thanks a lot for the help. But if Asterisk has private IP address and the only way to access it from remote sites is to have vpn connection to the site that asterisk existed (the site has vpn), then how that will happen from the Mobile to be able to run the softphone from the mobile? Any help? Regards Bilal ----------------- I installed out of curiosity today, and guess what? You can do SIP over

hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial application intructs to

Hi, How can I configure my asterisk server so that I can receive incomming calls comming from the same IP from where my server also receives phone registrations? The problem is that since the moment any extension registers at that IP (actually I have a registration proxy running at that IP), asterisk no more accepts calls coming from a SIP trunk I also have at that IP, replying back with