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Is it possible to force the Dial function to skip to the next priority if it doesn't find the server of the called contact within a few seconds? I know I can use: Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]) where I can use some short timeout in the "timeout" option, but if I do so, when some call is well succeeded, it will only ring for that time! Any ideas? Regards, Ricardo. -------------- next part -------------- An HTML attachment was scrubbed... UR...

...se 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 = emptyenv()) .CONNS$resource1 <- NULL .CONNS$resource2 <- NULL ## some more .CONNS resources... reg.finalizer(.CONNS, function(x) sapply(names(x), disconnect), onexit = TRUE) connect <- function(x) { ## here lies code to connect and update .CONNS[[x]] } disconnect <- function(x) { print(sprintf("disconnect(%s)", x)) ## here l...

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

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 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

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

...---- 1 root root 4096 Nov 13 17:37 rescan --w------- 1 root root 4096 Nov 13 17:37 reset -r--r--r-- 1 root root 4096 Oct 16 15:25 resource -rw------- 1 root root 131072 Nov 13 17:37 resource0 -rw------- 1 root root 4194304 Nov 13 17:37 resource1 -rw------- 1 root root 32 Nov 13 17:37 resource2 -rw------- 1 root root 16384 Nov 13 17:37 resource3 -rw------- 1 root root 4194304 Nov 13 17:37 rom lrwxrwxrwx 1 root root 0 Nov 13 17:37 subsystem -> ../../../../../../bus/pci -r--r--r-- 1 root root 4096 Oct 16 15:25 subsystem_device -r--r--r-- 1 root root 4096 Oct 16 15:25 subsys...

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

Hello I have created an iax exten in my iax.conf file: [300] type=friend username=300 secret=*** context=default host=dynamic callerid="some name" <300> auth=md5 Then in my extensions.conf I have: exten => 300,1,Dial(IAX/${EXTEN},20) exten => 300,2,Hangup I can dial from iaxComm (a soft IAX client) and that works fine. But when I try to dial 300 get: WARNING[22077]:

On Fri, Apr 24, 2020 at 4:35 PM Peter Crowther <peter.crowther@melandra.com> wrote: > > On Fri, 24 Apr 2020 at 21:10, Mauricio Tavares <raubvogel@gmail.com> wrote: >> >> Let's say I have libvirt >> >> [root@vmhost2 ~]# virsh version >> [...] >> >> Running hypervisor: QEMU 2.12.0 >> [root@vmhost2 ~]# >> [...] > > When

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

Greetings: I did some quick searching of my history of this list, and I tried a quick Google search as well, but perhaps someone on the list can quickly answer this question. I have a very nicely working Asterisk system at home with two Digium X100P FXO cards. When my SIP phones want to dial-out I have them setup to grab the first analog card (Zap/1) with the following extensions.conf segment: