similar to: Using Asterisk as sip user agent with more than one device

Hello, does anyone know how to get the say number (say.c) agi "application" to work in french [assuming that I have the French voice files] I have looked in the code and about a 1/3 of the way thru there is : /*--- ast_say_number_full: call language-specific functions */ /* Called from AGI */ int ast_say_number_full(struct ast_channel *chan, int num, char *ints, char *language, char

Hi, I'm using the macro below in extensions.conf for most of my outbound calls. One issue with my current configuration is that when I make an outbound call it doesn't properly detect that my PSTN line (Zap/1) is busy with another call and then overflow to my outbound IAX connections. I think the root cause is that DIALSTATUS gets reported as BUSY. The debug output is below. My desired

Firstly if this is not list specific then excuse me. I am looking for information on how I might use a Linux box with three network cards to service two subnets off of a single 10Mg VPN feed. We have currently got two 5 offices all connected over a VPN. 2 of the offices are to merge but for simplicity we don''t want to change all the IP addresses. We are moving the two networks

I can dial from 601 to a public number. The public number rings. I pickup and hear nothing, while on 601 it keeps ringing. (BTW, is it right to say "ringing" on the active phone?) The *CLI> doesn't show me anything useful: Executing Macro("SIP/601-8238", "dial-pstn|88097880|") in new stack Executing SetGlobalVar("SIP/601-8238",

On Fri, 26 Jun 2015, Ludovic Gasc wrote: > 1. What's the "official" notation of each line: "=>" or "=" ? In the > wiki of Asterisk, I see very often "=>", however, what's the reason for > both syntaxes authorized ? Historical ? I'm not 'official,' but I have a strong preference for just '=.' Using

> > Message: 24 > Date: Thu, 23 Oct 2008 03:53:55 -0400 > From: "John" <jses27 at gmail.com> > Subject: RE: [CentOS] OT: Setting a CentOS to gateway a private IP > address > To: "'CentOS mailing list'" <centos at centos.org> > Message-ID: <86C552A83B964E6499972CEAD4610189 at ethan27> > Content-Type: text/plain;

Hi William. Thanks a lot for your help. Im having some trouble recompiling my kernel after a installed the patch. Im running RH 7.3 with kernel 2.4.18-3. The patch I installed is routes-2.4.16-6.diff. I got no errors installing it. I added the multipath support, and recompiled it. The make dep and the make bzImage went fine. I got error during the make modules. These are the errors:

This is mildly off topic since it has more to do with TCP/IP networking than with CentOS per se. However, at the risk of ruffling a few feathers I would like some advice on how to resolve the following routing problem: Given: dual homed host running CentOS-5.2 with eth0 = 2xx.yyy.zzz.23 eth1 = 192.168.219.1 and a Cisco gateway at 2xx.1xx.y7y.1 with the following networks: interface

Hello, this is an example extensions.conf. [default] exten => 500,1,Answer exten => 8,1,SetGlobalVar(firstdigit=8) exten => 8,2,Goto(process,s,1) exten => 9,1,SetGlobalVar(firstdigit=9) exten => 9,2,Goto(process,s,1) I call extension 500 and send dtmf digit 9. This is printed to the CLI: -- Executing Answer("Zap/20-1", "") in new stack -- Accepting

Le 15/01/2020 à 19:50, C.Maj a écrit : > On 2020-01-15 11:24, Administrator wrote: > > 8<'s > >> One of the provider took a pcap and told us that expiration was set to 0 >> that's why they don't accept the registration. We took a pcap on our >> side when SIP packet goes out of our server and we see that the >> expiration parameter is setted to

Helo List, First I would like to apologize for my bad spelling as well as that I did not search the wiki first. I only have email access at the moment. I am having trouble setting both variables and global variables thru an extension. I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4 with an Xlite softphone. I have two xlite phones on diffent computers. One logs in as xlite1 and the other as

I got two phones on an ATA-186 (601, 602) and two phones on the TDM22B (603, 604). I have two lines on the TDM22B. I cannot figure out some of the problems: 1. 601 dials via ZAP/3-1 to local phone number at PSTN: ringing pickup on PSTN (empty) still ringing in the phone set 601 2. call from PSTN back: 601 picks up ... everything works !!! No caller id shows up 3. For testing I have only one

2015-06-26 17:11 GMT+02:00 Steve Edwards <asterisk.org at sedwards.com>: > On Fri, 26 Jun 2015, Ludovic Gasc wrote: > > 1. What's the "official" notation of each line: "=>" or "=" ? In the wiki >> of Asterisk, I see very often "=>", however, what's the reason for both >> syntaxes authorized ? Historical ? >>

Hi guys Ive been trying to get this to work for ages now, basicaly im trying to do if ${woteva} = "" (nothing), or its none existenant then do label 1, else label 2. for my last called function, so it will play a different message if theres no last call in the system or it was anonymous. ive tried exten => 1000,1,GotoIf($[${last-call${CALLERIDNUM}} = ""]?4:5) and heaps of

Hi there, I want to set up an extension set that acts different depending on the count of used lines. I have a EuroISDN E1 board with mISDN and I only want to offer 10 lines. Therefore I set up a global variables LINES in the general section of extensions.conf and instantiate it with 0. I a call is incoming I check the LINES variable wether is 10 or more. If so I make a call transfer. If not

Hi I am using Asterisk 1.2 and for the life of me, I am unable to transfer outbound calls (eg calls I initiate from sip extensions). When I press #, nothing happens. Inbound calls transfer fine, but only once per call. The problem happens: - With both software and hardware phones. - With calls going out through the ZAP channel and to internal SIP extensions. - After I have transferred an

Hi, I am trying to pass DIALSTATUS to a Macro so that i can set a variable when a call is placed (call is placed via a call file to another extension first). Basically i don't want to dial a number where a call is already bridged and thats why i am setting a variable. [macro-afterdial]; exten => s,1,Goto(s-${ARG1},1) exten => s-ANSWER,1,SetGlobalVar(NUM${ARG2} = "ACTIVE")

Hi. Can anyone help me with the following: [globals] OFFICEHOURS .................................... [internal] exten => *80,2,SetGlobalVar(OFFICEHOURS=100) exten => *80,2,SetGlobalVar(OFFICEHOURS=200) .................................... [incoming] exten => s,1,GotoIf($[${OFFICEHOURS} = 100}]?incoming-officehours:incoming-officehours-off 1. Am I using the right sytanx when

Hi I''m looking for a quick recipe for a newbie to control http traffic in my linux gw. My internet is overloaded already and vpn external clients are experiencing troubles (disconnecting in peak hours). Any suggestions ? Regards Guillermo Caracas/Venezuela On Thu, 2004-05-06 at 14:40, lartc-request@mailman.ds9a.nl wrote: > Send LARTC mailing list submissions to >

Hi Is it expected behaviour that a libvirt NAT-based network is faster than a libvirt routed network? I would guess that the routed network would be the fastest one in all cases, but I'm seeing the opposite in my setup. On a 100mbit internet connection, the routed network tops at 4mbyte/sec, while the nat-based network goes all the way to 10mbyte/sec. I've spend the last 6 hours on