similar to: Global variables in global variables

Le 25/01/2023 à 17:56, Antony Stone a écrit : > On Wednesday 25 January 2023 at 16:46:14, Daniel wrote: > > On Sunday 01 January 2023 at 17:30:03, Antony Stone wrote: > >>> The [globals] section of that dialplan includes: >>> >>> Kphones=SIP/KC470IP&SIP/KSnom870 >>> Sphones=SIP/SYealinkT38G&SIP/SGC610IP >>>

Hi. I have a very old dialplan (ie: a dialplan for a very old version of Asterisk) which I've just transferred to Asterisk 16.28.0 The [globals] section of that dialplan includes: Kphones=SIP/KC470IP&SIP/KSnom870 Sphones=SIP/SYealinkT38G&SIP/SGC610IP Allphones=${Kphones}&${Sphones} In the old system, this results in ${Allphones} containing:

On Wednesday 25 January 2023 at 19:17:04, Daniel wrote: > Asterisk 20.1.0 > > [globals] > Sphones=SIP/SYealinkT38G&SIP/SGC610IP > Kphones=SIP/KC470IP&SIP/KSnom870 > Allphones=${Sphones}&${Kphones} > > -s*CLI> dialplan show globals > Allphones=SIP/KC470IP&SIP/KSnom870&SIP/SYealinkT38G&SIP/SGC610IP >

Hello, using 1.8.4. using a very simple local AGI script in bash which has only one line in it: echo -e 'STREAM FILE welcome 123 \n' dialplan: exten => 5150,1,Answer() same => n,Set(CHANNEL(language)=en_AU) same => n,AGI(testagi.sh) same => n,Hangup console output: -- Executing [5150 at AllPhones:1] Answer("SIP/PBX-00000024", "") in new stack

Hello - I have a TDM400P with 2 FXO and 2 FXS modules. Feeding the FXS modules are two VOIP lines which are terminated by VOIP adapters and have regular RJ11 wires connecting to the FXS ports. Since the two different VOIP lines have different phone numbers, and I know and can tell asterisk which VOIP line is connected to which FXS port, can I cause a distinctive ring on the extensions if a call

Sipura has implemented auto-answer in version 0.9.5 of the SPA-841 firmware. However, it is implemented via the Call-Info header, which Asterisk stable doesn't currently support. The attached patch implments a quick hack to support the Call-Info header from the Dial() application by way of setting the CALL_INFO variable. For example, the following macro can be used to dial up a single

Hi all, I've just upgraded to latest 1.6.0 SVN from a few days ago and my Gosubs have stopped working. This is from the verbose logs: -- Executing [03333407271 at incoming-aaisp:4] GotoIf("IAX2/aaisp-3802", "1?5:7") in new stack -- Goto (incoming-aaisp,03333407271,5) -- Executing [03333407271 at incoming-aaisp:5] Gosub("IAX2/aaisp-3802",

Is there a way to detect if a call is a transfer in the dialplan? Here is my issue: I have an office with 2 extensions. Under normal circumstances any call that comes in should ring both extensions. I accomplish this through a queue. The problem is that if the call is answered on say extension 11 and the answerer wants to transfer the call to the other phone, extension 10, transferring

Hi, I have got my Asterisk server running with TDM400 card (2xFXO & 2xFXS). I originally had the system configured with a Panasonic fax machine on one of the extensions. Due to the high volume of fax spam, I figured it would be a much better idea to capture the faxes as TIF or PDF files to minimize wasted paper, etc. I have downloaded, compiled and installed spandsp and can see the rxfax

hello, I am running asterisk on one linux PC and want to talk through this server using Kphone installed on 2 different PC's. These are the extra lines added to sip.conf and extensions.conf respectively. sip.conf [jitha] type=friend host=dynamic secret=jitha context=sip dtmfmode=inband [sudhananda] type=friend host=dynamic secret=sudhananda context=sip extensions.conf [sip]

Could anyone help me set up Asterisk in such a way that it makes SIP-SIP transfers using the REFER / NOTIFY method according to RFC-3515 ? SCANARIO: - Asterisk registers with PSTN<->SIP VoIP provider "V" (Vonage) as a friend - Asterisk is located in Europe, Vonage in located US. - Asterisk acts as an autoattendant located in Europe. - Asterisk answers and incoming call from

Does anyone currently use Vonage with Asterisk? I've tried to set it up but it looks like Asterisk (at least the version that I have) does not handle well the SIP call dialog, sending a BYE with the wrong tag. As a result, when I hang up, Vonage sends back a 400 Bad Request and the call on the PSTN side does not hang up. I know that Vonage does a lot of nasty stuff which impacts UA's

Hello everyone. I've seen postings for connecting asterisk to vonage but I'm still having trouble achieving that. I have a vonage softphone and I'm trying to register to vonage using asterisk. I have not had any luck. I am behind a firewall. I've successfully gotten xlite to connect and work from the same network. When I change the port setting in [general] to 5061, I am able to

Hi Nitesh, check out my config that I have for the Faktortel config in the asterisk@home sourceforge forum, you'll probably be able to work out how to set it up from there. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Wednesday, February 23, 2005 4:12 PM To:

Hi, I'm trying to get Asterisk to connect to Vonage (softphone acct) to allow me to place and receive calls. I have successfully configured Asterisk to route inbound calls and send them to the correct extension, but I can't get outbound calls to work. I have Asterisk successfully registering with Vonage, but when an INVITE is sent out, I get a "404 Not Found" back from Vonage

My previous post hasn't even made it to the list yet (am I being moderated?), but I got Vonage's Softphone service working for inbound and outbound calls. Keep in mind that there's currently no perceived limit on simultaneous inbound calls, which makes this a wonderful solution for Asterisk (at least for my use). Below is a sanitized snippet from my working sip.conf; your mileage may

I've yet to successfully register and receive calls from my Vonage softphone. I've tried what few examples are given in former post this list and some other forums, but nobody seems to be stepping forward saying it works recently. Either they broke something to where you simply cannot use *, or the config examples need updated. If anyone can show a still working config,

Hi All, I have installed Asterisk and able to create Users and get them connected to Asterisk after authentication. My question is how can I make calls to different DIAX clients through my Asterisk server. I also have vonage softphone account, using that I tried calling 18882255322 -- Registered 'manoj' (AUTHENTICATED) at 59.93.73.0:4569 -- Registered 'diax'

Hi All, I tried with different configurations and referred many articles to configure Asterisk with a Vonage account I have but all my attempts failed. I am a newbie and hope this mailing list will help fixing my problem and configure Asterisk. The error I get after I make a call to outside number like 18007633555 is -- Accepting AUTHENTICATED call from 59.93.69.218, requested format =

I know Vonage doesn't officially have a "bring your own device" type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060329/5bc9f644/attachment.htm