similar to: Re: [Asterisk-Dev] ANNOUNCE: CLASS-like features for Asterisk

Yo, Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate call divert-feature. This one validates if the extension a call-forward is to be set to is actually valid for the current context and additionally saves this context into the DB and always uses it to originate the divert from, as you can't expect the

The following transfer led to a crash of asterisk, without leaving a core or any utterances in messages or debug file. It looks like the zombie which was created during the MASQ-transfer was not cleaned up... But why did it start a Dial??? And... why does Asterisk die when this happens?? Thanks!!! Michiel -- Zap/32-1 answered Zap/6-1 -- Stopped music on hold on Zap/6-1 -- Starting

Hei! I have a little problem with the subject. I use Asterisk CVS-HEAD-09/06/04-12:42:56 as a production *, but I do tests with a newer version Asterisk CVS-HEAD-11/18/04-10:01:32. Ok the problem is: in extension.conf I use macro for redirection, found on wiki pages: [macro-stdexten] ; ; Standard extension macro (with call forwarding): ; ${ARG1} - Extension(we could have used ${MACRO_EXTEN}

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out

hello my friend has the proxy set up his extention set up his password set up but he isn't getting a dial tone is there a second setting we need to put the address in? he is going to advenced settings line1 and in the proxy address box he is putting the info in below is the way he has it set up Sipura SPA Configuration Sipura Technology Inc Info System SIP Provisioning Regional Line 1 User 1

Dear All, I can not transfer call in progress. What's wrong with my macro? I think tT flags is enough right? [macro-stdexten] exten => s,1,Set(temp=${DB(CFU/${ARG1})}) ; Get CFU key exten => s,2,Set(DNDStatus=${DB(DND/${ARG1})}) ; Get DND key exten => s,3,GotoIf($["${temp}" = ""]?5) ; If not existing, goto priority 5 exten =>

Dear All, I have a problem with call transfer. When I dial a number and then I want to transfer current call to an extension, I'm getting disconnected. Transfering incoming call works fine. I'm using macro for dialing. extensions.conf: [from-internal] ignorepat => 9 exten => 200,1,Macro(stdexten,200,SIP/dzalewski) [macro-stdexten] exten => s,1,Set(temp=${DB(CFU/${ARG1})})

Hello all, I've got a T100P connected to an Adtran TA750 with a T1 crossover... This connects to a patch panel with phone ports. The Adtran is fully populated with FXS cards. All I get on any phone port is a fast clicking noise... No dialtone. Asterisk 'sees' the card, (the channels show up in /proc/zaptel). Incoming calls are routed to the zap/x channel, but no ring. I'm

Hi All, I'm upgrading some PBX's from 1.2 to 1.4 and having a bit of trouble with converting the Realtime application to the REALTIME function. I have the method down and understand simplistically what is going on, at least enough to get my old 1.2 apps to run in 1.4 functions. I do not understand why change from the app to the func? What the benefits? To me, the app seemed so

Ia had a server overload today because someone did a call forward to their own extension. To do a call forward I write a key called CFWD with the extensi?n number and number to dial . The main script tests if the key/value exists and dials the number stored in the database. What is an easy way to prevent dumb people from creating a loop? -- Telecomunicaciones Abiertas de M?xico S.A. de

Hi all, Has anyone already written something which allows saving and loading the internal DB settings? All users CFWD and speeldial settings are stored in the DB in my setup which makes it a pain to restart Asterisk.... Looking at showtree in db.c (why isn't that exposed in the CLI?) It shouldn't be too difficult, but I don't want to reinvent the wheel. On the same track, I am also

Hi all, I've got a provisioning file that I use to configure Cisco SPA112's. I'm wanting to get this file to do 3 things for me. I want to turn T.38 on, Call forwarding off, and Call waiting, off for both lines. but it's not working. This is what I'm using to enable T.38 for line 1. <FAX_Enable_T38_1_>Yes</FAX_Enable_T38_1_>

http://chan-sccp.berlios.de/ 20050713 ftp://ftp.berlios.de/pub/chan-sccp/chan_sccp-20050713.tar.bz2 I didn't have a spare 7960 to use this week, so maybe some line issue is still present. - fixed a memory leak on database updates (dnd, cfwd*) - fixed old memory leak on unload (now unload chan_sccp.so and load chan_sccp.so work. It does reload the config when asterisk is running) - socket

Hello, I'm configured Sipura-3000 to forward IP calls to PSTN number on no answer (In User1 tab Cfwd No Ans Dest: 123456@gw0) IPPhone ---IP---> Sipura-3000 ---PSTN---> PSTN User Generally it works fine, but Sipura sends back SIP OK to IPPhone just prior to dialing to PSTN number. How to configure Sipura to detect that the remote side on PSTN picks up the phone and only then to

> Ia had a server overload today because someone did a call forward > to their own extension. To do a call forward I write a key called CFWD > with the extensi?n number and number to dial . The main script tests if > the key/value exists and dials the number stored in the database. What > is an easy way to prevent dumb people from creating a loop? Right after you have

Could this possibly mean that any person who has CF set should never be available as CF Destination. Simple db entry/check can have this done. On Tue, Jun 2, 2015 at 5:34 PM, <dk at donkelly.biz> wrote: > > > > > *From:* asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] *On Behalf Of *Kevin Larsen > *Sent:* Tuesday, June

On Tuesday 02 Jun 2015, Carlos Chavez wrote: > Ia had a server overload today because someone did a call forward > to their own extension. To do a call forward I write a key called CFWD > with the extensi?n number and number to dial . The main script tests if > the key/value exists and dials the number stored in the database. What > is an easy way to prevent dumb people from

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of A J Stiles Sent: Wednesday, June 3, 2015 3:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Forward loop protection... On Tuesday 02 Jun 2015, Carlos Chavez wrote: > Ia had a server overload today

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin Larsen Sent: Tuesday, June 2, 2015 4:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Forward loop protection... > Ia had a server overload today because someone did a call forward > to their own extension. To do a

We have an Asterisk server acting as a hosted PBX system for many clients, and we're going through an upgrade to Asterisk 1.6 by moving our most important (and complicated) clients one at a time. But we're having a problem with one customer that I really can't explain. I can place calls directly to one phone at the customer's location (they also have an IVR that asks for an