search for: benchev

Hi List Is someone out there using one or more GSMgateway(s) from CyberTelecom ? Me and some friends are interested in buying some of them, but before we would like to ask, how the experiences are others have made. e.g. How easy to setup ? How reliable ? How's the voice quality ? etc. Any input/feedback is welcome. Greets Adibar --

...> > > > This script does what I want to do. But how do I set it up? > > > > http://www.junghanns.net/en/callback.html > > > > I see it uses PHP for scriptlanguage. So where do I place it (the .agi)? > > /var/lib/asterisk/agi-bin > and should be 755 > benchev > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > __________________________________...

I have an analog GSM Gateway that is connected to a normal SIP ATA device. Basically what it does is this : when you call the extension nr. of the SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia a Grandstream HT286. I would like to use the GSM Gateway to route my outbound cellular calls, how

Could anyone please help me with that: I have an analog telephone connected to Asterisk using a Sipura 2002 ATA. When calling the extension, the caller ID presented is always the number of that extension rather than the number of the calling one. While I learned that this is the new standard behaviour (?) of Asterisk, I want to show the original caller ID. I tried the options o and f in the

I am getting the following message when trying to lookup up a number via Dundi: Feb 1 13:39:24 NOTICE[20146]: pbx_dundi.c:1309 update_key: No such key 'office.pbx.bluegrass.net.pub' for creating RSA encrypted shared key for '00:a0:c9:55:91:89'! I have created keys on each box with "astgenkey -n office.pbx.bluegrass.net" using the host name for each box of course. I

...e) and keeps it that way no matter how long the call is ...( if no "Connect fee" -stays empty). i.e. [inbound] exten => 1122334455,1,Set(CALLERID(number)=${EXTEN}) exten => 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4) exten => 1122334455,3,Hangup Thanks in advance, benchev

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Hello! There's the "g"-option for the Dial-cmd that allows to execute the next extensions in the current context when the callee hangs up. I would need the same for a call where the caller hangs up, concretely i have to inform a agi-application of the end of a call. Does someone know a way to do this from the dialplan? thanks Christian

Hi guys, i've been using iax2 show netstats and i wonder if someone could explain what all these means, just in case i have them wrong. Because i am looking for something that tells me that there is delay , and/or packet loss. -------- LOCAL --------------------- -------- REMOTE -------------------- Channel RTT Jit Del Lost % Drop OOO Kpkts

Hello, I'm trying to use MySQL for Dialplans and have followed the Asterisk RealTime Extensions setup. The MySQL table is called "extensions" and I have entered two records.. ext 1000 and 2000. I also added switch => Realtime/default@extentions in extensions.conf and extensions => mysql,asterisk,extensions in extconfig.conf I do a *CLI> dialplan reload but when I show

Hi I wonder whether anyone got the Sipura ata 3000 to decode British Telecoms callerid and pass it to asterisk? The userguide seems to suggest that this is not possible, is that right? Conrad

Hey everybody, I've been using mysql databases more and more. I've run across a couple of instances where I've either made a mistake on the ip address of the mysql database or for whatever reason, mysql wasn't running. In those instances, I've noted that the mysql command will hang indefinitely (I've counted to 40 before killing it). The offending line is: exten =>

...mobile, open all tree lines and listen "stereo" with no problem. I think it is something to do with the machine since on another one: Machine AMD2.8G w/1G RAM Debian (testing) ; kernel 2.6.12-1-386 asterisk 1.2.1-n-all incl. astcc everything seams to work fine? Any experience of a kind? benchev -------------------- -- Executing Answer("SIP/asterisk-d349", "") in new stack -- Executing NoOp("SIP/asterisk-d349", "Call from spa3000") in new stack -- Executing DeadAGI("SIP/asterisk-d349", "astcc.agi") in new stack -- Launched AGI S...

What I'm retrying to do is update mysql field with the new message ID that was just recorded. Ideally, I'd like to specify the field to update using a variable ${BINID} and use ${NEWPHRASENAME} for the value - I'm not sure asterisk will allow using a variable for the field name and if not, I'll attempt to create an exten for each bin to update. Here the method I'd like to

Hello all, I have an ISDN termination box (TR1) that converts ISDN(Bri) to 2 normal analogue lines. The same number is assigned to these lines. These lines are connected to 2 spa3k registered to my asterisk box. When calls arrive, TR1 try to pass call to the first spa. If spa not takes the call immediately then try to pass to the other spa. The only configuration I found works is to put the

I want to limit the number of simultaneous incoming calls that my IAX DID can accept to, say, 2. The IAX DID provider sets no limit. The code below does work, but when the limit is in effect, new callers hear a "call cannot be completed as dialed.." message instead of a busy signal. Maybe this is an issue with the provider, but I do not like this and want callers to hear a busy signal.

Hi, I'm trying to start with Asterisk, but I could not put 2 softphones to talk. The asterisk server rejects the connections always when I dial. May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 192.168.0.106 What is necessary to put it to work? There is no need to configure external lines. extensions.conf [internal1] exten => 311000,1,Dial(SIP/teste1) [internal2] exten

Hello! Is it possible to connect Siemens HiPath 4000 to Asterisk? What equipment required on Siemens side? I mean IP not E1. Sorry for asking here. Siemens-related websites use "salesperson language". There is no technical information.

Paste your extensions.conf section that is relevant. -Matthew ----- Original Message ----- From: "Greg - Cirelle Enterprises" <gcirino@cirelle.com> To: <asterisk-dev@lists.digium.com> Sent: Monday, December 27, 2004 4:32 PM Subject: [Asterisk-Dev] realtime voicemail > Let me clarify my last message. > > If I put in the wrong password I get polled > again for

I have not used astcc with pin codes so far, since I set-up the phone number as card number. Some of my users want now to dial in to the system and than use their card, which is their phone number. For that I would need a way of authentication, like a pin. I want to use something like: What is your card number: <user keys in the number> Enter your pin: <user enter a long pin>