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...ls and Transfers via drag and drop. The difference with other similar tools is that it displays status in real time (no refreshing necessary), and its graphically appealing. It's a work in progress... so expect some bugs. I appreciate any feedback you can give me. Best regards, -- Nicolas Gudino <nicolas@house.com.ar> House Internet S.R.L.

Hey Nicolas, That did it. I ran that export command you suggested, then launched *, everything worked fine. I'm still looking for info on what that command actually does. Can you shed some light please? Thanks. JR -----Original Message----- From: JR Richardson [mailto:jr.richardson@cox.net] Sent: Tuesday, December 30, 2003 6:44 PM To: 'asterisk-users@lists.digium.com' Subject:

Just upgraded to cvs Head this morning and noticed our voicemail notification (via email) is failing with: Jul 1 07:48:38 WARNING[1217669936]: app_voicemail.c:837 sendmail: E-mail addres s missing for mailbox [3000]. E-mail will not be sent. However, a valid address in voicemail.conf has been working just fine until now. Sendmail is running, etc. If I add a "second" email address

I have a question for the group. To get this running do I need any Digium Cards? I understand I will need them to connect to the public phone system. I'm looking at just using IP Phones or IP Softphones just to test this app. Thanks for any help you could give.

...s] AGI problem (crash) in RH9 You may wish to upgrade your kernel to 2.4.20-20.9 through 'up2date'. Regards, Ray Burkholder > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Nicolas Gudino > Sent: October 17, 2003 16:17 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] AGI problem (crash) in RH9 > > > Hi Ivar, > > Try putting this line before launching asterisk: > > export LD_ASSUME_KERNEL=2.4.1 > > Best regards, > >...

hi all what about this... I just put together a box on a web shop (komplett.no) that will cost me NOK ~1850 (? 216) plus a small ?50 drive and cables, so say ?300. This consists of a cheap MB with a duron 1400, 256MB SDRAM and two HFC-PCI cards (if capijod will finish off the zaptel-driver soon). This is all in a cheap PC case. What do you think? Should this be doable? as a product? With only IP

...om the 's' extension. It works great, but the CDR shows 's' as the destination. I need a CDR record with the number passed to the dial command as dst in the CDR. I'm sure there is a proper way to handle this.. and I'm sure someone can help me figure it out. Thanks! Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina

...S-10/10/03-19:24:38. In the console nothing shows up either. Do I have to upgrade to a more recent CVS version? Do I need to enter more parameters or configurations in other places? Does anyone have call pickup between sip phones working? If so, which version are you using? Thanks!! -- Nicolas Gudino <nicolas@house.com.ar> House Internet S.R.L.

hey i want to implement phone card application based on PIN......... for this i am planning to use the AGI......... which programming language ( c , python, java .etc) should i use? i mean which one is effective. please suggest me. _________________________________________________________________ The new MSN 8: smart spam protection and 2 months FREE*

Hi everybody, I'm trying to develop a web application for controlling if SIP users are registered in * or not, and show it in a web. My problem is that I don't now if it's possible to do a Shell Script to control this: 1. Connect to console. 2. Execute command. 3. Obtain users registered. 4. Update a BdD. This is possible? There are any best way to implement this? Thanks a lot.

We use an IAX2 trunk to our remote office and would like for the receptionist to be able to transfer incoming calls from this trunk. but all calls come in as one user, Is there a way to get a breakout on the flash GUI of the incoming calls? Thanks, Justin

Hello, I know not too long ago I saw /something/ _somewhere_ about an adjustment to call parking that allowed blind transfers from SIP phones to park a call and still be able to hear the parking lot stall number. Unfortunately, I have no idea where I saw that (google turned up little, couldn't find it on the list either). I'm using Sipura SPA-2000 adapters and it doesn't seem to

Hi, 1. When an agent is active on a call, i need the ablity for a third person to join the conversation. Basically barge in by a supervisor, participate in the conversation and then leave. 2. As an extension to the above, while on call, can the agent request a conference from another agent and later hang him up. 3. Is there any way for a call to be put in the queue destined for a specific agent

Hi, I guess I need some help with management interface. I would like to watch calls through the management interface, but I don't know how to identify which call an event belongs to or in other words how to associate a call and uniqueid field of event. Let's say I send the following manager command: action: originate channel: sip/12125551111@pbx1 callerid: 12125551111 MaxRetries: 1

Hi, How can I (through the manager interface) steal a call from one phone, and transfer it to another? Does asterisk provide for actions like this? It's a common action in Lucent systems it seems. Cheers, Ben -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040816/7a79aa0f/attachment.htm

Can someone confirm what I should expect the correct behaviour to be on incoming calls if I have multiple SIP phones configured for the same username? I'd expect all the phones registered under the username that that extension is associated with to ring, and the first one that answers gets it. What I get, is just the first phone that registered gets a ring. The second one doesn't ring at

...le to use the Cocoa framework from Python and to integrate with Interface Builder and Xcode on MacOSX. However, PyObjC also supports OpenStep on Linux. This means in principle it is possible to use PyObjC to write a cross-platform GUI tool that would run under MacOSX and OpenStep on Linux. Nicolas Gudino (the author of the Flash Operator Panel for Asterisk) and I have discussed this possibility and we will be looking further into this. I don't want to make any promises yet, but with a bit of luck and perhaps a bit of help from others, we may be able to make the existing Asterisk Assistants for...

Hi, We?re looking at options for logging agents into the system programmatically via Perl/PHP and I was wondering if anyone else is doing this and if so, how. We're using AgentCallbackLogin now but would like to set up a web interface instead. I've been looking at Asterisk::Manager and didn't see anything relevant and wanted to ask the group before we dove into the Asterisk source.

I'm looking for an enhanced speed dial "dashboard" as DSS (Manager integration) for Operator console integrated in a voip phone (softphone or hardphone, opensource or commercial) to diplay the status of phones (sip, zap, iax...) connected to asterisk. I see in snom site the snom 220 with keypad 220. Can it display the status of internal and external lines (free, busy..) and

I have a Sipura 2000 working fine, but whenever I dial any extension there is a delay of 5-10 seconds before it starts ringing. However, if I dial the extension and hit the pound key after the number, it goes through right away. Is there any way around this?