search for: lanvik

-----Original Message----- From: Chad Brown Sent: Tuesday, December 21, 2004 8:02 PM To: 'el_flynn@lanvik-icu.com' Subject: RE: [Asterisk-Users] Dialplan help - Can dial any user but not thePSTN Flynn, Yes, that makes sense. However, in my case I have incoming calls arriving on an IAX channel from a PSTN gateway. I think the concept is the same. That said, if incoming calls have access to the in...

Hi all, Just wondering if anyone's come across this issue, and what might be a fix for it: We've got several BT-101's deployed, and upgraded to firmware v.1.0.5.16. The phone can do proper supervised transfer, but _only_ once. If the user attempts to transfer a second time, it won't work. any suggestions/hints/tips are welcome.. Flynn

I just started poking around with writing a python module to interface to the Manager API, and it suddenly hit me... how the heck are you supposed to program this thing? All the events seem to be dumped to all the open connections. If I send a command, such as a login, there seems to me to be no way to determine which response are intended for me, and which may be intended for another open

I am currently developing a asterisk ncurses interface using the manager API. The project is currently awaiting sourceforge's approval but I have a beta online at http://sig.lange.googlepages.com/assman . The projects real home will be assman.sf.net. This project really consists of two parts, libassman is a C manager API and assman is the ncurses portion. It's still beta but I have been

Hi all, I used to have an OpenLine4 card, but decided against using it due to some problems with hangup detect. Does anyone on the list actively use Voicetronix's OpenSwitch12? What are your opinions on the card? Cheers, Flynn

Hi all, I'm just wondering about these VoIP services -- do you have to sign up one account -per- client that will be using the service? I've got multiple extensions behind my Asterisk box, and I want to be able to allow all my staff to place calls via the provider. So if I sign up for one account, will multiple users behind my Asterisk box be able to make calls, using that same

Hello all, I'd like to mention that we've put together a simple Java-based application that provides a somewhat point-and-click interface to create an Asterisk dialplan. You can get to the dialplanner at http://www.lanvik-icu.com/asterisk/dialplanner/index.php You can create contexts and extensions, then select the appropriate command from a list. Then you'll be prompted to enter the arguments for that command. The dialplanner will show a nested tree-based view of your dialplan. Once you're done, you can c...

hi all, i've got a proposed setup that i was wondering if you guys could comment on. the client wants * and a couple of SIP phones to be on a separate network than the rest of the office, so that in case their primary network crashes for some reason the PBX won't be affected. one other factor: the client may at some later point set up SIP UAs sitting on the primary network that will

Hi there, I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. Most of the VoIP phones I've looked at only have 4-6 line presentations; is anyone aware of one that has more? I tried to get some info about Snom's Keypad 220 since it has loads of programmable

Hi, I'm running two boxes side by side, identical specs and setup but with differing dialplans. Both are on ast/zap/libpri versions 1.0.9. Both boxes share the same folder for voicemail, exported via NFS from another file server. Everything was working fine for an extended period of time, until just recently when someone rebooted Box A. Now when I dial an extension associated with a SIP

Hello, I am very new to asterisk and voip in general and so far have managed to get the FXO card and a few sip phones working fine. My question is where does the Sipura SPA 2000 come in the picture? Can it be used as an extension (i.e FXS) ? Or is it to be used as a line (i.e FXO)? Or it can be used as both? My understanding is that its just like another ATA186. Is that true? I guess what I

Hi all, I was wondering if anyone had any pointers on how to determine whether or not a module has gone wonky on the TDM400P? I have a 2 FXO (channels 3 and 4) and 2 FXS unit (channels 1 and 2). The bad (?) module in question is the FXO module on channel 3. I can't dial in to or out of that channel; dialing in gives a busy signal, dialing out just shows * hanging around after attempting a

Consider this dialplan fragment, where the call is being dialed into [macro-process-routing] over an iax2 channel from another (same build) Asterisk server: [macro-process-routing] ; This is the entrypoint of the debug call but is also refered to by Macro(process-routing) elsewhere in the dialplan ; XXX-NNN-6800 exten => _6800,1,Macro(6800-interceptor) ; This is matched when 8 is

...ium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ------------------------------ Message: 8 Date: Tue, 18 Jan 2005 16:21:04 +0800 From: el Flynn <el_flynn@lanvik-icu.com> Subject: Re: [Asterisk-Users] Out of 5 Grandstream BudgeTone 101 THREE are defect !!! (from Pulverstore) To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <41ECC6F0.4030508@lanvik-icu.com> Content-Type: text/plain; ch...

I am trying to connect to a vocal server from an asterisk server. A call is received via iax2 to my asterisk server. I then initiate a SIP connection to the vocal server. everything works great except dtmf doesnt work. A cisco 5300 can connect to this vocal server and do dtmf without a problem. I have my dtmf set to rfc2833 in the general section of the sip.conf . I can confirm that the

I've just installed an X100P, built the kernel module, and tried to use it to make an outgoing call (via a phone connected to an ATA-186). However, I just get a reorder tone and see this on the console: -- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in new stack NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type

Hi, Is anyone aware of an integrated management tool for asterisk? Specifically, I'm looking for something that can: 1) Generate CDR reports 2) Manage a 'switchboard' 3) Add/remove/edit extensions So far I've seen applications that do one of the three, but I haven't come across something that does all three. This tool would be useful installed for a packaged * box that

hi, I need some feedback on an asterisk installation my company is planning to supply. Here's the basic structure: PSTN -> Asterisk -> Channel Bank -> SDH --- fiber --- SDH -> analog extension Some additional info: 1) Fiber connects the main exchange with 8 other substations, with an average distance of 13 or so kilometers between each station. 2) Channel bank needs to connect

Hi there, Does anyone on the list know what sort of power the Adtran Total Access 850 channel bank consumes? I'm trying to put together 10 of them and need to know what sort of UPS should be hooked up to them. Client is asking for 10 hours backup time... or should I just go with the 850's optional 8-hour battery backup system? Flynn

Hi, I need some recommendations on a suitable setup for our operator console. I've got 12 incoming analog lines hitting the * box. * is then connected to 70-75 extensions via some channel banks. Would one of the Mitel/Snom SIP phone types that support multiple line appearances be best, or would something PC-based as suggested in the Asterisk GUI Wiki be more suitable? What do some of you