similar to: Forward but wait for acknowledgement

When I drop my file into the outgoing folder, the call is completed but the 'Context' entry is not respected. Instead, it drops into the default context. It does drop "properly" into the default context and function as would be expected. I looked through the source but didn't see any reason it would be completely ignoring the context. Call file: (where

We currently have a Merlin Legend system. The voicemail is falling apart (with the transition to a 10 digit timestamp on Sept. 8, 2001, the system locked up and refused to take calls; the official solution is to change the system time back to a year with a matching calendar). We are in the process of preparing the network infrastructure to support a VoIP system with Asterisk, but won't be

I'm in the process of setting up a test/demonstration system to show that VoIP is realistic and applicable for our needs. We put a 7905 and 7960 on a request for quote that went out the other day (to people like CDW & Microwarehouse). All of the vendors returned thier quotes without including the Cisco phones. So my question: where do you buy your phones? We can't buy direct from

I've seen mention of it here and there... does anyone have mgcp working with a 7960? I've gotten the phone to work in "basic phone mode", is that all I'll get, or am I missing something? ___________________________________________________________ Steve Creel screel@turbs.com

It just dawned on me as I was playing with the manager interface - it can't be very difficult at all to write an Win32 app that serves as a "lamp field". Between 'Newchannel', 'Newstate', and 'Hangup' events, all of the information is there. I've heard several requests for line appearances, but mgcp and sccp channels don't currently include support.

Does anyone have any suggestions for excluding certain users from the directory? Can I just leave the 'Name' field empty in voicemail.conf Certain voicemail boxes shouldn't show up in the directory (company president, etc). I assume this can be handled safely by just leaving out the 'name' in voicemail.conf To go a step further, it would be good to allow them to put the

I have asterisk running as a voicemail system off of our Merlin Legend switch. We replaced our old Audix Voice Power (when the power supply fan died and burned it up) with asterisk a week ago. Many thanks to those who provided information about integrated VMI on the legend. The Audix system would, after a mailbox was closed, wait a few seconds, then use that line to dial the switch and update

swar sir, can u please unsubscribe me for your list b.regards jihad chalhoub --- asterisk-users-request@lists.digium.com wrote: > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, > visit > > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message

I'm trying to implement parking with only one button to park and unpark a call. Scenario: Call is answered, I press the button (on a Cisco SPA504) to park the call, it comes to [from-office] context where the call is parked successfully (there is no parking lot number announcement though). To unpark, I press the same button, it comes to [from-office] context, and the call is picked

I'm trying to implement call parking with asterisk and Cisco SPA504G phones: features.conf parkext => 700 parkpos => 701-702 context => parkedcalls I defined one of the unused keys to park the calls: Key2: fnc=sd;ext=700 at 10.0.1.103;vid=1;nme=Park I also defined two other keys to pickup/unpark the calls: Key3: fnc=blf+sd+cp;sub=701 at 10.0.1.103 Key4: fnc=blf+sd+cp;sub=702 at

Hi, First off, a big thanks to Digium (Mark, John, and Martin) for helping sort out a BellSouth config issue on our PRI. T100P working like a champ! Now it's back to tweaking the configuration on our SIP phones (7960s). The message_uri parameter in the phone's configuration file is working great. Dials comedian mail directly. Is there a way to let voicemail2 know what the incoming

I've lurked for a while, but I think this is one of my first "pleas" for help. I'm having issues where a parked call using the macro below is getting "stuck". Users park the call via a blfxfer key on an Aastra phone. If the call is a blind transfer, it tries to park the call. If it isn't a blind transfer, it tries to unpark the call. Only 2 extensions (2759 and

We are planning to have about 100+ parking lots defined in features.conf , each with about 4 unique park positions. Asterisk will be handling all the parking and unparking (we don't exclusively use Park/ParkedCall in the dialplan): [parkinglot_a] parkpos => 1-4 context=parked [parkinglot_b] parkpos => 5-8 context=parked As far as I can tell, Asterisk adds/removes extensions to the

Hi, Is there a way to do a blind transfer within an asterisk dialplan (like '##')? The reason I need this (I think) rather than a regular Goto() is that I'm trying to do one-touch parking. I can park a call using one-touch parking and then pick it up again, however if I try to re-park the call, it gets lost. I think that is because asterisk thinks I'm still on the park extension.

you can do this with MeetMe, but you don't have to. you can also use Parking, which makes things a little simpler. in either case, the strategy is going to be something like: 1. Record the soundfile 2. Park the inbound caller 3. Use a .call file or the manager interface to initiate an outbound call to the mobile phone 4. play soundfile and prompt the mobile phone user to accept/reject the

I would like to build into our application a button to "park" and a button to "unpark" calls. Consider this scenario: Agent A gets a call, and obtains a reference number. He needs to send this client to another Agent. Agent A pushes the "park" button. The call is then parked into a designated slot based on the reference number . Agent A then calls Agent B and

Hi All, I got a doubt about something I want to do with asterisk. I have this office (site a) with only a Panasonic analog PBX and another office (site b) with an Asterisk Box with an ADIT 600 . I want to interconnect both via IAX. Is it possible to put a new asterisk box in site a without the channel bank and put a card (FXS or FXO???) and connect it to the pbx as a CO line ? What

How can I loop through something x number of times in the dialplan? i.e. if I get an invalid extension I want to re-play the menu, but not forever. Maybe 3 tries or something. I'm pretty sure that I've seen it before, where you can increment a variable and do "Gotos" based on it. But I've searched the Asterisk handbook, searched the user archives, and Googled for it,

I bought a couple of 7905G phones with a Callmanager license but i found on the site these phones can have a SIP image (which i downloaded) but before i upload the image i want to know if anybody tested them ? Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031016/4c2821af/attachment.htm

I guess the subject says it all.. :) I am running the CVS from right now.. +- 12:25 GMT MySQL CDR logging is installed and working.. Anyone got any ideas?