similar to: Callback / Camp / Extention Free notify?

Hi all, Having fixed my problems with the call waiting ringing on the GS phones, I needed to extend that with a campon facility (available on some legacy systems - sort of callwaiting without phone ringing). I've managed to implement that by adding/modifying app_queue.c. Basically, when calling the SIP phone, and if busy, I can camp the call onto that extension, with MOH, allowing the caller

Hi, A potential customer would like to be able to do this: If a call comes in for an employee who is on the phone, allow the front-desk to push the caller in a queue directly to the employee. Now, this is easily done by using queues, but I am curious: What is the performance impact on a system if _every_ employee (phone) has their own queue. How scalable is that in comparison to

Hi I can't for the life of me work out why this is not working. When in the campon contect if you hit a DTMF key 2 you get moved to the exten => 2 defined in the mainmenu context not the exten => 2 defined in the campon context. What is wrong? The same happens if you hit key 1. [campon] exten => _*1XXX,1,Answer exten => _*1XXX,2,SetCallerID(${CALLERIDNUM}) exten =>

Hello Happy New Year to all!! I've just completed porting from Asterisk 1.2 to 1.4. I did this by doing a clean install on a new box, and moving over configuration and scripts where needed. All went surprisingly well! Anyway, one lingering issue is that the function key "lamps" on our Snom phones have all stopped working! We're using a mix of Snom 290/320/360 phones and

Can I ask for some advice on dial-plan construction please I have setup my dialplan to use 9 to get a zap trunk, leaving everything else for internal extensions. However, this creates a problem in that my callerid is correct, but doesn't work to re-dial the incoming caller. So if I simply click "missed calls" on my Snom phone and hit redial then it tries to dial an internal

Hi Is there any way to park a call on an extension. Say a customer calls in and wants to speak with Steve. Steve is currently on the phone. How can I park the call so that when Steve hangs up his phone immediately rings and he is connected to the parked call. Thanks Kelly

Hi all, I'm about to start setting up my first asterisk/cti system in our test lab. I've read through all the documentation I can find and relevant posts in the list archives but can't seem to find anything explaining how to go about initiating an http request upon an incoming call. I basically want asterisk to request an uri on our intranet, which will pass call details to our

Whenever I try to drag calls to the Parking Lot or On Hold, FOP would drop my calls. I have searched online and have found similar problem, such as the link below. I have tried their solution but still the FOP is not working correctly. I even installed the HUDLite server and is getting the same results. www.freepbx.org/forum/freepbx/users/flas...ot-transfering-calls Here is the log when I tried

Hi, I am using Group and Group_Count to limit the number of calls to go out over a single peer as our channels with that peer is limited to 8. If we dial and outside number over this peer and then transfer the call into a MeetMe conference the Group gets decremented when it should not? This is most likely an error on my behalf, however I am not sure what the correct solution is. I have set

I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH

I have been discussing an asterisk solution with a company that has a custom written dialogic based solution. The issue is that their dialogic solution can read callerid from incoming calls, even if the callerid is blocked. I have read before that Asterisk can do this, and they want me to make sure that their new system will be able to do this. A quick poke around inside the zaptel source code

Hi, I am having some trouble with parked calls timing out. In features.conf: [general] parkext => 800 ; What extension to dial to park parkpos => 801-820 ; What extensions to park calls on context=parkedcalls parkingtime=120 After the Park timesout it calls the phone that the call was parked from. If the phone is BUSY the call just get dropped. (Call waiting

Any idea how to read an external file, grab some stuff and push it back into an Asterisk variable? I can do it the other way with: system(echo "${UNIQUEID} =>" >> /home/ast/curr_calls) but I'm a bit stumped on how to go the other way around.... much thanks, Paul Hales

Hi, I explain what I want to do.. All the operators share their phones. The number of the operator isn't constant, so it's possible that two operators share all the phones. They need to move all around, so they pick up the first phone they find. If there are only few operator is very annoying for them to ear the other phones ringing while they are at the phone! So I'dd like to limit

We have set up an IP telephoney system hosted by Asterisk and its working pretty well. We primarily use SIP and hardware IP phones. We have the ability to transfer calls to another SIP phone using either the "Transfer" button on the phone (these phones are Grandstream BudgeTone 100s) or using the "#" key (the T/t flag must be set in the Dial command in asterisk for

thankyou both very much for your swift responses and helpful insight... While my knowledge of administrating Asterisk is fairly decent, i must say my knowledgebase and ability in troubleshooting it is fairly lousy... all these wonderful suggestions you have had about turning this log on here, etc sounds great, but i don't know where to begin on that! how do you recommend i turn these on or

I did a cvs update this afternoon and since then asterisk doesn't seem to clean up the channels after they hangup. This has been working perfectly for quite some time previously... I do a show channels and it shows the channels still up. The only way out is to kill and restart asterisk.... I am frantically trying to would out how to get a non-current CVS copy of the source and get it back

Does anyone know of any startups using Asterisk? What about established companies? Ones that are hiring would be nice.... :) Doug.

When agent tries to transfer a phone call (*2 - att transfer) he hangs up. Why? When a phone call isn't from queue then att transfer works fine. In features conf I have *1 for recording, *2 for att transfer and #1 for blind. In queue blind transfer works. For disconnect I have #0. I guess that * is somewhere defined as for hang-up the call, but where? I can't find it anywhere. Any help

Does anybody know of any IP phones (ideally SIP based) that have interfaces to plug into a pro audio system (eg for phone interviews). Something can probably be hacked up with a headset connector or the 1/8" jacks on a 7970 but I'm wondering if there's something better out there. Thanks, Julien