similar to: *8 (call pickup) using Manager or AGI interfaces?

Hi. I've found a problem when I pickup a remote sip phone with *8. There're both budgetones 102 and are both in the same group. When one sip phone is ringing, I can pickup the call from another sip phone, but the first one keeps playing a loud busy signal... that don't go away until I receive another call or go off hook and then on hook on the first phone. I think that could be a

I'm working through my list of features people will expect, and Hold Pickup is at the top at the moment -- has anyone done any work on this? We've had some unpleasant experiences with call parking, and everyone seems to like the Hold Pickup model. If you don't know what I mean by Hold Pickup, it's sort of a reverse transfer; pick up the nearest phone and dial

Hello all, We have a customer who upgraded from Asterisk 1.6 to 1.8, and pickup groups which previously worked fine have stopped working. Can anyone advise if there has been a change in how pickups work? Here is an example where 1000101 is trying to pick up a call to 1000103: <SIP/product-local-00000005>AGI Rx << EXEC Dial "Local/1000103 at product-pickup

Hi, In my office I%u2019m using mixed architecture of Zap and Sip phones, everything works fine but I have got some problems with picking up Sip channels. To be certain I can%u2019t do it at all, after I%u2019m dialing *8 the console says "nothing to pick up" (despite I configure appropriate callgroup and pickupgroup). Do I need some additional application or Asterisk code

Hi, I'm wondering if it's possible to get Caller ID information from a Call Pickup. specifically on a Cisco 79xx SIP handset. I've setup a speed dial line on my 7940 to dial *8 so I can quickly pickup a call, but because the 7940 initiates the call the information on the screen is "To: *8" I gather because of the call setup it's probably not possible to

I have an analogue trunk to an AT&T Definity. It has a DISA context defined. From a Definity handset call the analogue port extension 1008 and wait for dial tone from asterisk. It takes between 3&4 rings. Likewise from Asterisk SIP handset <PBX Access No><PBX Extn> takes nearly 10 secs to ring. Is this configurable? Ian Cowley -----Original Message----- From:

I am using PBXware for configuring users and extensions. Pbxware uses Internal script called init.sh to process the calls based on its own version of extensions.conf defined in the GUI. I have IAX2 Extensions 56 and 101 and SIP extensions 50 and 51. I have used IAX2 extension 101 and dialed SIP Extension 51 But the PBXWare's Init.sh AGI command identifies the DNIS as another IAX

hello, does anybody have some agi script that can do following : when extension didn't pickup phone call, system send mail notify ( via sendmail ) to user mailbox with date, time and caller id ? ( something like missed call ) please can somebody help me with whis ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax:

hi, i'm attempting to develop an enhanced privacy manager using AGI. the flow is roughly as follows: 1) intercept calls for users w/privacy manager enabled and execute an AGI. 2) play a notification to the caller, and prompt them for their name, recording their voice. 3) loop an "awaiting caller's response" sound. i have the above steps working but here's where

Hi all, I'm new on this list, my name is Noel. :D I developed a system using AGI and now I'm trying to develop a system that listen events fired by Manager API. I have realized that I don't receive events from commands made by an AGI script like "play a file" or "record a file". Is there a way to receive such events? Sorry about the poor English. Thanks in

Hi, What do I need to do to get asterisk to NOT pickup a Zap channel when it rings? The channel in question is used for outbound calls only, and all incoming calls are answered by an analog phone elsewhere in the building that does not run through asterisk... so.. either make it not answer.. or make it delay for like 90 seconds.. I've tried wait's.. but it still seems to pickup the

Hi. Today I was asked about a function of asterisk. That's what it should to: a call arrives -> put it in a queue -> remain here ;) Then, when someone wants to answer, just dial an extension and the older call that's in the queue is picked up. A sort of app_queueonly + app_pickolderqueuecall . As far as I know asterisk doesn't support that, so was wondering if someone

Hi all, how can I pickup a call from a queue? Which context parameter do I have to specify? The context that calls the queues application is ext-queues. This is what I tried so long (777 is the extension of the queue I want to pickup from): exten => _**ZXX,1,Noop(Attempt to Pickup ${EXTEN:2} by ${CALLERID(num)}) exten => _**ZXX,n,Pickup(${EXTEN:2}@from-did-direct) exten =>

I'm having trouble with call pickups. Suppose ring group is 100 and has extensions 101 and 102. Someone calls 100, 101 rings and 102 wants to pick the call up. If 102 dials **100, call pickup works. If 102 dials **101, call pickup fails. In my dialplan I have: exten => **101,1,NoOp(pickup extension) exten => **101,n,Pickup(101) exten => **101,n,NoOp(pickup group) exten =>

I have setup as per the monitor example configuration on the wiki site and all works well for an extension dialing 8 then the number. However, if I dial from an AGI script the recording stops after a few seconds. I see an extra answer in the console and suspect that is the reason. Could any kind soul help me to get around this? Extensions.conf.. exten =>

I have installed the Perl library from http://asterisk.gnuinter.net/asterisk-perl and am wondering how I reference agi-test.agi from extensions.conf I have added exten => s,1,AGI,agi-test.agi but that doesn't seem to do it. Is there a certain directory .agi files should be, is that the problem? TYIA Dan

Hey there list ! I'm receiving negative feedback when people try to pickup another ringing phone by pressing *8 on there own Grandstream device. These are my setting that should make pickup possible : all my sip-clients (Grandstream) have this in their config (sip.conf) : callgroup=1 pickupgroup=1 canreinvite=no qualify=yes So they are all in the same pickupgroup... This the

Looks to me your server is hacked through the webserver of website. Stop apache, flush the postfix queue, are there new mails entering you postfix queue? If not, hunt down the leak, if it does, its not apache2 ;-/ Greetz, Louis > -----Oorspronkelijk bericht----- > Van: HomerWSmith at lightlink.com > [mailto:owner-postfix-users at postfix.org] Namens Homer Wilson Smith >

Hi, I am pretty new to Asterisk. I am trying to make sure some human being answers the phone not the voice mail machine. How can I programmatically identify that? Here is my Sub: sub DialPhysician { my ($self, $con, $PhysicianPhone, $call_id, $conv_id) = (@_); to_log($self, "Inside Dial Physician", 2); my $DocPhone = "1".

Hi, according to the description of Pickup() on page http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup I can use this command to pickup a call at a certain extensions. When I try this with e.g. exten => *8200,1,Pickup(200) Asterisk tells me that the highest value for the Pickup command is 63. Wenn I enter the number of a callgroup instead of an extension, I can pickup the call.