similar to: Looking for list with asterisk default extensions

hello, using a legacy PBX to access a Asterisk Zap channel (Legacy PBX FXS --> FXO application Asterisk/TDM400P) I want to be able to "flash" the asterisk pbx. However by pressing the FLASH button on the extension connected to the Legacy PBX gets me the flash features on the Legacy PBX, not on the Asterisk PBX side. I thought of using the following codes (listed below) from

Hi All! Have a problem with remote call pickup via sip. When 1 sip phone is ringing and I'm trying to pickup a call from another sip phone by dialing *8# I'm getting: -- Sent into invalid extension '*8#' in context 'from-sip-post' on SIP/ciscok-8d39 such configs: zapata.conf ------ context=inbound-analog callgroup=2 channel=2 ------ sip.conf ------ [ciscok] type=friend

Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance,

Been learning * now for a couple of weeks and have all basic features running including VM, MoH, FX lines, iaxtel, and FWD. However, I seem to be lacking documentation on a lot of technical things and am wondering if I overlooked something that is obvious to others. (I do have the Handbook, have been doing a fair amount of google searches, and read the README.* files.) Examples, Where should I

Hi, I asked a time ago what were callgroup and pickup group used for. I have done some proofs and all, and I'm not sure if I have pick the idea up well!! That's what I understand: For example: group=1 callgroup =2 and pickupgroup=2 and my phone is a membership of the group 1. that's mean that when a phone that belong to group 2 is ringing, I'll be able to answer this call dialing

I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured

I know *8 kind of works with SIP but what about the rest should they work do they work with a zap device? *0# sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid

Someone recently posted the following list as functions built into * *0# sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid I'm running a CVS from a couple of weeks ago with multiple C7960's, snom 200,

is where anyone who knows what is needed to get the pickupexten (*8) running ? gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff I've activated it in features.conf (default *8) and also tested other extensions res_features.so is loaded show features says: Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind

Hi, Our current PBX (Panasonic) has a setting that enables users to "snatch" calls ringing at other extensions. I'm not sure snatching is a correct term for this so let me elaborate. Let's say you sit in a room with five other people. Each one has it's own extension. One person goes out. As soon as he leaves the room his phone starts ringing. The other guys in

I'm trying to set up asterisk so any phone connected to channel 1-16 of my Adit600 channel bank can pick up a call coming in on channel 24. I do not wish to ring any of the 16 channels on an incoming call -- this is strictly so I can pick up the line if I see it ringing and wish to answer at work. I have channel 24 in call group 3, and channels 1-16 in pickup groups 1 and 3. However

Hi, I have been trying to workout how to use the call pickup. So Far, I have the following in zapata.conf [channels] signalling => fxo_ks context => local pickupgroup=1 callgroup=1 channel => 1-3 When I dial *8# all I hear is busy tone. What have I missed? thanks Jay.

Hello, we are running a Asterisk (1.2) installation with about 80 snom phones (300,320,360). Now have the demand for a special manager - assistant setup for a few extensions. Since Shared Line Appearance is not available in 1.2 I?m wondering how to realize this... What we need is that the manager can decide whether he wants to get calls or not. If not he must have the possibility to redirect

More than one hundred messages related to *8 or call pickup problem in last 6 months!! Please someone in the development team could clarify this and make himself responsible for the response. By now It seems a bad joke. We have spent thousand dollars with hardware, sip phones, working men hours, and with digium stuff (E1, fxo, fxs cards etc) and we have had the *8 problem (sip callee ringing

So I'm having trouble getting call-pickup working. Got a few different SIP phones (cisco 7940's and SPA-841s) all with pickupgroup=0 in sip.conf. I can't seem to get it working. This *is* possible from SIP phones, right? Do I need to add anything to my dial-plan? -- Paul A. Dugas Dugas Enterprises, LLC paul@dugasenterprises.com 1711 Indian Ridge Drive

I'm using the Debian Sarge package of Asterisk - 1.0.7 + bristuff. When I call from a zap channel or a SIP phone to another SIP phone, then dial *8 from a third SIP phone, I get 503 Service Unavailable on the third phone and I get this at the Asterisk console: Jun 28 09:01:24 DEBUG[16774]: res_features.c:1709 ast_pickup_call: No call pickup possible... Jun 28 09:01:24 NOTICE[16774]:

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

Hello all. I saw a few weeks ago a discussion about cal pickup, *8, not working but did not find a message about it being resolved, I look for a bug on the bug list but did not find anything about it not working, nor a bug open. I installed asterisk 0.9.0, have one sip fxo gateway and only sip phones, all of them have callgroup=1 and pickupgroup=1 but I can not get a call that is

hi all, I need help ! What are Built-in Extension Numbers ? if i dial *69 with callreturn=yes in zapata.conf i don't get the last caller . How may i use Built-in Extension Numbers ? I should not add extensions in dial plan !? Harry from voip-info.org: There are some "extension numbers" that are built into the Zap channel module. You may override these in your Dialplan, i.e.

Hi- I have posted a photo of the TE410P Digium card on my site, so that those wishing to purchase a compatible motherboard can see physically what the PCI slot requirement is: http://www.evtmedia.com/TE410P.htm I believe the required slot is a 64-bit, 3.3 Volt PCI, most commonly found on Xeon-based motherboards. A 64-bit slot is longer than the TE410P board requires, but the PCI connector