search for: pickupextens

How to configure Digium S101i adapter to work with "pickupexten *8" ? I have few Sipura adapters and "*8" work OK but my new Digium S101i refuses to cooperate. -- #Joseph

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, I'm having trouble capturing calls using the chan_dahdi with dynamic span. Here my settings: chan_dahdi.conf [trunkgroups] [channels] context=default switchtype=national facilityenable=yes rxwink=300 ; Atlas seems to use long (250ms) winks ; where the ring cadence is changed *after* the callerid spill. usecallerid=yes hidecallerid=no

I can not make a "call pickup" to work with Sipura-3000. I have one SIP phone and one is connected to ATA Sipura-3000 I've in all sip.conf context callgroup=1 pickupgroup=1 in features.conf I've tired: pickupexten = *88 pickupexten = *8 Nothing works. What am I missing? -- #Joseph

Hi group. I have huge problem. My pickup exten #8 isn't working. This is what I have configured. pbx*CLI> show features Builtin Feature Default Current --------------- ------- ------- Pickup *8 #8 In sip.conf I have callgroup=2 pickupgroup=2 For called party and same for person that is trying to pick up the call. The person that is trying

Hello all!, I?ve running asterisk 1.4.23.1 and I need to get working pick up from feature.conf. It does no work, only I can connect but cant send audio over the phone. Is there a bug with this feature?. Thanks for any response! Cheers! Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was

Hi guys! I?m using Asterisk 1.2 with mISDN support. I have problems with Pickup calls with my Grandstream Buttons . I set up on Dial Plan this: Exten => _**XXX,1,Pickup(SIP/{EXTEN:2}) but it doesn?t work if the call comes from mISDN. So, I wanna do something to this: Exten => _**XXX,1,SendDtmf(*8#) because if I introduce *8# into my telephone i can pickup a call from everywhere. BUT the

Does anyone know why the bt100 cannot park and pickup a parked call? attendant announces the call is parked at extension 701 but the call cannot be retrieved by any other phone. also, the bt100 constantly rings when the phone is hung up after parking. anyone experienced this? using the basic features.conf [general] parkext => 700 ; What ext. to dial to park parkpos =>

Daniel Heckl <daniel.heckl at gmail.com> schrieb: > You are searching for ?Call Pickup?. It is implemented in Asterisk by > default. > > https://wiki.asterisk.org/wiki/display/AST/Call+Pickup > <https://wiki.asterisk.org/wiki/display/AST/Call+Pickup> Take a look under > section ?Configuration Options?. Hi, Daniel! Thanks for your answer... I'm using Asterisk

Hi all First I want to wish for everone a happy new year... Well... I have run asterisk 1.4.16.1 in a server. I have this IVR, in extensions.conf: [ura] ;exten => s, 1, Wait,1 exten => s, 1, Answer() exten => s, n, Noop() exten => s, n(debug),DumpChan() exten => s, n, Set(LANGUAGE()=pt_BR) exten => s, n, Set(CALLFILENAME=/var/spool/asterisk/monitor/entrada/) exten => s,

Hi i have two problems with my Grandstream GXP2000 : 1- When i wan pickup a call, that's don't work's (*8EXTEN) and when i test whit Softphone, i have a error too, he say me *8201@MyContext not found .. in features.conf, i have: [general] parkext => 700 parkpos => 701-720 context => parkedcalls

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

Hi all. I am trying following scenerio for call park & pickup. voice is flowing established between B & C, after call-pickup ( instead of A & B ). can anyone please clarify why it is happening like this, ( or ) do i need some more configuration for park&pickup ? A B

Hi, I try to set up attended transfer in my Asterisk Box . My features.conf look like this: [general] parkext => 100 parkpos => 1-5 context => parkedcalls parkingtime => 100 transferdigittimeout => 3l courtesytone = beep xfersound = beep xferfailsound = invalid featuredigittimeout = 500 ;adsipark = yes pickupexten = *8 [featuremap] atxfer => *2 blindxfer => # disconnect

Hi all, There are a lot of posts around the web about my question; unfortunately I have not been able to get any of the solutions to work. I'm using Asterisk 1.6.2.8 under CentOS 5.5. I'm trying to get call pickup working for the secretaries that monitor their bosses' phones. The BLF and the speed dial works great on the Linksys phones. Call pickup is the problem. My features.conf

Hi, I would like to use the new atxfer (native assisted transfer, see mantis item #3241) , but I've partially been able to make it work. I can receive a call and then having the caller hear MOH while talking with another extension (the one I want to transfer to), but then I can't make the caller and the trasferred talk hanging up or pressing any key combination I'm aware of. My

Hi list, For months everything worked super here in our setup. This week I implemented some new idea in our webbased calendar system. I thought it would be nice to have an option that tells asterisk you are not available for calls during an appointment. For this to work I could no longer use the ringgroup setup: Dial(SIP/10&SIP/11&SIP/12,40,tr) So I thought, why not use the Local channel

In the list I found some messages that *8 doesn't work so well. Is there any possibility to create a extention that you can call, and if you are fast enough, pick up a number? (Also if you are outside your callgroup) like pseudo code: exten => 888, 1, EnterPhoneNumber() exten => 888, 2, EnterPass() exten => 888, 3, TransferCallToThisPhone() exten => 888, 103, Invalid()

Two questions: 1. Does anyone have a good way to transfer a call from inside comedian mail to the current extension? The problem is: let's say the phone goes to voicemail after 4 rings, yet I don't hear it until the 3rd ring. I come running into my office but miss it by a split second. Is there a way I can barge in on the person leaving a message for my mailbox while they're

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf: