similar to: Transfer after pickup

Hi to all, i already searched the archive without finding a solution to my problem. I have asterisk installation 1.2.18 to support multiple virtiual PBXs. I use SIP peer in the format <ID>-<EXT> to let every virtual PBX to share the same numbers of EXT. Ex. (PBX ID 10 Extensions) 10-101 10-102 10-103 (PBX ID 20 Extensions) 20-101 20-102 20-103 I use some rules in the dialplan to

i have TB instaled and i cant get call pickup when another phone rings i tried ** , *8 , *8# , **+ext but nothing seems to be ok.on extention menu i put call pickup=1 and call group=1 but nothing look at my features.conf; ; Sample Parking configuration ; [general] ; do not manually enter parkinglot config information, use the parkinglot module ; ; the parking_additional.inc file is

I cannot seem to get any transfers to work at all. The console show I have #1 amd #2 set up for Blind and Attended Transfer, but when I hit these buttons on my handset nothing happens (other than I hear the dtmf tones on the other end of the line). roo*CLI> show features Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8

Hi there, I have setup Asterisk with a couple of Sipura SPA-841's and Grandstream ATA's. The problem is that with both of these devices the Unattended call transfer process seems to be just like Attended but instead you hang up as soon as you have dialled the number of the party your are transferring to. The call transfer all works fine BUT as you complete your side of the transfer

*CLI> show features Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # ## Attended Transfer *2 One Touch Monitor *1 Disconnect Call * *0 Dial option is tTwWr I tried to call from 601 to 615 601 keys in *0

Hi, I've got attended (superivised) transfer working with a handful of SIP phones, connected via different ATA's to an Asterisk CVS-D2005.05.28.22.00.00-07/12/05-20:47:08. pingu*CLI> show features Feature Default Current ------- ------- ------- Pickup *8 *8 Blind Transfer # ** Attended Transfer

I have a TDM400 in the server. I want to press **1XX to pickup a call. It is ok if I pickup a call dialled from 1XX to 1YY (internal network call). However, it is failed to pick up a call from PSTN thro' TDM400 card. It seems I can't guess the correct context of it. How can I know the context of the call using CLI? The default context of the TDM400 is from-pstn but pickup still

Hi, Since we're moving from a legacy PABX that has been serving one of our customers for more than 15 years, we'd like this process to require no "human habits" change among the users. Software: Asterisk 1.4.22 Hardware: Polycom phones (mainly 430/601) Here are the "problems": We did configure call groups, pickup groups, ... - When someone picks up a call from

Eric Wrote: ----------- The trick is not to use options you don't understand. "show application dial" will show you what the t and T options are for. Most people use the transfer feature of their phone, rather than using the T/t hack on the Dial line. Sounds like you are using CVS-HEAD and so will have to configure stuff in /etc/asterisk/features.conf. /Snip/ Eric, Thanks for

Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I'm not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David

Asterisk seems to be ignoring DTMF for features in Asterisk 1.4.0b3 My SNOM sends the dtmf-relay and Asterisk gets it because I can see it in the sip debug. However, once seen, Asterisk doesn't actually do anything about it. I've checked features and that seems fine. Is this a bug or something that I've screwed up? For the record, here's the features setting: asterisk*CLI>

Greetings list, We recently installed some Linksys SPA962 + SPA932 sidecars into a client's offices. The BLF functionality works fine, but call pickup using the BLF is something of an issue. Following the advice on voip-info.org, I configured part of their dialplan as follows: exten => _**2XX,1,Pickup(SIP/${EXTEN:2}) exten => _**2XX,n,Dial(SIP/${EXTEN:2},15,tw) exten =>

I need attended transfers, but I do not have time to talk to another extension and see if they accept the transfer, my features.conf is: [general] parkext => 700 ; What ext. to dial to park parkpos => 701-720 ; What extensions to park calls on context => parkedcalls ; Which context parked calls are in parkingtime => 220 ; Number of

Hi all, We have an asterisk version v1.6.1.20 with a TDM400 board (2 FXS and 2 FXO). We want to do a transfer "blind" and "attended" from a line external connected to one FXO. We have made configuration, and transfers from internal lines (FXS) work fine but from (FXO) not. We have made 2 test, one work fine from FXS and the other form FXO no. Test 1, work fine: 1) A

I have a small problem, Whenever we pickup a call using *8 then try to transfer it via flash or # transfer the call is dropped. Any ideas? Whe have all called exten's in extension.conf ending in |t som that # transfers work. What am I doing wrong? Chad Sawyer, Manager, Network Administrator -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have a system with two cards: a HFC-PCI ISDN and a TDM21B (2 FXO and 1 FXS), running Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l with freePBX beta5 dialplan. I have connected an analog phone to TDM FXS port, but when I pickup the phone to make a call, Asterisk "hangs up" the call. Let me explain: In another system, when I pickup the phone, Asterisk give me tone to dial: >---

On 12/21/2014 04:42 AM, Patrick Beaumont wrote: > Have you enabled DTMF logging and seen the DTMF codes being recognised by > Asterisk? I had a bunch of soft phones that I had to change to using ?sip > info? for the DTMF signalling as the RFC signalling was not always being > recognised. This would cause transfers to appear as if the user had not > dialled any digits. > > >

Hi, I have been trying to enable attended transfer for callee. When the callee pressed *2, DTMF tone was heard by the caller. But when the caller pressed *2, attended transfer started. It's strange. I used two SIP phones. My Asterisk version is "Asterisk CVS-HEAD built by root@router on a i686 running Linux on 2005-06-27 06:07:18". In features.conf, I have: [featuremap]

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

using CVS HEAD :) Some config snippets: extensions.conf: [from-ip] exten => 51,1,Dial(SIP/1301,20,t) exten => 52,1,Queue(ddi831,t) exten => 53,1,Queue(marketing,t) [internal] exten => _13XX,1,Dial(SIP/${EXTEN},20,Tt) queues.conf: [ddi831] strategy=roundrobin timeout=10 announce-frequency=0 announce-holdtime=no member => SIP/1301 [marketing] strategy=roundrobin timeout=10