similar to: I don't think I understand "Call pickup"

Hi there How do I change the dialplan runtime, if I for example wants all calls on the main number to be answered by a voicemail (when it is out-of-office hours). I want to be able to change the configuration by pressing a DTMF combination e.g. *82. Can't figure out whether it is necessary to change contexts or how to do it. I have read a lot of examples and config documentation, but I

Hi I have kind of an odd problem. When dialing in from an outside line via a TE410P card it seems like MusicOnHold and MP3Player doesn't work properly (for me anyway). The remote end which is calling * doesn't hear the music but just keeps ringing. But if I insert a Playback("file_which_dont_exist") just before the Moh or MP3Player I can hear the music. Actually I observed the

Hello fellow * users Hope this isn't a stupid question; I've done my research but could not find a proper answer. I have 8 different destinations which I want to match. The numbers are: ###### 00 ###### 20 ###### 30 ###### 40 ###### 15 ###### 35 ###### 12 ###### 44 Right now I've solved it by doing this: exten => _######[0234]0,1,HangUp exten => _######[13]5,1,HangUp exten

The problem: T100P connected to an Adit600. Channel 1-16 are FXS, 17-24 FXO. I have Zap/24 in callgroup 3 and Zap/1-16 in pickupgroup 3. When a call comes in on Zap/24 I cannot pick it up with *8 from Zap/1-16. *CLI> show version Asterisk CVS-04/27/04-23:48:08 built by root@tuck on a i686 running Linux The zapata.conf and extensions.conf are located here:

Hello I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom IP300 phones. My config files look like this: features.conf pickupextn = *8 zapata.conf context=frompstnisdn group=1 callgroup=1 pickupgroup=1 I also edited sip.conf like this: group=1 callgroup=1 pickupgroup=1 But on internal and incoming calls if I dial *8 from any phone I cannot pickup. Do I need to add

Hi there I have a Welltech Wellgate SIP device and I want to be able to do a supervised transfer. I've read that in order to do that I have to use flash hook. The problem is just that I can't flash hook with this device. I'm in contact with the developer of the SIP device but don't know what to tell him in order to get him to fix this. What is happening when you flash hook, I

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.

Hi there, it appears that call pick-up only works _within_ a technolgoy, i.e. with a SIP phone when another SIP phone is ringing. Is that correct, or is my configuration faulty? * Case 1: SIP phone 1 ringing - SIP phone 2 can pick the call up with *8 We are happy! :-) * Case 2: IAX phone ringing - SIP phone can't pick the call up: NOTICE[10250]: Nothing to pick up * Case 3: SIP phone

Should setting a namedcallgroup & namedpickupgroup supersede numeric callgroups and pickupgroup ? I've got 5 peers on my 13.7.0 box, Three of them have a namedcallgroup & namedpickupgroup of 'kiniston' and Two of them have a namedcallgroup & namedpickupgroup of 'sanday'. I'm not specifying a numeric callgroup or pickupgroup so all the peers are defaulting to

Hello, I have set pickupgroup and callgroup for zap, sip and iax2 devices. Everything is working good with zap and sip and between these two. Iax2 pickupgroup and callgroup seems to be broken. I cannot pickup a call to IAX2 from SIP. Is there somewhere a bug ? I am running: Asterisk 1.2.9.1 Bartosz

Hello there Yesterday I updated my pwlib, openh323 and Asterisk from CVS. After making "clean opt" in pwlib and openh323 and make "clean install" in Asterisk i get an "Undefined symbol" error when I try to start Asterisk. As far as I can see its when loading the h323 channel driver the error occurs. Do I have to update other things as well, by reading the various

Hi, Anyone knows why there's a limit of 32 callgroup/pickupgroup in * ? It is coded as unsigned int but there's an hardcoded "if( X > 31 )" like line. IMHO, 32 group is very low and I wonder what impact it would have to increase it to 2^16-1 . Anyone?

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've discovered that SIP channels sometimes get stuck in *. I've read some posts from Fri 29 Aug 2003 which mentions this issue, but there doesn't seem to be any final answers I don't know if this is related to the 0001604 bug? Below is a list from one of the incidents: I know the (d) means that it is scheduled for destruction but the 10.1.1.45 channel hasn't

Hi! I have the following weird problem: phones A,B and C are in the same callgroup/pickupgroup. A call B, B is ringing, C calls *8. Now, B is CANCELed, C gets 200 OK, but A is still in Ringing. Is there anything else I have to configure? thanks Klaus

Hi everybody I've just installed a new Redhat 8.0 and configured it with Asterisk, zaptel and libpri. Afterwards I installed a TE410P and configured this as well. But when starting Asterisk I get the following error message: ------------------------------------------------------- -- Registered channel 1, PRI Signalling signalling ..... -- Registered channel 15, PRI Signalling

Hi I have an Asterisk installation with some SIP and MGPC devices, and I also have a TE410P on a E1 line. If I make an outside ISDN data call to asterisk the phone rings as usual and if I answer it, I just hear some clicks. I've read that the D-Channel has information about the call, if its voice or data. Is it somehow possible to end/ignore this call already before it is ringing?

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

Man, I really feel stupid, but after banging my head on a brick wall for several hours ... I need help! I've gotten the BLF to work on my shiny new GXP2010. The GXP is exten 5707, and I've got an xlite on 5608. When I make a call from an outside line, I dial SIP/5608. The little blinky light on the GXP that's monitoring 5608 goes, well, "blink blink". :) I then press

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