similar to: Transfer calls "on demand"

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

Ishfaq Malik <ish at pack-net.co.uk> schrieb: Hi Ishfaq > Look into Busy Lamp Field/Presence > > Here's a starting point: > > http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceStates-SECT-1.html Thanks a lot, but it does not seems to work... Here my configuration: sip.conf: [general] allowsubscribe=yes subscribecontext = default

Hi again! With the "call pickup"-function I can now pickup a call directed to another phone in my Asterisk. Very nice. My problem, now, is that I can't see on my phone, that the other phone (in another room) rings. Is it possible to signal the incoming call on other extension? I use two phones "Thomson ST2022". Thanks a lot Luca Bertoncello (lucabert at lucabert.de)

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

Ishfaq Malik <ish at pack-net.co.uk> schrieb: > The hints have to be in the same contexts in extensions.conf as defines in > the sip.conf subscribecontext which can be set per peer. Well, [anika_incoming] will be included in [default], of course... But I tried to define anika_incoming in subscribecontext, too. No changes... > Also, have you configured the phones as well? What do

Hi and happy new year! My question: - two extensions: 1111 and 2222 - an active call on 1111 - incoming calls to 1111 should be forwarded to 1111 (call advice!) and 2222 I know how can I forward an incoming call to more than an extension, but I have no idea how can I get the information, that 1111 has already an active call... I think, I need something like: exten =>

Hi list! I'm trying to configure Asterisk to record incoming calls, if the called press *3. I added in features.conf: automixmon => *3 then, in my dialplan: exten => 1,n,Dial(SIP/00493511111111,20,RcxX) Well, if I **CALL** a number I'm able to record the call, but if I'll be called, and press *3 nothing happens... In the console I can't see anything, too. Could you

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >

Hi list! I configured Asterisk to forward the incoming call for a number to both phones. I wrote that: exten => _00493512222222,n,Dial(SIP/00493512222222&SIP/00493511111111,,R) of course it works... Now the problem is, that when a phone get the call, on the other phone I get "1 missed call"... Is it possible to avoid that and signaling the other phone, that the call was

Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in

Zitat von Israel Gottlieb <isrlgb at gmail.com>: > If you the c option in the dial command it will send answered > else where sip message to the phone and most ip phones understand that > The cell will always display a missed call? I'm very sorry, but I can't understand what you mean... Could you explain, maybe with an example? Thanks Luca Bertoncello (lucabert at

Israel Gottlieb <isrlgb at gmail.com> schrieb: > At the end of the Command you could use options one of them is the c (not > apital) which sends a cancel event to the phone > http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Shalom Israel, unfortunately it does not work as expected... I wrote: exten => _00493512222222,n,Dial(SIP/00493512222222&SIP/00493511111111,,Rc)

On some SIP phones it is possible to turn off the missed call notifications, but I am not aware of a way to do the same on any cell phones. On 5 Jun 2015 07:29, "Mehmet Avcioglu" <mehmet at activecom.net> wrote: > > There is no signal that is sent to display a missed call. Your cell phone > does that. If it rings and is not answered it counts that as a miss. The > only

Hi list! So, new day, new problem... I tried right now to call from my cellphone a peer in my Asterisk. The cellphone has correct credentials, but it's NOT registered on my Asterisk, now. I just tried to call a peer in my network, from a peer not yet registered. And it works... :( The very curious thing is, that I can't find how the call will be accepted... Every section in my dialplan

#using "rollapply" to calculate a moving sum or running sum? #I am tryign to use rollapply to calcualte a moving sum? #I tried rollapply and get the error message #"Error in seq.default(start.at, NROW(data), by = by) : # wrong sign in 'by' argument" #example: mymatrix <- ( matrix(data=1:100, nrow=5, ncol=20) ) mymatrix_cumsum <- ( matrix(data=NA, nrow=5,

Hi list! I deleted completely Samba on my Server and reinstalled it again. Now it seems to work, but I have a little problem: if I try to access the home directory of a user and write something, the owner of the created directory/file is a UID, but not the user itself... Could someone help me? This is the definition of the share in my smb.conf: [homes] comment = Home Directories

This is correct >>that gpupdate tries to copy somethings from \\cch.intra\sysvol and not from \\dc1\sysvol... >>There a no server with name cch.intra, this is just the Realm... No not REALM, but DNSdomain but with the same name as the REALM.   You “should” be able to “ping cch.intra” or browse to \\cch.intra   if not, then your missing dns records.     If you have only