similar to: Forwarding call if extension busy

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 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

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I must be missing something here but I can't figure out why I can't get DEVICE_STATE() to give me anything other than "NOT_INUSE". I have two extensions: 6666 and 6668. I used 6668 to make a call to yet another phone, so I know that it's busy. I then use 6666 to call 6668 and in the dialplan have a noop to see what

Is there a way skip / ignore the member whose status is busy in the Queue. I have two channel member in queue and i have set the peer limit 2 for these members. I want to skip those member who are currently on the call (answered to calls) and now their status is busy, if Queue see the busy status caller will not enter in the Queue and go to the next priority. [test-queue] strategy = rrmemory

Hi list, I need your help ,I have an incoming call x the ivr and the operator takes the call. ext "101" , If a second call reenters and the operator is talking, I want to send to the extension 102 I use the Variable DIALSTATUS , but not working check IVR [IVRINMA] exten => s,1,Wait(1) exten => s,n,Set(CHANNEL(language)=es) same=> n,Set(TIMEOUT(digit)=4) same=>

On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan <mjordan at digium.com> wrote: > On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna <jlamanna at gmail.com> wrote: > > Also, how big does the cache in frame.c grow to? > > I've recompiled with MALLOC_DEBUG on that server: > > > > asterisk -rx "memory show summary" > > > > .... > >

Hi Is there any way to set the presence state of a peer to in-use in asterisk 1.8? The idea is to integrate DND buttons on phones to BLF. Regards -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street

Hi list! Right now I configured my Asterisk to forward the calls for the number X to both phones (mine and the phone of my wife). It works, of course, but I'm not enthusiast... I see what we have at office: if one phone rings, other phones in the same group can "catch the call", so that if a colleague is not present, another colleague can catch the call. I'd like to have the

Hi! Trying to figure out what I am doing wrong... 1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256) 1 Cell phone 00733025975 attached through H323. extensions.conf exten => 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1) exten => 975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 975-INUSE,2,Hangup() exten =>

Hi all! I am trying to figure out how DEVICE_STATE is working, no luck so far. sip.conf [0317998975] type=friend regexten=0317998975 secret=???? username=0317998975 callerid="Magnus Benngard" mailbox=0317998975 at inputinterior.se host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes disallow=all allow=alaw extensions.conf exten => 0317998975,hint,SIP/0317998975 exten =>

Also, how big does the cache in frame.c grow to? I've recompiled with MALLOC_DEBUG on that server: asterisk -rx "memory show summary" .... 1780466242 bytes (1780181594 cache) in 2352909 allocations in file frame.c ... Seems like a ridiculous cache. On Mon, Nov 24, 2014 at 9:02 AM, James Lamanna <jlamanna at gmail.com> wrote: > cat /proc/cpuinfo lists 4 cores. >

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)

Hi all, How can I implement a full-featured Call-Waiting behavior on the Asterisk level (e.g. I don't want to relay on end-equipment capabilities)? I found it very strange that such a basic feature is not built-in in Asterisk (and I've googled a lot in search for this). Here is what I need: SomeuserX is calling MyUserA. They are on conversation (assumption: voice is via the Asterisk)

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 Snom870 Handsets We are in the process of moving to an Asterisk based PBX. At the moment most things work as we wish. However, I have just notices that when I force a reload using 'amportal a reload' I see this loop start in 'asterisk -rvvvvvvvvvv': > Channel Local/s at tc-maint-000002a4;1

Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside

First off this is not my work for extensions.conf it is modified from http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbackl ogin-to-standard-dialplan-methods-part-1/ So credit to Leif Madsen <http://www.leifmadsen.com> But as to my question [AgentLogin] ;A replaced version of AgentCallbackLogin() using a GoSub() ; exten =>

Hi everybody Did anybody by any chance ever work out how to log in and out agents on Asterisk 6+? I used to have it working perfect in Asterisk 1.2 but since I upgraded to 6 the agent login functions are gone and the readme file that came with it made no sense to me. I noticed somebody on the net posted that they had the same problem but used Voicemail to authenticate users, but that seemed a

Hi, Here http://www.voip-info.org/wiki/view/Asterisk+func+device_State you can find a link to download a backported for Asterisk 1.4 version of DEVICE_STATE function. (Elsewhere, you can find reference to another backported function DEVSTATE which seems to behave the same as DEVICE_STATE). As I would like to prepare as much as possible, my dialplan to 1.6 and beyond, I would prefer to use

In testing Asterisk 11, I've found that Asterisk doesn't seem to be sending BLF updates to SIP peers that have subscribed to a hint looking at a Custom device if that Custom device state is RINGING or RING_INUSE. All other states seem to be working correctly. The hint section of the dialplan is: [hints] exten => _3XX,hint,Custom:${EXTEN} Console shows the following for core show