[Asterisk-Users] Call Pickup
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. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup ???
I was wondering if you can pick up a ringing channel by dialing *8# when you and the other phones are in pickupgroup. Could you do something to the effect of If the caller was put on a certain extension and just sitting there... Could you grab the caller by doing something like *8 where the caller is sitting? Here is my issue. I need to find a way to put the caller on hold and play MOH. I don't want to use park unless I can park them on a given extension. And if I use flash the user gets a dial tone. He can either transfer to another user at that point, and if he hangs up the line rings back until he picks up. Any ideas ?? -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call pickup
Hi
I am using asterisk 0.7.1
I am testing with 3 SIP phone.
Phone A call to Phon B , and Phone B is Ringing.
I want to pickup that call , So I press '*8' for pickup the call on Phone
C.
But I can not pickup the call.
I can see "NOTICE[6151]:chan_sip.c:5198 handle_requst: Nothing to pick up"
in console.
;== sip.conf
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
;bindaddr = 61.36.179.152 ; Address to bind to
bindaddr = 0.0.0.0
;externip = 200.201.202.203 ; Address that we're going to put in SIP
messages if we're behind a NAT
;localnet = 61.36.179.0 ; Internal NETWORK address
;localmask = 255.255.255.128 ; Internal netmask
;context = default ; Default for incoming calls
context = from-sip
;srvlookup = yes ; Enable SRV lookups on outbound calls
;pedantic = yes ; Enable slow, pedantic checking for Pingtel
;tos=lowdelay
;tos=184
maxexpirey=3600 ; Max length of incoming registration we allow
defaultexpirey=160 ; Default length of incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
;videosupport=yes ; Turn on support for SIP video
;disallow=all ; Disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc
;
register => [EMAIL PROTECTED] ; Register with a SIP provider
;register => [EMAIL PROTECTED]/1234 ; Register 2345 at sip provider as
1234 here.
;
callgroup=1
pickupgroup=1
[hst239]
type=friend
secret=young
dtmfmode=inband
host=61.36.179.239
threewaycall = yes
callgroup=1
pickupgroup=1
context=from-sip
[hst220]
type=friend
host=61.36.179.220
callgroup=1
pickupgroup=1
threewaycall = yes
context=from-sip
[hst238]
type=friend
host=61.36.179.238
dtmfmode=inband
callgroup=1
pickupgroup=1
threewaycall = yes
context=from-sip
[hst155]
type=friend
host=210.98.251.155
callgroup=1
pickupgroup=1
threewaycall = yes
context=sip-from
[61.36.179.167]
type=friend
username=9002000
host=61.36.179.167
callgroup=1
pickupgroup=1
context=from-sip
= extensions.conf ===
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
; You can include other config files, use the #include command (without the
';')
; Note that this is different from the "include" command that includes
contexts within
; other contexts. The #include command works in all asterisk configuration
files.
;#include "filename.conf"
; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
[iaxtel700]
exten => _91700NXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
;
; International long distance through trunk
;
exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9011.,2,Congestion
[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91NXXNXX,2,Congestion
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9NXX,2,Congestion
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91800NXX,2,Congestion
exten => _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXX,2,Congestion
exten => _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXX,2,Congestion
exten => _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXX,2,Congestion
[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
exten => 6601,1,WaitMusicOnHold(30) ; hur
;
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote
; IAX switching you transparently get access to the remote
;
; switch => IAX2/user:[EMAIL PROTECTED]/local
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MAC
[Asterisk-Users] Call Pickup
Hi, I put a pickupgroup=0 line for each user in sip.conf. After restarting asterisk I called my collegues phone with my cell phone, I heard it ringing and saw "ringing" in the asterisk console. Then I dialed *8 with my phone and got on the console: Jan 24 20:41:45 NOTICE[13747]: chan_sip.c:7321 handle_request: Nothing to pick up -- SIP/collegue-92e5 is ringing while the other phone kept ringing. I'm using asterisk-1.0.3 What went wrong? Thanks for any hints! Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup
On my present phone system I can "pickup" a call that is ringing on another phone. How do I do this with asterisk? I searched on the wiki for pickup and did not find anything. Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call pickup
I don't understand how to get call pickup to work with asterisk. Have I to define *8 extension in the dialplan? to what? Have I to include something, like for parked call? Has the stable 1.0.2 version the pickup group feature? or I need to patch it with bristuff? Thank you Leandro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call pickup
hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? thanks -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup
You need to have a pending call in the system (some extensions that is ringing to test that). If you have 3 FXS ports try to place a call from the first one to the 2nd and then instead of taking the 2nd off hook dial *8 on the 3rd phone Martin On Thu, 17 Jul 2003, Jay Tyndall wrote: > 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. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call pickup problem
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 auto-generated by the Parkinglot Module, do ; not hand edit that file #include parking_additional.inc #include features_general_custom.conf [applicationmap] #include features_applicationmap_additional.conf ; *** IMPORTANT NOTE *** ; The original blindxfer was '#', and has been changed to '##' to avoid ; issues with sending DTMF '#' to remote parties. [featuremap] blindxfer => ##; Blind Transfer disconnect => **; Disconnect Call automon => *1; One Touch Record ;atxfer => *2; Attended Xfer please tell the right steps for make it working thank you ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call pickup (group)
Hi, Asterisk has a feature called pickupgroup, meaning you can pickup the call that is ringing on your collegues phone. Can this type of behaviour be emulated in extension logic or AGI (maybe together with manager login) ? We need the group settings to be tied into a database which makes it a little more dynamic :-> Any suggestions are welcome. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup Problem
I'm having a problem with the call pickup with the latest CVS. Before I updated to the latest CVS it was working fine. Now, whenever anyone tries to pickup a call using *8 it dumps all calls going on at the time and hangs up on the incoming call. Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call pickup fails.
Hello all. I saw a few weeks ago a discussion about cal pickup, *8, not working but did not find a message about it being resolved, I look for a bug on the bug list but did not find anything about it not working, nor a bug open. I installed asterisk 0.9.0, have one sip fxo gateway and only sip phones, all of them have callgroup=1 and pickupgroup=1 but I can not get a call that is ringing in another phone, there is a message on the * console that says something like "Nothing to pickup" every time I try it. Any hints ? Thank's. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup
You also need callgroup=0 in the sip.conf per user as well. callgroup = the group this sip entry belongs to pickupgroup = the group(s) this sip entry is allowed to pickup Julian. Roger Schreiter wrote: Hi, I put a pickupgroup=0 line for each user in sip.conf. After restarting asterisk I called my collegues phone with my cell phone, I heard it ringing and saw "ringing" in the asterisk console. Then I dialed *8 with my phone and got on the console: Jan 24 20:41:45 NOTICE[13747]: chan_sip.c:7321 handle_request: Nothing to pick up -- SIP/collegue-92e5 is ringing while the other phone kept ringing. I'm using asterisk-1.0.3 What went wrong? Thanks for any hints! Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup
Roger Schreiter wrote: Hi, I put a pickupgroup=0 line for each user in sip.conf. As far as I'm aware, pickup groups are only for zap interfaces... -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup
Matt, they work fine on zap and sip. I wish they worked on IAX. Ernie On Jan 24, 2005, at 12:20 PM, Matt Riddell wrote: Roger Schreiter wrote: Hi, I put a pickupgroup=0 line for each user in sip.conf. As far as I'm aware, pickup groups are only for zap interfaces... -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup
On 24 Jan 2005, at 19:20, Matt Riddell wrote: Roger Schreiter wrote: Hi, I put a pickupgroup=0 line for each user in sip.conf. As far as I'm aware, pickup groups are only for zap interfaces... No, I have pickup groups working for SIP devices. As a simple thing, shouldn't the numbering for the groups start from 1? Try changing it to pickupgroup=1, thats how I have it defined for my SIP phones (Sipuras / Xlites) Phil. -- Phil Quinney IT Consultant - Any-Ideas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup
We're using them on Cisco 79XX phones without any problems, although we are using CVS-HEAD. The wiki for features.conf does mention SIP call pickup. Julian. Matt Riddell wrote: Roger Schreiter wrote: Hi, I put a pickupgroup=0 line for each user in sip.conf. As far as I'm aware, pickup groups are only for zap interfaces... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup
On 24 Jan 2005, at 19:20, Matt Riddell wrote: Roger Schreiter wrote: Hi, I put a pickupgroup=0 line for each user in sip.conf. As far as I'm aware, pickup groups are only for zap interfaces... No, I have pickup groups working for SIP devices. As a simple thing, shouldn't the numbering for the groups start from 1? Try changing it to pickupgroup=1, thats how I have it defined for my SIP phones (Sipuras / Xlites) Phil. (Apologies if this turns out as a double post...) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Pickup
I have a rather long dial plan, but it includes support for call waiting.
However, the setgroup checkgroup commands don't seem to be working. Can
anyone help on this one?
Excerpts are below. First exten-vm is dialed and then dial-new.
As I understand, priority 1 increments the active channels for the caller
and then in "dial-new" priority 8 increments for Arg3, or the Callee
extension. Problem is, that priority 9 always goes on to 10 (i.e. group
never is "on-the-phone".
Am I missing something?
When ext201 dials 202, CLI shows:
-- Executing Macro("SIP/201-8571", "exten-vm|202|202") in new stack
-- Executing SetGroup("SIP/201-8571", "201") in new stack
-- Executing SetMusicOnHold("SIP/201-8571", "default") in new stack
-- Executing SetVar("SIP/201-8571", "FROMCONTEXT=exten-vm") in new stack
-- Executing GotoIf("SIP/201-8571", "0?9:5") in new stack
-- Goto (macro-exten-vm,s,5)
-- Executing Macro("SIP/201-8571", "dial-new|15|tr|202|202") in new
stack
-- Executing DBget("SIP/201-8571", "CallForwardIm=CF/202") in new stack
-- DBget: varname=CallForwardIm, family=CF, key=202
-- DBget: Value not found in database.
-- Executing Goto("SIP/201-8571", "s|4") in new stack
-- Goto (macro-dial-new,s,4)
-- Executing DBget("SIP/201-8571", "DNDStatus=DND/202") in new stack
-- DBget: varname=DNDStatus, family=DND, key=202
-- DBget: Value not found in database.
-- Executing Goto("SIP/201-8571", "s|8") in new stack
-- Goto (macro-dial-new,s,8)
-- Executing SetGroup("SIP/201-8571", "202") in new stack
I'll be most grateful for any assistance.
Thanks
Mike
[macro-exten-vm]
exten => s,1,SetGroup(${CALLERIDNUM})
exten => s,2,SetMusicOnHold(default)
exten => s,3,Setvar(FROMCONTEXT=exten-vm) exten =>
s,4,GotoIf($[${CALLERIDNUM} = ${ARG2}]?9:5) ;check self-voicemail exten =>
s,5,Macro(dial-new,${RINGTIMER},${DIAL_OPTIONS},${ARG2},${ARG1})
[macro-dial-new]
;now check if destination is on a call
exten => s,8,SetGroup(${ARG3})
exten => s,9,CheckGroup(1)
;go to 110 then 25 if on the phone (CW handler), go to 10 if not on the
phone
exten => s,110,Goto(s,25)
;line is clear, begin dial sequence
exten => s,10,Setvar(ChanType=${E${ARG3}}) ;Get the channel type
exten => s,11,Dial(${ChanType}/${ARG3},${ARG1},${ARG2})
Mike Sander
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Re: [Asterisk-Users] Call Pickup
On Thu, 2004-11-11 at 19:57 -0500, Jerry Geis wrote: > On my present phone system I can "pickup" a call that is ringing on another > phone. > > How do I do this with asterisk? I searched on the wiki for pickup > and did not find anything. pickupgroups/callgroups -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup
On Thu, Nov 11, 2004 at 07:57:11PM -0500, Jerry Geis said: > On my present phone system I can "pickup" a call that is ringing on another > phone. > > How do I do this with asterisk? I searched on the wiki for pickup > and did not find anything. Hmm. I just did a search on "call pickup" on the wiki and it had 547 results. The first hit mentioned *8 in Asterisk. The second hit showed how to configure groups. The third was features.conf which shows that you can change *8 to some other code. Are you looking at the right wiki http://www.voip-info.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup
On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said: > I don't understand how to get call pickup to work with asterisk. > Have I to define *8 extension in the dialplan? to what? > Have I to include something, like for parked call? > Has the stable 1.0.2 version the pickup group feature? > or I need to patch it with bristuff? Search the wiki for call pickup. It's all there. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup
- Original Message - From: "Walt Reed" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Tuesday, November 16, 2004 1:04 PM Subject: Re: [Asterisk-Users] Call pickup > > On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said: > > I don't understand how to get call pickup to work with asterisk. > > Have I to define *8 extension in the dialplan? to what? > > Have I to include something, like for parked call? > > Has the stable 1.0.2 version the pickup group feature? > > or I need to patch it with bristuff? > > Search the wiki for call pickup. It's all there. Unfortunately I have already read all the readable on wiki without understanding the needed steps to get call pickup to work. Can you please answer my questions? Thank you Leandro > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup
On Tue, Nov 16, 2004 at 01:26:22PM +0100, Leandro said: > From: "Walt Reed" <[EMAIL PROTECTED]> > > On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said: > > > I don't understand how to get call pickup to work with asterisk. > > > Have I to define *8 extension in the dialplan? to what? > > > Have I to include something, like for parked call? > > > Has the stable 1.0.2 version the pickup group feature? > > > or I need to patch it with bristuff? > > > > Search the wiki for call pickup. It's all there. > > Unfortunately I have already read all the readable on wiki without > understanding the needed steps to get call pickup to work. Can you please > answer my questions? What particular part do you not understand? The first search result hit describes call pickup in general. The second describes how to create pickup groups. You need to do this. The third shows where *8 is defined and that you can change it to something else. *8 has been built-into asterisk for a very long time. In 1.0.2 you can change it to some other code. That's it. Once you have defined your groups for all the different channels you have (SIP, Zap, IAX, etc.), it just works. If you have problems, you will need to give detailed information on how you have your groups set in all the various channels involved, log examples, etc. Make sure you look at the example configuration files that come with asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup
> > On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said: > > > I don't understand how to get call pickup to work with asterisk. > > > Have I to define *8 extension in the dialplan? to what? > > > Have I to include something, like for parked call? > > > Has the stable 1.0.2 version the pickup group feature? > > > or I need to patch it with bristuff? > > > > Search the wiki for call pickup. It's all there. > > Unfortunately I have already read all the readable on wiki without > understanding the needed steps to get call pickup to work. Can you please > answer my questions? It really isn't that hard. Here's an example. In zapata.conf, an entry might look like: context-inbound-bus signalling=fxs_ks callgroup=2 channel => 1 In sip.conf, an phone entry might look like: [3002] type= username=3002 secret= pickupgroup=2 Since the above reflects a zap interface was assigned to callgroup=2, the sip phone with pickupgroup=2 "can" pick that ringing call up by pressing *8 (or *8#). If a different sip phone is defined with pickupgroup=17, it would not be able to get callgroup=2 assignments. To take that a step further, you could also have a sip.conf entry like: [3004] type= username=3004 secret= pickupgroup=2 callgroup=2 and whenever x3004 is ringing, the sip phone at 3002 can pick that ringing call up as well as the zap interface noted above. If both are ringing at exactly the same time, I'm not sure which will be picked up, but one of them will be. On my sip phone (Cisco 7960) I have to use *8# to pickup calls. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup
- Original Message -
From:
Walt
Reed
To: Leandro
Cc: Walt Reed ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, November 16, 2004 2:11
PM
Subject: Re: [Asterisk-Users] Call
pickup
On Tue, Nov 16, 2004 at 01:26:22PM +0100, Leandro said:> From:
"Walt Reed" <[EMAIL PROTECTED]>> >
On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said:> > > I
don't understand how to get call pickup to work with asterisk.> >
> Have I to define *8 extension in the dialplan? to what?> > >
Have I to include something, like for parked call?> > > Has the
stable 1.0.2 version the pickup group feature?> > > or I need to
patch it with bristuff?> >> > Search the wiki for call
pickup. It's all there.> > Unfortunately I have already read all
the readable on wiki without> understanding the needed steps to get
call pickup to work. Can you please> answer my questions?What
particular part do you not understand?The first search result hit
describes call pickup in general.The second describes how to create
pickup groups. You need to do this.The third shows where *8 is defined
and that you can change it tosomething else. *8 has been built-into
asterisk for a very long time. In1.0.2 you can change it to some other
code.That's it. Once you have defined your groups for all the
differentchannels you have (SIP, Zap, IAX, etc.), it just works. If you
haveproblems, you will need to give detailed information on how you
haveyour groups set in all the various channels involved, log examples,
etc.Make sure you look at the example configuration files that come
withasterisk.
I really hate to ask silly questions and thank you
for your time, but pickup group doesn't work yet. Maybe the pickup doesn't work
across Zap channels.
This is what I get when Zap/25 is ringing Zap/14
and Zap/7 try to pickup. I get "invalid extension" when I press *8#
- Starting simple switch on
'Zap/25-1' -- Executing Answer("Zap/25-1", "") in new
stack -- Executing Dial("Zap/25-1", "Zap/14") in new
stack -- Called 14 -- Zap/14-1 is
ringing -- Executing DigitTimeout("Zap/7-1", "3") in new
stack -- Set Digit Timeout to 3 --
Executing ResponseTimeout("Zap/7-1", "10") in new stack --
Set Response Timeout to 10 -- Zap/14-1 is
ringing -- Invalid extension '*' in context 'interno' on
Zap/7-1 == CDR updated on Zap/7-1 -- Executing
Playtones("Zap/7-1", "Congestion") in new stack -- Invalid
extension '8' in context 'interno' on Zap/7-1 == CDR updated on
Zap/7-1 -- Executing Playtones("Zap/7-1", "Congestion") in
new stack -- Invalid extension '#' in context 'interno' on
Zap/7-1 == CDR updated on Zap/7-1 -- Executing
Playtones("Zap/7-1", "Congestion") in new stack --
Zap/14-1 is ringing -- Hungup 'Zap/7-1'
This is my /etc/asterisk/zapata.conf
context=internosignalling=fxo_lsflash=100group=1callgroup=5pickupgroup=5channel
=> 1-24
context=pstnsignalling=fxs_kscallgroup=5pickupgroup=5group=2channel
=> 25
context=voipsignalling=fxs_kscallgroup=5pickupgroup=5group=3channel
=> 26
This is the dialplan
[interno]include => parkedcalls
exten => t,1,Hangupexten =>
i,1,Playtones(Congestion)
exten => s,1,DigitTimeout,3
exten => s,2,ResponseTimeout,10
exten =>
4,1,Goto(componiinternoserie4,s,1)exten =>
5,1,Goto(componiinternoserie5,s,1)exten =>
6,1,Goto(componiinternoserie6,s,1)
exten => 0,1,Goto(impegnolinea,s,1)
exten =>
3001,1,MusicOnHold()
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RE: [Asterisk-Users] Call pickup
Hi,
Have you configured features.conf file? the line which
enabled call pickup is commented and you have to un comment the line for call
pickup to work. Also you can define the numbering for call pickup
there
Thanks.
Yusuf
Alakavuk
Teknik Danışman - Technical
Consultant
Grid Bilişim
Teknolojileri A.Ş.
Kuştepe Mahallesi Leylak
Sokak
Murat İş Merkezi A Blok Kat:2
Daire:9
34387 Şişli İstanbul
Türkiye
Tel :
+90 (212) 336 92 55
Fax : +90
(212) 266 25 50
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
LeandroSent: 19 Kasım 2004 Cuma 17:52To: Walt Reed;
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Call pickup
- Original Message -
From:
Walt
Reed
To: Leandro
Cc: Walt Reed ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, November 16, 2004 2:11
PM
Subject: Re: [Asterisk-Users] Call
pickup
On Tue, Nov 16, 2004 at 01:26:22PM +0100, Leandro said:> From:
"Walt Reed" <[EMAIL PROTECTED]>> >
On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said:> > > I
don't understand how to get call pickup to work with asterisk.> >
> Have I to define *8 extension in the dialplan? to what?> > >
Have I to include something, like for parked call?> > > Has the
stable 1.0.2 version the pickup group feature?> > > or I need to
patch it with bristuff?> >> > Search the wiki for call
pickup. It's all there.> > Unfortunately I have already read all
the readable on wiki without> understanding the needed steps to get
call pickup to work. Can you please> answer my questions?What
particular part do you not understand?The first search result hit
describes call pickup in general.The second describes how to create
pickup groups. You need to do this.The third shows where *8 is defined
and that you can change it tosomething else. *8 has been built-into
asterisk for a very long time. In1.0.2 you can change it to some other
code.That's it. Once you have defined your groups for all the
differentchannels you have (SIP, Zap, IAX, etc.), it just works. If you
haveproblems, you will need to give detailed information on how you
haveyour groups set in all the various channels involved, log examples,
etc.Make sure you look at the example configuration files that come
withasterisk.
I really hate to ask silly questions and thank you
for your time, but pickup group doesn't work yet. Maybe the pickup doesn't work
across Zap channels.
This is what I get when Zap/25 is ringing Zap/14
and Zap/7 try to pickup. I get "invalid extension" when I press *8#
- Starting simple switch on
'Zap/25-1' -- Executing Answer("Zap/25-1", "") in new
stack -- Executing Dial("Zap/25-1", "Zap/14") in new
stack -- Called 14 -- Zap/14-1 is
ringing -- Executing DigitTimeout("Zap/7-1", "3") in new
stack -- Set Digit Timeout to 3 --
Executing ResponseTimeout("Zap/7-1", "10") in new stack --
Set Response Timeout to 10 -- Zap/14-1 is
ringing -- Invalid extension '*' in context 'interno' on
Zap/7-1 == CDR updated on Zap/7-1 -- Executing
Playtones("Zap/7-1", "Congestion") in new stack -- Invalid
extension '8' in context 'interno' on Zap/7-1 == CDR updated on
Zap/7-1 -- Executing Playtones("Zap/7-1", "Congestion") in
new stack -- Invalid extension '#' in context 'interno' on
Zap/7-1 == CDR updated on Zap/7-1 -- Executing
Playtones("Zap/7-1", "Congestion") in new stack --
Zap/14-1 is ringing -- Hungup 'Zap/7-1'
This is my /etc/asterisk/zapata.conf
context=internosignalling=fxo_lsflash=100group=1callgroup=5pickupgroup=5channel
=> 1-24
context=pstnsignalling=fxs_kscallgroup=5pickupgroup=5group=2channel
=> 25
context=voipsignalling=fxs_kscallgroup=5pickupgroup=5group=3channel
=> 26
This is the dialplan
[interno]include => parkedcalls
exten => t,1,Hangupexten =>
i,1,Playtones(Congestion)
exten => s,1,DigitTimeout,3
exten => s,2,ResponseTimeout,10
exten =>
4,1,Goto(componiinternoserie4,s,1)exten =>
5,1,Goto(componiinternoserie5,s,1)exten =>
6,1,Goto(componiinternoserie6,s,1)
exten => 0,1,Goto(impegnolinea,s,1)
exten =>
3001,1,MusicOnHold()
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Re: [Asterisk-Users] Call pickup
- Original Message - From: Yusuf Alakavuk To: 'Asterisk Users Mailing List - Non-Commercial Discussion' ; 'Walt Reed' Sent: Friday, November 19, 2004 5:02 PM Subject: RE: [Asterisk-Users] Call pickup Hi, Have you configured features.conf file? the line which enabled call pickup is commented and you have to un comment the line for call pickup to work. Also you can define the numbering for call pickup there Are you referring to pickupexten=*8? Thank you for your try, but unfortunately, I have already uncommented it in features.conf :-( ;; Sample Parking configuration; [general]parkext => 700 ; What ext. to dial to parkparkpos => 701-720 ; What extensions to park calls oncontext => parkedcalls ; Which context parked calls are in;parkingtime => 45 ; Number of seconds a call can be parked for ; (default is 45 seconds);transferdigittimeout => 3 ; Number of seconds to wait between digits when transfering a call;courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call;adsipark = yes ; if you want ADSI parking announcements pickupexten = *8 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup issue
I know this has been discussed heavily but i have a bizzare issue with call pickup. I have 3 asterisk servers all built the same on centos 4.1 and call pickup works on two of them but not on the third. They have identical configurations. I'm using asterisk 1-0-9 and zaptel with ztdummy. All phones are sip with a mix of sip clients and cisco 7960's. All i get in the asterisk debug is - Sep 22 09:32:50 NOTICE[9562]: chan_sip.c:7427 handle_request: Nothing to pick up Anyone offer any ideas? Ta! ___ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup Woes
Hello all, I have an asterisk @ home system running 1.2.4. Call pickup seems to be a bit of a problem. I’ve looked at a lot of posts and the wiki, which states that you need to define the pickup extension in features.conf and the pickup groups in sip.conf. I’ve done this, however there is no definition for *8 in extensions.conf. Is there supposed to be and it has been removed? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call pickup
Am 05.10.2011 20:42, schrieb Marek Cervenka: hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? You can have that with subscriptions/hints, for example Snom phones can display not only a call to one of the peers but also the caller and callee identification. This works jaw to cheek with BLF (busy lamp field) which allows to monitor other extensions' status (in_use, ringing...). Of course you can be member of a pickup group without "monitoring" the status of any of the peers, and you can monitor a peer's status without being in the same pickup group (although not pickup the call then, obviously :-) Regards Martin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call pickup
On 10/05/2011 11:21 PM, A. M. Hoffmeister wrote: > Am 05.10.2011 20:42, schrieb Marek Cervenka: >> hello, >> >> is there some way to notify people in the same pickup group about call >> from caller to callee? >> >> i.e. i have call from 111 to 222 >> there are 222,333,444 in the same pickup group >> >> 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup >> the call with *8 >> >> siemens have this on their sip openstage phones. how they do this? > > You can have that with subscriptions/hints, for example Snom phones > can display not only a call to one of the peers but also the caller > and callee > identification. > can you point me to some doc/examples? how this is implemented in SIP? i think about sending some notify from dialplan (i have incoming call, i know who is in pickup group, i can send call to callee and before send some "NOTIFY" to other phones in the pickupgroup) i found only one app like this - jabbersend. but i need this notification on phone screen > This works jaw to cheek with BLF (busy lamp field) which allows to > monitor > other extensions' status (in_use, ringing...). > > Of course you can be member of a pickup group without "monitoring" the > status of any of the peers, and you can monitor a peer's status without > being in the same pickup group (although not pickup the call then, > obviously :-) > -- --- Marek Cervenka Centrum Vypocetni Techniky jabber - cerv...@njs.netlab.cz CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz RHCE 100-175-678 === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call pickup
Search for dialog-info pickup -Original Message- From: Marek Cervenka Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 07 Oct 2011 09:47:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call pickup On 10/05/2011 11:21 PM, A. M. Hoffmeister wrote: > Am 05.10.2011 20:42, schrieb Marek Cervenka: >> hello, >> >> is there some way to notify people in the same pickup group about call >> from caller to callee? >> >> i.e. i have call from 111 to 222 >> there are 222,333,444 in the same pickup group >> >> 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup >> the call with *8 >> >> siemens have this on their sip openstage phones. how they do this? > > You can have that with subscriptions/hints, for example Snom phones > can display not only a call to one of the peers but also the caller > and callee > identification. > can you point me to some doc/examples? how this is implemented in SIP? i think about sending some notify from dialplan (i have incoming call, i know who is in pickup group, i can send call to callee and before send some "NOTIFY" to other phones in the pickupgroup) i found only one app like this - jabbersend. but i need this notification on phone screen > This works jaw to cheek with BLF (busy lamp field) which allows to > monitor > other extensions' status (in_use, ringing...). > > Of course you can be member of a pickup group without "monitoring" the > status of any of the peers, and you can monitor a peer's status without > being in the same pickup group (although not pickup the call then, > obviously :-) > -- --- Marek Cervenka Centrum Vypocetni Techniky jabber - cerv...@njs.netlab.cz CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz RHCE 100-175-678 === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call pickup -> Change shortcode
Hello Is it possible, (without hacking the source), to change the code for call pickup because my SIP gateways uses * key as End-Of-Dial. If I have to hack the source can somebody tell me where to look? Mvh Mickey Binder Comflex A/S Tlf.: 43997102 Ë^®+$RÇ«²f¢)à+-Ë^®+$RÇ«²X¬¶Çb+¦r¡¶ÚþX¬¶Çb+¦r¿¨¥©ÿ+-wèý«-z¸¬ë ®
[asterisk-users] call pickup - Asterisk 1.4.19.1 -
Hello List, Does anyone here have call pickup (with *8 ) working ok on Asterisk version 1.4.19.1 ? Thanks in advice, -- Jose P. Espinal Slackware-Es.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call pickup with IAX
Hi all Somebody know with IAX support pickup call feature in the last 1.4 .X asterisk release ? With SIP I use features.conf and works fine, but no way to make works with IAX. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call pickup fails.
> I saw a few weeks ago a discussion about cal pickup, *8, not working > but did not find a message about it being resolved, I look for a bug on > the bug list but did not find anything about it not working, nor a bug open. > I installed asterisk 0.9.0, have one sip fxo gateway and only sip > phones, all of them have callgroup=1 and pickupgroup=1 but I can not get > a call that is ringing in another phone, there is a message on the * > console that says something like "Nothing to pickup" every time I try it. > Any hints ? It's been working fine for me on cvs Head for months. We have to use *8# from a sip phone however. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup Woes
There shouldn't be one, have you tried it? what is the CLI output? On 3/19/06, Adam Dale <[EMAIL PROTECTED]> wrote: > > > > Hello all, > > > > I have an asterisk @ home system running 1.2.4. Call pickup seems to be a > bit of a problem. I've looked at a lot of posts and the wiki, which states > that you need to define the pickup extension in features.conf and the pickup > groups in sip.conf. I've done this, however there is no definition for *8 in > extensions.conf. > > > > Is there supposed to be and it has been removed? > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup Woes
C F wrote: groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Its not in extensions.conf, its in features.conf -- in extensions.conf you have to configure callgroups for each of your extensions, so that you can pick them up with *8. -- National Manager - Special Projects < Sydney / Melbourne / Canberra / Hobart / London /> 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ .>> Open Source - Own it - Squiz.net ./> ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Pickup Woes
I've configured the following in features.conf pickupexten = *8 ; Configure the pickup extension. Default is *8 and all SIP extensions are configured as pickupgroup=1. These phones can make and receive calls, and also use features such as *69, *70 and *98. When I dial *8 I get a beeping as if there is no valid extension and no debugging information when I open the console with asterisk -vvvr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Monday, 20 March 2006 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: >>groups in sip.conf. I've done this, however there is no definition for *8 in >>extensions.conf. Its not in extensions.conf, its in features.conf -- in extensions.conf you have to configure callgroups for each of your extensions, so that you can pick them up with *8. -- National Manager - Special Projects < Sydney / Melbourne / Canberra / Hobart / London /> 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ .>> Open Source - Own it - Squiz.net ./> ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup Woes
You have to configre the Dialplan in your sip phone to accept *8 What phone are you using? On 3/19/06, Adam Dale <[EMAIL PROTECTED]> wrote: > I've configured the following in features.conf > > pickupexten = *8 ; Configure the pickup extension. Default is *8 > > and all SIP extensions are configured as pickupgroup=1. > > These phones can make and receive calls, and also use features such as *69, > *70 and *98. > > When I dial *8 I get a beeping as if there is no valid extension and no > debugging information when I open the console with asterisk -vvvr > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller > Sent: Monday, 20 March 2006 9:51 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Call Pickup Woes > > C F wrote: > >>groups in sip.conf. I've done this, however there is no definition for *8 > in > >>extensions.conf. > > Its not in extensions.conf, its in features.conf -- in extensions.conf > you have to configure callgroups for each of your extensions, so that > you can pick them up with *8. > > -- > National Manager - Special Projects > > < Sydney / Melbourne / Canberra / Hobart / London /> >2/340 Gore Street T: +61 (0) 3 9486 0411 >Fitzroy, VIC F: +61 (0) 3 9486 0611 >3065 W: http://www.squiz.net/ > > .>> Open Source - Own it - Squiz.net ./> > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Pickup Woes
I am using Cisco 7940/60/70's -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 20 March 2006 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes You have to configre the Dialplan in your sip phone to accept *8 What phone are you using? On 3/19/06, Adam Dale <[EMAIL PROTECTED]> wrote: > I've configured the following in features.conf > > pickupexten = *8 ; Configure the pickup extension. Default is *8 > > and all SIP extensions are configured as pickupgroup=1. > > These phones can make and receive calls, and also use features such as *69, > *70 and *98. > > When I dial *8 I get a beeping as if there is no valid extension and no > debugging information when I open the console with asterisk -vvvr > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller > Sent: Monday, 20 March 2006 9:51 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Call Pickup Woes > > C F wrote: > >>groups in sip.conf. I've done this, however there is no definition for *8 > in > >>extensions.conf. > > Its not in extensions.conf, its in features.conf -- in extensions.conf > you have to configure callgroups for each of your extensions, so that > you can pick them up with *8. > > -- > National Manager - Special Projects > > < Sydney / Melbourne / Canberra / Hobart / London /> >2/340 Gore Street T: +61 (0) 3 9486 0411 >Fitzroy, VIC F: +61 (0) 3 9486 0611 >3065 W: http://www.squiz.net/ > > .>> Open Source - Own it - Squiz.net ./> > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup Woes
Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale <[EMAIL PROTECTED]> wrote: > I am using Cisco 7940/60/70's > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of C F > Sent: Monday, 20 March 2006 10:39 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Call Pickup Woes > > You have to configre the Dialplan in your sip phone to accept *8 > What phone are you using? > > On 3/19/06, Adam Dale <[EMAIL PROTECTED]> wrote: > > I've configured the following in features.conf > > > > pickupexten = *8 ; Configure the pickup extension. Default is *8 > > > > and all SIP extensions are configured as pickupgroup=1. > > > > These phones can make and receive calls, and also use features such as > *69, > > *70 and *98. > > > > When I dial *8 I get a beeping as if there is no valid extension and no > > debugging information when I open the console with asterisk -vvvr > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller > > Sent: Monday, 20 March 2006 9:51 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Call Pickup Woes > > > > C F wrote: > > >>groups in sip.conf. I've done this, however there is no definition for > *8 > > in > > >>extensions.conf. > > > > Its not in extensions.conf, its in features.conf -- in extensions.conf > > you have to configure callgroups for each of your extensions, so that > > you can pick them up with *8. > > > > -- > > National Manager - Special Projects > > > > < Sydney / Melbourne / Canberra / Hobart / London /> > >2/340 Gore Street T: +61 (0) 3 9486 0411 > >Fitzroy, VIC F: +61 (0) 3 9486 0611 > >3065 W: http://www.squiz.net/ > > > > .>> Open Source - Own it - Squiz.net ./> > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup Woes
C F wrote: Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale <[EMAIL PROTECTED]> wrote: I am using Cisco 7940/60/70's Don't you mean the dialplan.xml. This is what I have: -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Pickup Woes
Thank you very much. I'll now investigate how to set up dialplan.xml. I've never had to set it up before. Cheers, Much appreciated. :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 20 March 2006 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale <[EMAIL PROTECTED]> wrote: > I am using Cisco 7940/60/70's > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of C F > Sent: Monday, 20 March 2006 10:39 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Call Pickup Woes > > You have to configre the Dialplan in your sip phone to accept *8 > What phone are you using? > > On 3/19/06, Adam Dale <[EMAIL PROTECTED]> wrote: > > I've configured the following in features.conf > > > > pickupexten = *8 ; Configure the pickup extension. Default is *8 > > > > and all SIP extensions are configured as pickupgroup=1. > > > > These phones can make and receive calls, and also use features such as > *69, > > *70 and *98. > > > > When I dial *8 I get a beeping as if there is no valid extension and no > > debugging information when I open the console with asterisk -vvvr > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller > > Sent: Monday, 20 March 2006 9:51 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Call Pickup Woes > > > > C F wrote: > > >>groups in sip.conf. I've done this, however there is no definition for > *8 > > in > > >>extensions.conf. > > > > Its not in extensions.conf, its in features.conf -- in extensions.conf > > you have to configure callgroups for each of your extensions, so that > > you can pick them up with *8. > > > > -- > > National Manager - Special Projects > > > > < Sydney / Melbourne / Canberra / Hobart / London /> > >2/340 Gore Street T: +61 (0) 3 9486 0411 > >Fitzroy, VIC F: +61 (0) 3 9486 0611 > >3065 W: http://www.squiz.net/ > > > > .>> Open Source - Own it - Squiz.net ./> > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup Woes
in AAH you can set the callgroup and pickup group within each extensions setup. On 3/19/06, Adam Dale <[EMAIL PROTECTED]> wrote: > Thank you very much. I'll now investigate how to set up dialplan.xml. I've > never had to set it up before. > > Cheers, > > Much appreciated. :) > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of C F > Sent: Monday, 20 March 2006 11:54 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Call Pickup Woes > > Now I'm sure it's a dialplan problem, configure your dialplan to allow > *8. You can do that in the SIPDefault.cnf file > > On 3/19/06, Adam Dale <[EMAIL PROTECTED]> wrote: > > I am using Cisco 7940/60/70's > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of C F > > Sent: Monday, 20 March 2006 10:39 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Call Pickup Woes > > > > You have to configre the Dialplan in your sip phone to accept *8 > > What phone are you using? > > > > On 3/19/06, Adam Dale <[EMAIL PROTECTED]> wrote: > > > I've configured the following in features.conf > > > > > > pickupexten = *8 ; Configure the pickup extension. Default is *8 > > > > > > and all SIP extensions are configured as pickupgroup=1. > > > > > > These phones can make and receive calls, and also use features such as > > *69, > > > *70 and *98. > > > > > > When I dial *8 I get a beeping as if there is no valid extension and no > > > debugging information when I open the console with asterisk -vvvr > > > > > > > > > -Original Message- > > > From: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller > > > Sent: Monday, 20 March 2006 9:51 AM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: Re: [Asterisk-Users] Call Pickup Woes > > > > > > C F wrote: > > > >>groups in sip.conf. I've done this, however there is no definition for > > *8 > > > in > > > >>extensions.conf. > > > > > > Its not in extensions.conf, its in features.conf -- in extensions.conf > > > you have to configure callgroups for each of your extensions, so that > > > you can pick them up with *8. > > > > > > -- > > > National Manager - Special Projects > > > > > > < Sydney / Melbourne / Canberra / Hobart / London /> > > >2/340 Gore Street T: +61 (0) 3 9486 0411 > > >Fitzroy, VIC F: +61 (0) 3 9486 0611 > > >3065 W: http://www.squiz.net/ > > > > > > .>> Open Source - Own it - Squiz.net ./> > > > ___ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Pickup Woes
H, I'm still a little stumped. I edited SIPDefault to and created a dialplan.xml file which is being uploaded to the phone. Still no output on the asterisk console wheh I dial *8. :( dialplan.xml SIPDefault.cnf extract: # XML file that specifies the dialplan desired dial_template: "dialplan" :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, 20 March 2006 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: > Now I'm sure it's a dialplan problem, configure your dialplan to allow > *8. You can do that in the SIPDefault.cnf file > > On 3/19/06, Adam Dale <[EMAIL PROTECTED]> wrote: > >> I am using Cisco 7940/60/70's >> Don't you mean the dialplan.xml. This is what I have: -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup Woes
You don't need to mess with the dialplan.xml on a cisco phone. Try dialing *8# to pick up a ringing phone. It works just fine here with nothing special in features.conf or extensions.conf. Adam Dale wrote: H, I'm still a little stumped. I edited SIPDefault to and created a dialplan.xml file which is being uploaded to the phone. Still no output on the asterisk console wheh I dial *8. :( dialplan.xml SIPDefault.cnf extract: # XML file that specifies the dialplan desired dial_template: "dialplan" :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, 20 March 2006 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes C F wrote: Now I'm sure it's a dialplan problem, configure your dialplan to allow *8. You can do that in the SIPDefault.cnf file On 3/19/06, Adam Dale <[EMAIL PROTECTED]> wrote: I am using Cisco 7940/60/70's Don't you mean the dialplan.xml. This is what I have: ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Pickup Woes
Unfortunatly I get a beeping sound and that's it. Just like when I dial something that does not have a match in extensions.conf :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, 20 March 2006 1:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes You don't need to mess with the dialplan.xml on a cisco phone. Try dialing *8# to pick up a ringing phone. It works just fine here with nothing special in features.conf or extensions.conf. Adam Dale wrote: > H, I'm still a little stumped. I edited SIPDefault to and created a > dialplan.xml file which is being uploaded to the phone. Still no output > on the asterisk console wheh I dial *8. :( > > dialplan.xml > > > > > > SIPDefault.cnf extract: > > # XML file that specifies the dialplan desired > dial_template: "dialplan" > > :( > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle > Sent: Monday, 20 March 2006 12:10 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Call Pickup Woes > > C F wrote: >> Now I'm sure it's a dialplan problem, configure your dialplan to allow >> *8. You can do that in the SIPDefault.cnf file >> >> On 3/19/06, Adam Dale <[EMAIL PROTECTED]> wrote: >> >>> I am using Cisco 7940/60/70's >>> > > Don't you mean the dialplan.xml. > > This is what I have: > > > > > > > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup Woes
What about setting up DYNAMIC_FEATURES=>pickupexten inside your [globals] ? This is needed for, as the variable name says, dynamic features. And don't forget to set callgroup/pickupgroup to each one in your sip.conf Does anyone tested the new application Pickup()? []'s MM On Mon, 2006-03-20 at 09:24 +1100, Adam Dale wrote: > Hello all, > > > > I have an asterisk @ home system running 1.2.4. Call pickup seems to > be a bit of a problem. I’ve looked at a lot of posts and the wiki, > which states that you need to define the pickup extension in > features.conf and the pickup groups in sip.conf. I’ve done this, > however there is no definition for *8 in extensions.conf. > > > > Is there supposed to be and it has been removed? > > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup Woes
Melcon Moraes wrote: On Mon, 2006-03-20 at 09:24 +1100, Adam Dale wrote: Hello all, I have an asterisk @ home system running 1.2.4. Call pickup seems to be a bit of a problem. I’ve looked at a lot of posts and the wiki, which states that you need to define the pickup extension in features.conf and the pickup groups in sip.conf. I’ve done this, however there is no definition for *8 in extensions.conf. I've confirmed this morning. Call pickup is broken in 1.24. I've upgraded our system to 1.25 over the weekend and tested out call pickup this morning. It now works. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Pickup Woes
> And don't forget to set callgroup/pickupgroup to > each one in your sip.conf Call pickup works among IAX phones? Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Pickup Woes
Cheers Doug, Thank you all for the help. I'll upgrade to 1.2.5 soon. Much appreciated! Thanks to all who contributed! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, 20 March 2006 11:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Pickup Woes Melcon Moraes wrote: > On Mon, 2006-03-20 at 09:24 +1100, Adam Dale wrote: > >> Hello all, >> >> >> >> I have an asterisk @ home system running 1.2.4. Call pickup seems to >> be a bit of a problem. I?ve looked at a lot of posts and the wiki, >> which states that you need to define the pickup extension in >> features.conf and the pickup groups in sip.conf. I?ve done this, >> however there is no definition for *8 in extensions.conf. >> >> I've confirmed this morning. Call pickup is broken in 1.24. I've upgraded our system to 1.25 over the weekend and tested out call pickup this morning. It now works. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call pickup and SIP phones
Hi List, I have two Cisco ATA, one of them with two phones attached, and the other with just one phone. The ATA with two phones is behind a NAT, and Asterisk and the other ATA have public IP addresses. I can place and receive and blind transfer calls between them all. (Sometimes I loose registration from the ATA behind the NAT, but I think I have to upgrade to the latest firmware in the ATA) Now I'm trying to setup call pickup. I added the lines: pickupgroup=1 callgroup=1 to every entry in sip.conf, but when I try to pickup a call dialing *8 or *8# from the idle phone , nothing happens. I'm using CVS version CVS-10/10/03-19:24:38. In the console nothing shows up either. Do I have to upgrade to a more recent CVS version? Do I need to enter more parameters or configurations in other places? Does anyone have call pickup between sip phones working? If so, which version are you using? Thanks!! -- Nicolas Gudino <[EMAIL PROTECTED]> House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call pickup - still keeps ringing?
Title: Call pickup - still keeps ringing? On my system with 0.7.1 call pickup from SIP to SIP still leaves the originally dialed phone ringing for 10's of seconds after the call has been picked up on another line. There was a post a back in the fall that said this had been broken in a code update. Does 0.7.1 still have this problem or is it my user error? To be clear, I can do the call pickup with *8 just fine. However, the phone that was called keeps ringing for about 30 seconds ~after~ I've done the pickup on another phone. I can shorten this by shortening the response timeout but that doesn't work for the customer. Thanks.
Re: [asterisk-users] call pickup - Asterisk 1.4.19.1 -
works very well , features.conf 2008/5/1 Jose P. Espinal <[EMAIL PROTECTED]>: > Hello List, > > Does anyone here have call pickup (with *8 ) working ok on Asterisk > version 1.4.19.1 ? > > Thanks in advice, > > -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call pickup - Asterisk 1.4.19.1 -
Call pickup (defaults to *8) does not work for IAX2 channels. troxlinux wrote: > works very well , features.conf > > > > 2008/5/1 Jose P. Espinal <[EMAIL PROTECTED]>: >> Hello List, >> >> Does anyone here have call pickup (with *8 ) working ok on Asterisk >> version 1.4.19.1 ? -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call pickup and ring groups
I'm having trouble with call pickups.
Suppose ring group is 100 and has extensions 101 and 102.
Someone calls 100, 101 rings and 102 wants to pick the call up. If 102 dials
**100, call pickup works. If 102 dials **101, call pickup fails.
In my dialplan I have:
exten => **101,1,NoOp(pickup extension)
exten => **101,n,Pickup(101)
exten => **101,n,NoOp(pickup group)
exten => **101,n,Pickup(100)
exten => **101,n,Hangup
When 102 dials **101 I see this on the CLI:
-- SIP/4060-08868de8 is ringing
Extension Changed 4060 new state Ringing for Notify User 4061
-- Executing NoOp("SIP/4053-0886ba08", "pickup extension") in new stack
-- Executing Pickup("SIP/4053-0886ba08", "4060") in new stack
== Spawn extension (from-internal, **4060, 2) exited non-zero on
'SIP/4053-0886ba08'
It does NOT continue and display "pickup group" so it just hangs up the call.
It *should* go on and reach the "Pickup(100)" instruction...
Why is it failing?
I've noticed this only after I recently upgraded from Asterisk 1.2.30 to
1.2.31.1.
Asterisk 1.4.21.2 does not have this "bug".
Can someone please let me know if the 1.2 branch can be fixed (should I file a
bug report or will it be ignored since 1.2 only has security fixes)?
Thanks
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[Fwd: Re: [Asterisk-Users] call pickup fails.]
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 forever) al least for
6 months.
This made us to lose at least a couple of clients ("a IP PBX where you
are not able to pickup correctly other SIP extensions, are you fooling,
come back next year" ) an we keep reading again and again people saying
it is not working, and a couple of enlighted people saying their have
the luck to have it working!!
Please this is not serious!
This should be fixed for every-one-of-us (if you are one of the lucky
boys send a sip.conf to THIS LIST or post it in wiki-asterisk with a
couple of client definitions where people from the earth will be able to
pick up it) or be recogniced as not working (most of the time if you
prefer) and ask for someone to solve it (as an open bug report for example).
Is not so complicated stuff to put a callgroup=1 an a pickupgroup=1 in a
file to suspect we are all fools not getting it to work because of some
sort of mental illness, or I'm wrong. If someone feels himself
intelligent by this, he have a problem!!
The money we have invested in Digium and Asterisk stuff in the last six
months is the same money half of the people in my country
has to live eighteen years!! More or less 450 times our basic salary
here, so:
Please, there is people betting on open source software and loosing
money out there because of these "funny details", and that's the same
people making Digium earn their bucks.
Sorry for my "bad" (o I should say mad?) english :(
Thanks for your attention guys
Luis
Pd: despite *8 pickup, asterisk is great (most of the time) :)
Original Message ----
Subject:Re: [Asterisk-Users] call pickup fails.
Date: Thu, 27 May 2004 07:38:44 -0600
From: Rich Adamson <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
References: <[EMAIL PROTECTED]>
I saw a few weeks ago a discussion about cal pickup, *8, not working
but did not find a message about it being resolved, I look for a bug on
the bug list but did not find anything about it not working, nor a bug open.
I installed asterisk 0.9.0, have one sip fxo gateway and only sip
phones, all of them have callgroup=1 and pickupgroup=1 but I can not get
a call that is ringing in another phone, there is a message on the *
console that says something like "Nothing to pickup" every time I try it.
Any hints ?
It's been working fine for me on cvs Head for months. We have to use
*8# from a sip phone however.
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[Asterisk-Users] call pickup with Sipura-3000
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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call pickup and snom phones
First of all, excuse me if this is considered as OT. I'm trying to use the asterisk call pickup function on the 220 Snom phones, in other phones works well. But if I dial *8# in the snom phones, the call is no picked up. In others phones this combo of keys works perfectly. Someone could give me a clue? Any info will be appreciated. Ismael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call pickup with snom phones
hi, is there anybody who was able to setup call pickup with a snom phone? searching through the web brought up this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+snom section "call pickup" but this doesen't seem to work with current releases of the snom firmware (and looking through the patch of easywe it never worked very good at all) current snom firmware doesn't seem to send the required INVITE/REPLACE messages. any help is appreciated. regards frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup in [EMAIL PROTECTED]
Anyone out there knows how the call-pickup works on [EMAIL PROTECTED] I tried *8 and it did not work. Can a IAXs client also me assigned into a call-pickup group? Thanks in advance, Stephen ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call pickup between different protocols
Hi, I'm unable to pickup a call (*8) directed to a SIP phone from a IAX2 phone. Is it normal? I don't see ant pickupgroup/callgroup setting in iax.conf... -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup with CID info
I was wondering if someone could help me with this, I’ve searched high and low to find more info but with no success. When I want to transfer a call from another ringing sip phone to my sip handset I dial *8. This works but the caller ID shows up as *8 on my handset. What I want to do is be able to have the original caller id come up on my SIP phone rather than *8. There was a page on VOIP info (http://www.voip-info.org/wiki/view/Asterisk+bounty+Call+Pickup+with+CID+info.+-+SIP ) that talked about doing this but it doesn’t seem to be available anymore. Any suggestions or help would be greatly appreciated. Regards, Bevan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call pickup (*8) on SIP devices.
Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call pickup and ring groups
--- On Fri, 3/6/09, Vieri wrote:
> I'm having trouble with call pickups.
>
> Suppose ring group is 100 and has extensions 101 and 102.
>
> Someone calls 100, 101 rings and 102 wants to pick the call
> up. If 102 dials **100, call pickup works. If 102 dials
> **101, call pickup fails.
>
> In my dialplan I have:
>
> exten => **101,1,NoOp(pickup extension)
> exten => **101,n,Pickup(101)
> exten => **101,n,NoOp(pickup group)
> exten => **101,n,Pickup(100)
> exten => **101,n,Hangup
>
> When 102 dials **101 I see this on the CLI:
>
> -- SIP/4060-08868de8 is ringing
> Extension Changed 4060 new state Ringing for Notify User
> 4061
> -- Executing NoOp("SIP/4053-0886ba08",
> "pickup extension") in new stack
> -- Executing Pickup("SIP/4053-0886ba08",
> "4060") in new stack
> == Spawn extension (from-internal, **4060, 2) exited
> non-zero on 'SIP/4053-0886ba08'
>
> It does NOT continue and display "pickup group"
> so it just hangs up the call.
> It *should* go on and reach the "Pickup(100)"
> instruction...
>
> Why is it failing?
>
> I've noticed this only after I recently upgraded from
> Asterisk 1.2.30 to 1.2.31.1.
>
> Asterisk 1.4.21.2 does not have this "bug".
>
> Can someone please let me know if the 1.2 branch can be
> fixed (should I file a bug report or will it be ignored
> since 1.2 only has security fixes)?
>
> Thanks
Sorry for the CLI mix-up:
in my original example, 4053 is extension 102 and 4060 is 101.
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[asterisk-users] Call Pickup with 1.6.2.1 and Snom
Hi, I've used various patches with asterisk 1.4 to have support for call pickup and notification with good results. Now I'm trying vanilla 1.6.2 with its official support for "dialog-info +xml" notifications with no success. This is what i'm doing: - Phone A has a key configured as type "extension" pointing to Phone B. - In sip.conf I added notifycid=ignore-context - Phone A and B and C are in the same callgroup and pickupgroup - Phone A and B and C are in the same context Phone C calls Phone B and asterisk generates a notification for phone A: sip:35...@10.40.23.179 sip:35...@10.40.23.179 early With this notification, Phone A shows on the screen that Phone C is calling Phone B, and the function key blinks. If one presses the blinking function key, the phone generates an Invite with replaces, to try to pickup the call: INVITE sip:35...@10.40.23.179 SIP/2.0 Via: SIP/2.0/UDP 10.40.24.175:5060;branch=z9hG4bK-qoz3zjhmyfcw;rport From: "Lab 4" ;tag=o28fq65rfu To: "Lab 1" Call-ID: 3c2672b3f35a-dpd0zv11yegl CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 Replaces: pickup-3c26701519b8-5xxapzoav2u4 P-Key-Flags: keys="3" User-Agent: snom320/7.1.39 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 368 Then asterisk receives the pickup request: [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag: Totag: [Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35...@rededelca [Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no NAT) [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method INVITE - callid 3c2672b3f35a-dpd0zv11yegl [Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Using INVITE request as basis request - 3c2672b3f35a-dpd0zv11yegl [...] [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: INVITE part of call transfer. Replaces [pickup-3c26701519b8-5xxapzoav2u4] [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag: Totag: [Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35...@rededelca [Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no NAT) [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method INVITE - callid 3c2672b3f35a-dpd0zv11yegl [Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Using INVITE request as basis request - 3c2672b3f35a-dpd0zv11yegl [...] [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: About to call Pickup(35...@pickupmark) [Feb 11 10:44:13] DEBUG[4649] devicestate.c: Changing state for SIP/35504 - state 2 (In use) [Feb 11 10:44:13] DEBUG[4649] devicestate.c: device 'SIP/35504' state '2' [Feb 11 10:44:13] NOTICE[4659] app_directed_pickup.c: No target channel found for 35505. [Feb 11 10:44:13] DEBUG[4659] channel.c: Hanging up channel 'SIP/35504-000f' [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Hangup call SIP/35504-000f, SIP callid 3c2672b3f35a-dpd0zv11yegl After this obviously phone A hasn't picked up the call, and Phone B keeps ringing. Did I miss something in the dialplan or is it a bug? -- Loris Santamaria linux user #70506 xmpp:lo...@lgs.com.ve Links Global Services, C.A.http://www.lgs.com.ve Tel: 0286 952.06.87 Cel: 0414 095.00.10 sip:1...@lgs.com.ve -O9 -omg-optimize -fomit-instructions -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Fwd: Re: [Asterisk-Users] call pickup fails.]
> 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. I'm not sure what you're asking for, but *8# has been working just fine here since about October last year and still working fine on current Head cvs. If you're asking for something else, then how about rewording it. If you really are talking about plain old call pickup, our cisco 7960's work just fine with a sip.conf entry like: [3001] type=friend username=3001 secret=mysecret host=dynamic context=sip-in callgroup=2 pickupgroup=2 mailbox=3001 with extensions.conf entries like: exten => 3002,1,Dial(SIP/3002,15) exten => 3002,2,Voicemail2(u3002) exten => 3002,102,Voicemail2(b3002) exten => 3002,103,Hangup and incoming fxo lines in zapata.conf like: context=inbound-bus callgroup=2 channel => 4 If that's what you want and it isn't working, then I'd suggest reviewing your dialplan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call pickup with Sipura-3000
On Sat, 2005-02-26 at 18:22 -0700, Joseph wrote: > 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? I found it! It can be solved by defining: pickupexten = 33 ;any unique number or in Line 1 dia plan (xx.|*xx) ;this permits passing *8 through Line1 -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call pickup with Sipura-3000
When I pick up calls on my Sipura I just dial *8# instead of *8. The # will end the Sipura's dial plan. If you put *8 into the dialplan, that would work too. --On Saturday, February 26, 2005 11:39 PM -0700 Joseph <[EMAIL PROTECTED]> wrote: On Sat, 2005-02-26 at 18:22 -0700, Joseph wrote: 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? I found it! It can be solved by defining: pickupexten = 33 ;any unique number or in Line 1 dia plan (xx.|*xx) ;this permits passing *8 through Line1 -- # Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call pickup with Sipura-3000
> On Sat, 2005-02-26 at 18:22 -0700, Joseph wrote: > > 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? > > I found it! > It can be solved by defining: > pickupexten = 33 ;any unique number > > or in Line 1 dia plan > (xx.|*xx) ;this permits passing *8 through Line1 Or, without the dial plan change, just dial *8# like the wiki suggests. The "#" in this case says I'm done dialing, now send the digits to asterisk. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup and snom phones
You need to have the pickupgroups added in sip.conf Then - in order to pick up, use *8 ( and not *8# ). Under each extension ( here in group 1 ) add the following lines to sip.conf : callgroup=1 pickupgroup=1 -- Pertti [EMAIL PROTECTED] wrote: First of all, excuse me if this is considered as OT. I'm trying to use the asterisk call pickup function on the 220 Snom phones, in other phones works well. But if I dial *8# in the snom phones, the call is no picked up. In others phones this combo of keys works perfectly. Someone could give me a clue? Any info will be appreciated. Ismael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call-pickup function in Queue application
Hi All, I need a function that I believe isn't available in Asterisk, but I don't know if I'm correct about this. I have a queue and I want agents that are in that queue to have the ability to answer a call in the queue with calling an extention. For example, if I'm an agent and my colleague forgot to logout I could take the call when his phone is still ringing without walking to his desk or waiting for round robin. Can anyone tell me if this already is avalible ? Regards, Attilla ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup problem with cisco 7960 (SIP)
Please help! When I try to place a call pickup from a cisco phone 7960 using *8 the call is picked up but the other phone continues ringing. Is there any problem with call pickup in SIP. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. Everyone.. :) Its a known issue.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
WipeOut wrote: Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. Everyone.. :) Its a known issue.. Later.. OOhh :( Any known workaround ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Ing. Angel Gomez Garcia wrote: WipeOut wrote: Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. Everyone.. :) Its a known issue.. Later.. OOhh :( Any known workaround ? Not that I know of.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
WipeOut wrote: Ing. Angel Gomez Garcia wrote: WipeOut wrote: Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. Everyone.. :) Its a known issue.. Later.. OOhh :( Any known workaround ? Not that I know of.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Here with a snom200/SIP and ATA-186/MGCP everything works fine (i dial "*8" to pick up a call). -Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Thomas Dingermann wrote: WipeOut wrote: Ing. Angel Gomez Garcia wrote: WipeOut wrote: Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. Everyone.. :) Its a known issue.. Later.. OOhh :( Any known workaround ? Not that I know of.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Here with a snom200/SIP and ATA-186/MGCP everything works fine (i dial "*8" to pick up a call). -Thomas May be becasue you are using SIP and MGCP.. But when using 2 SIP UA's the phone definately keeps on ringing.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Here are some ideas for anyone with some extra time on there hands. SIP phones on call pickup either use a special REGISTER or you can place a call with the magic extension and have the switch hang up on you and immediately call you back. With the second option, you could dial "*8", Asterisk could check to see if this was a SIP channel and if so, hangup on the call and somehow add that channel to the current list of ringing phones. From the knowledge I have of Asterisk which basically is configuration info only I'm not sure if this is a clean approach with the current architecture. In the commercial SIP world, a forking proxy would be used and the pickup phone's contact would be added to the ringing phones contact list for that call only. The proxy would fork an additional INVITE to the pickup phone and both phones would ring. That way the pickup phone or the original destination phone could answer the call. Maybe these ideas will spark some coding. :) WipeOut wrote: Ing. Angel Gomez Garcia wrote: WipeOut wrote: Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. Everyone.. Its a known issue.. Later.. OOhh Any known workaround ? Not that I know of.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
okay someone find me on IRC where I can ssh in and i'll really try to fix this. Mark On Thu, 23 Oct 2003, WipeOut wrote: > Ing. Angel Gomez Garcia wrote: > > > > >Hello. > > > >I have this issue, when I pickup a call that is ringing in a SIP > > Phone, it keeps ringing. > >There is bug #116 that mention something about these, but it does > > not seem to be resolved , at least, not yet. > >Anybody else has seen it behavior ? > > > >Thank's. > > > Everyone.. :) > > Its a known issue.. > > Later.. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Yes Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call pickup (*8) on SIP devices.
Just submitted a patch for this on asterisk-dev.
Quick fix add the following line above line 5022 in chan_sip.c
ast_setstate(c,AST_STATE_DOWN);
Should look like this when you are done.
} else {
5021ast_mutex_unlock(&p->lock);
5022ast_setstate(c,
AST_STATE_DOWN);
5023ast_hangup(c);
5024ast_mutex_lock(&p->lock);
c = NULL;
-Scott
-Original Message-
From: James Sizemore [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 23, 2003 2:04 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Yes
Ing. Angel Gomez Garcia wrote:
>
>Hello.
>
>I have this issue, when I pickup a call that is ringing in a SIP
> Phone, it keeps ringing.
>There is bug #116 that mention something about these, but it does
> not seem to be resolved , at least, not yet.
>Anybody else has seen it behavior ?
>
>Thank's.
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Bisker, Scott (7805) wrote: Just submitted a patch for this on asterisk-dev GGrreeaatt!! Will test ASAP. Thank's. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call pickup (*8) on SIP devices.
> Just submitted a patch for this on asterisk-dev. > > Quick fix add the following line above line 5022 in chan_sip.c > > ast_setstate(c,AST_STATE_DOWN); Just updated to current cvs a few minutes ago primarily to get the call pickup to function properly. Using C7960's and Snom 200 on RH9. All compiled and installed cleanly. Maybe I'm misunderstanding the call pickup functions; here's a couple of samples from my sip.conf: [3000] type=friend username=3000 secret=mypassword host=dynamic context=from-sip callgroup=2 pickupgroup=2 mailbox=3000 [3001] type=friend username=3001 secret=mypassword2 host=dynamic context=from-sip callgroup=2 pickupgroup=2 callgroup=2 mailbox=3001 [3002] type=friend username=3002 secret=mypassword3 host=dynamic context=from-sip callgroup=2 pickupgroup=2 mailbox=3002 If station 3002 calls 3001, I'm expecting the user at 3000 to hear the rining at 3001, and dial *8# to pick it up. When I try that, *8# does not pick up the call and only receives a busy. Are my expectations incorrect, my definitions, or what? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Did you try to use *8 only instead of *8# ? Last time when I tried *8 picked the call with known results but I haven't tested any patches yet. I really hope call pickup now works. -- Pertti Rich Adamson wrote: Just submitted a patch for this on asterisk-dev. Quick fix add the following line above line 5022 in chan_sip.c ast_setstate(c,AST_STATE_DOWN); Just updated to current cvs a few minutes ago primarily to get the call pickup to function properly. Using C7960's and Snom 200 on RH9. All compiled and installed cleanly. Maybe I'm misunderstanding the call pickup functions; here's a couple of samples from my sip.conf: [3000] type=friend username=3000 secret=mypassword host=dynamic context=from-sip callgroup=2 pickupgroup=2 mailbox=3000 [3001] type=friend username=3001 secret=mypassword2 host=dynamic context=from-sip callgroup=2 pickupgroup=2 callgroup=2 mailbox=3001 [3002] type=friend username=3002 secret=mypassword3 host=dynamic context=from-sip callgroup=2 pickupgroup=2 mailbox=3002 If station 3002 calls 3001, I'm expecting the user at 3000 to hear the rining at 3001, and dial *8# to pick it up. When I try that, *8# does not pick up the call and only receives a busy. Are my expectations incorrect, my definitions, or what? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call pickup via *8 from ata186 (SIP)
Hello, Does call pickup works with subj? at the same pbx it works with MGCP but bit ata-186 with SIP it doesnt work, just nothing happens. Anyone have it working? Also it seems that when typing "reload" on the console, the asterisk doesnt reread the mgcp.conf. Thanks -- Anton Yurchenko<[EMAIL PROTECTED]> Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call pickup via *8 from ata186 (SIP)
Hello, Does call pickup works with ATA-186 SIP? at the same pbx it works with MGCP but bit ata-186 with SIP it doesnt work, just nothing happens. Anyone have it working? Also it seems that when typing "reload" on the console, the asterisk doesnt reread the mgcp.conf. please answer Thanks -- Anton Yurchenko<[EMAIL PROTECTED]> Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Pickup (*8) / Attended forward and CallerID
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 another person, the display of his phone only shows *8 and not the original CallerID. - When doing an attended transfer, the callerid of the original caller (A calls B, then B forwards to C => We want to show C the original callerid somewhere on his phone's screen). - When using the blind transfer feature, the CallerID is fine. I know this has already been discussed in 2006 (from digium's BTS), and would like to know if this situation did change, or not. Is it still considered as features ? Is it considered as bugs ? Will it be implemented in another way in some future release ? ...? Thanks Laurent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Pickup with more than one argument
Dear all, Does Pickup application accept multiple extensions pickup syntax, like the following line? Pickup(extension1&extension2&...) I've tried it in Asterisk 1.2.17 but it doesn't work. Does it work in Asterisk 1.4 already? Or is any other way in any version of Asterisk that I can use to do the same thing? Thanks, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Pickup with 1.6.2.1 and Snom
hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can
make call outside and exten 2006 to 2010 can not make call outside. heres my
dial plan.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic
[2001]
type=friend
context=outside
secret=1234
host=dynamic
[2002]
type=friend
context=outside
secret=1234
host=dynamic
[2003]
type=friend
contex=outside
secret=1234
host=dynamic
[2004]
type=friend
contex=outside
secret=1234
host=dynamic
[2005]
type=friend
contex=outside
secret=1234
host=dynamic
[2006]
type=friend
contex=internal
secret=1234
host=dynamic
[2007]
type=friend
contex=internal
secret=1234
host=dynamic
[2008]
type=friend
contex=internal
secret=1234
host=dynamic
[2009]
type=friend
contex=internal
secret=1234
host=dynamic
[2010]
type=friend
contex=internal
secret=1234
host=dynamic
vi /etc/asterisk/extensions.conf
[from-zaptel]
exten => s,1,wait(2)
exten => s,n,dial(sip/2000)
exten => s,n,dial(sip/2001)
exten => s,n,Playback(tt-weasels)
[others]
include => internal
include => outside
[inside]
exten => _20XX,1,Dial(SIP/${EXTEN})
exten => _20XX,n,VoiceMail(${ext...@others,u)
exten => _20XX,n,Hangup()
[outside]
exten => 2001,1,Dial(Zap/1-1/${EXTEN})
exten => 2001,n,Hangup
exten => 2002,1,Dial(Zap/1-1/${EXTEN})
exten => 2002,n,Hangup
exten => 2003,1,Dial(Zap/1-1/${EXTEN})
exten => 2003,n,Hangup
exten => 2004,1,Dial(Zap/1-1/${EXTEN})
exten => 2004,n,Hangup
exten => 2005,1,Dial(Zap/1-1/${EXTEN})
exten => 2005,n,Hangup
this is the log when i am calling from exten 2000 to outside
Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)
Verbosity is at least 3
[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite: Call
from '2002' to extension '919369613616' rejected because extension not found.
any help n support will be highly appreciated
--- On Sat, 13/2/10, Loris Santamaria wrote:
From: Loris Santamaria
Subject: [asterisk-users] Call Pickup with 1.6.2.1 and Snom
To: asterisk-users@lists.digium.com
Date: Saturday, 13 February, 2010, 8:39 AM
Hi,
I've used various patches with asterisk 1.4 to have support for call
pickup and notification with good results.
Now I'm trying vanilla 1.6.2 with its official support for "dialog-info
+xml" notifications with no success. This is what i'm doing:
- Phone A has a key configured as type "extension" pointing to Phone B.
- In sip.conf I added notifycid=ignore-context
- Phone A and B and C are in the same callgroup and pickupgroup
- Phone A and B and C are in the same context
Phone C calls Phone B and asterisk generates a notification for phone A:
sip:35...@10.40.23.179
sip:35...@10.40.23.179
early
With this notification, Phone A shows on the screen that Phone C is
calling Phone B, and the function key blinks. If one presses the
blinking function key, the phone generates an Invite with replaces, to
try to pickup the call:
INVITE sip:35...@10.40.23.179 SIP/2.0
Via: SIP/2.0/UDP 10.40.24.175:5060;branch=z9hG4bK-qoz3zjhmyfcw;rport
From: "Lab 4" ;tag=o28fq65rfu
To: "Lab 1"
Call-ID: 3c2672b3f35a-dpd0zv11yegl
CSeq: 1 INVITE
Max-Forwards: 70
Contact: ;flow-id=1
Replaces: pickup-3c26701519b8-5xxapzoav2u4
P-Key-Flags: keys="3"
User-Agent: snom320/7.1.39
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 368
Then asterisk receives the pickup request:
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use
Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag: Totag:
[Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35...@rededelca
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no
NAT)
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method
INVITE - callid 3c2672b3f35a-dpd0zv11yegl
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Using INVITE request as basis
request - 3c2672b3f35a-dpd0zv11yegl
[...]
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: INVITE part of call transfer.
Replaces [pickup-3c26701519b8-5xxapzoav2u4]
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use
Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag: Totag:
[Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35...@rededelca
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no
NAT)
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method
INVITE - callid 3c2672b3f35a-dpd0zv11yegl
[Feb 11 10:44:13] VERBOSE[4659] chan_sip
Re: [asterisk-users] Call Pickup with 1.6.2.1 and Snom
Hi, I've experienced the same thing in the 1.6.2 release, with the 1.6.1 all work as expected. There is nothing in the changelog ... So, I think it's a bug ? -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Loris Santamaria Envoyé : samedi 13 février 2010 04:09 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Call Pickup with 1.6.2.1 and Snom Hi, I've used various patches with asterisk 1.4 to have support for call pickup and notification with good results. Now I'm trying vanilla 1.6.2 with its official support for "dialog-info +xml" notifications with no success. This is what i'm doing: - Phone A has a key configured as type "extension" pointing to Phone B. - In sip.conf I added notifycid=ignore-context - Phone A and B and C are in the same callgroup and pickupgroup - Phone A and B and C are in the same context Phone C calls Phone B and asterisk generates a notification for phone A: sip:35...@10.40.23.179 sip:35...@10.40.23.179 early With this notification, Phone A shows on the screen that Phone C is calling Phone B, and the function key blinks. If one presses the blinking function key, the phone generates an Invite with replaces, to try to pickup the call: INVITE sip:35...@10.40.23.179 SIP/2.0 Via: SIP/2.0/UDP 10.40.24.175:5060;branch=z9hG4bK-qoz3zjhmyfcw;rport From: "Lab 4" ;tag=o28fq65rfu To: "Lab 1" Call-ID: 3c2672b3f35a-dpd0zv11yegl CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 Replaces: pickup-3c26701519b8-5xxapzoav2u4 P-Key-Flags: keys="3" User-Agent: snom320/7.1.39 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 368 Then asterisk receives the pickup request: [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag: Totag: [Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35...@rededelca [Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no NAT) [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method INVITE - callid 3c2672b3f35a-dpd0zv11yegl [Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Using INVITE request as basis request - 3c2672b3f35a-dpd0zv11yegl [...] [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: INVITE part of call transfer. Replaces [pickup-3c26701519b8-5xxapzoav2u4] [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag: Totag: [Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35...@rededelca [Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no NAT) [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method INVITE - callid 3c2672b3f35a-dpd0zv11yegl [Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Using INVITE request as basis request - 3c2672b3f35a-dpd0zv11yegl [...] [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: About to call Pickup(35...@pickupmark) [Feb 11 10:44:13] DEBUG[4649] devicestate.c: Changing state for SIP/35504 - state 2 (In use) [Feb 11 10:44:13] DEBUG[4649] devicestate.c: device 'SIP/35504' state '2' [Feb 11 10:44:13] NOTICE[4659] app_directed_pickup.c: No target channel found for 35505. [Feb 11 10:44:13] DEBUG[4659] channel.c: Hanging up channel 'SIP/35504-000f' [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Hangup call SIP/35504-000f, SIP callid 3c2672b3f35a-dpd0zv11yegl After this obviously phone A hasn't picked up the call, and Phone B keeps ringing. Did I miss something in the dialplan or is it a bug? -- Loris Santamaria linux user #70506 xmpp:lo...@lgs.com.ve Links Global Services, C.A.http://www.lgs.com.ve Tel: 0286 952.06.87 Cel: 0414 095.00.10 sip:1...@lgs.com.ve -O9 -omg-optimize -fomit-instructions -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call pickup across technologies (SIP, IAX, MGCP)?
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 ringing - IAX phone can't pick the call up: NOTICE[12300]: Rejected connect attempt from 192.168.x.y, reque st '[EMAIL PROTECTED]' does not exist The same applies to MGCP and SIP phone interaction. [features.conf] pickupexten = *8 [sip.conf and iax.conf] callgroup=1 pickupgroup=2 Cheers, Philipp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup with Dialog on snom display
Hello Everybody, I'm using the snom Phones together with Asterisk and I already able to see which Peer is used via "hint" priority. Then a LED on the snom phone is blinking. But I don't see who is calling the other phone. I know that the snom phones are already support this feature. But how I can enable this on Asterisk? Regards Bastian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup between ZAP and SIP technologies
Hi, I have this scenario. In my desk I have a phone connected to a FXS module of my * server. On another desk there is a phone but it is a SIP softphone (SJphone). I hear the SIP softphone is ringing, then I try to take that call with my Zap phone in my desk dialing *8, but I get fast busy tone. Is there I way do this to work ? I mean pickup phones that are ringing on different technologies ? Ardg. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call-pickup function in Queue application
Hi Attilla, I'm not sure if there is something like that available or not, but I know there are some alternatives. You can set the time out limit to say 15 seconds, which for me is about 3-4 rings on the phone before it goes looking for the next agent. The other option you can manually remove the interface from the queue via the CLI by the following: remove queue member from However, I'm not sure if that will have an effect on the call...hopefully it will just send the caller looking for the next number. I haven't personally tried it. I know some phones like the Polycom 601 have a buddy watch option. As far as I know, and someone can step in and correct me if I am wrong, that will just show if the person is on the phone or not. I don't think you can pick up on the line. Kevin Attilla De Groot wrote: Hi All, I need a function that I believe isn't available in Asterisk, but I don't know if I'm correct about this. I have a queue and I want agents that are in that queue to have the ability to answer a call in the queue with calling an extention. For example, if I'm an agent and my colleague forgot to logout I could take the call when his phone is still ringing without walking to his desk or waiting for round robin. Can anyone tell me if this already is avalible ? Regards, Attilla ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call-pickup function in Queue application
On Jun 4, 2006, at 10:33 PM, Kevin Smith wrote: Hi Attilla, I'm not sure if there is something like that available or not, but I know there are some alternatives. You can set the time out limit to say 15 seconds, which for me is about 3-4 rings on the phone before it goes looking for the next agent. The other option you can manually remove the interface from the queue via the CLI by the following: remove queue member from However, I'm not sure if that will have an effect on the call...hopefully it will just send the caller looking for the next number. I haven't personally tried it. I know some phones like the Polycom 601 have a buddy watch option. As far as I know, and someone can step in and correct me if I am wrong, that will just show if the person is on the phone or not. I don't think you can pick up on the line. Kevin Hi Kevin, Well I thought about those alternatives and I suggested them, but the person who wants them said that such a feature was avalible on another pbx where he used to work. And well, he would like the same thing on the Asterisk PBX. I already have the time at 15 seconds, and well removing a member from the queue might send it to the next agent. But if there are more then two agents in the queue there is not really a point. Regards, Attilla ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call-pickup function in Queue application
Attilla De Groot wrote: Hi All, I have a queue and I want agents that are in that queue to have the ability to answer a call in the queue with calling an extention. For example, if I'm an agent and my colleague forgot to logout I could take the call when his phone is still ringing without walking to his desk or waiting for round robin. http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call-pickup function in Queue application
Attilla De Groot wrote: > > On Jun 4, 2006, at 10:33 PM, Kevin Smith wrote: > >> Hi Attilla, >> >> I'm not sure if there is something like that available or not, but I >> know there are some alternatives. You can set the time out limit to >> say 15 seconds, which for me is about 3-4 rings on the phone before it >> goes looking for the next agent. The other option you can manually >> remove the interface from the queue via the CLI by the following: >> >> remove queue member from >> >> However, I'm not sure if that will have an effect on the >> call...hopefully it will just send the caller looking for the next >> number. I haven't personally tried it. >> >> I know some phones like the Polycom 601 have a buddy watch option. As >> far as I know, and someone can step in and correct me if I am wrong, >> that will just show if the person is on the phone or not. I don't >> think you can pick up on the line. >> >> Kevin > > Hi Kevin, > > > Well I thought about those alternatives and I suggested them, but the > person who wants them said that such a feature was avalible on another > pbx where he used to work. And well, he would like the same thing on the > Asterisk PBX. > > I already have the time at 15 seconds, and well removing a member from > the queue might send it to the next agent. But if there are more then > two agents in the queue there is not really a point. Depending on the device type could you not use call pickups with *8? Not sure if it works with queues, but it definitely works with normal calls. http://www.voip-info.org/wiki-PBX+Call+Pickup -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
