Displaying 20 results from an estimated 2000 matches similar to: "Zap and call pickup -- it don't work."
2004 May 28
1
Zap callgroup/pickupgroup question
I'm trying to set up asterisk so any phone connected to channel 1-16 of my
Adit600 channel bank can pick up a call coming in on channel 24. I do not
wish to ring any of the 16 channels on an incoming call -- this is strictly
so I can pick up the line if I see it ringing and wish to answer at work.
I have channel 24 in call group 3, and channels 1-16 in pickup groups 1 and 3.
However
2005 Jun 28
2
Trying to get *8 call pickup to work
I'm using the Debian Sarge package of Asterisk - 1.0.7 + bristuff. When
I call from a zap channel or a SIP phone to another SIP phone, then dial
*8 from a third SIP phone, I get 503 Service Unavailable on the
third phone and I get this at the Asterisk console:
Jun 28 09:01:24 DEBUG[16774]: res_features.c:1709 ast_pickup_call: No call pickup possible...
Jun 28 09:01:24 NOTICE[16774]:
2004 May 17
4
*8 problem still there?
I upgraded to the latest stable version of 1.0 today and am still seeing the
*8 problem where the phone that was originally dialed keeps on ringing even
after another phone picks up.
Are other people also seeing this? Has somebody figured out how to make this
go away?
Thanks!
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2005 Jun 02
0
chan_capi + mISDN + Fritz PTP
I'm now up&running with
- mISDN with avmfritz driver for Fritz PCI card
- chan_capi from debian recompiled with a patch (see below)
- EuroISDN with Point-to-Point (ptp) mode (Austria)
- With Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k from debian sarge
But am having some problems:
1) I needed to patch chan_capi.c from debian sarge (see below) to give
the new channel to asterisk in state
2006 Nov 04
2
Asterisk upgrade from 1.0.9 to 1.2.6 not working
Hi,
I am trying to upgrade my system (running 2.4 kernel) from 1.0.9 to
1.2.6, everything upgraded fine, however asterisk is not seeing any
zap/sip/iax2 channels.
I compiled in this order: libpri/zaptel/asterisk. Zaptel comes up
fine... ztcfg -vv shows all of my channels, however asterisk lacks the
'zap show' 'sip show' or 'iax2 show' commands, further, if I try to
force
2003 Dec 09
1
dialling peer problems
I'm trying to use Jeremy's suggestion of dialling using just the peer name
instead of user:pass@peer but I'm running into some really funky issues.
It does the same thing with both VoicePulse and another * server I have.
[voicepulse]
type=peer
host=gw5.voicepulse.com
trunk=yes
user=USERNAME
pass=PASSWORD
and in my dialplan:
Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r)
The log shows
2003 Jul 07
1
callgroup and pickupgroup
Hi,
I asked a time ago what were callgroup and pickup
group used for. I have done some proofs and all, and
I'm not sure if I have pick the idea up well!!
That's what I understand:
For example: group=1 callgroup =2 and pickupgroup=2
and my phone is a membership of the group 1.
that's mean that when a phone that belong to group 2
is ringing, I'll be able to answer this call dialing
2003 Sep 08
8
Callgroup, Pickupgroup and SIP
I have just started to play with callgroups and pickupgroups..
I updates my * from CVS this morning (about 15 mins ago)..
I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf..
I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB..
Have I not configured
2003 Nov 06
2
this is the code that breaks outgoing calls on grandstream
Here is the diff from chan_sip.c 15 days ago and 16 days ago. 15 days ago is the point outgoing calls made via grandstream budgetone stopped working.
Any help on why it breaks? Any possible fix?
/tmp# diff asterisk/channels/chan_sip.c asterisk.works/channels/chan_sip.c
289d288
< int capability;
3921,3922d3919
< p->capability = user->capability;
2005 Aug 15
2
asterisk + chan_mISDN = undefined symbol: ast_pickup_call
Hi all!
I'm getting an error when I try to start asterisk with chan_misdn.
I patched my kernel (2.6.12.4), and compiled the whole stuff (kernel,
mISDNuser, asterisk, chan_misdn). I got mISDN from
http://isdn.jolly.de/download/v3.0/
I'm using a CVS Snapshot of asterisk, which was checked out about 5
hours ago.
This is the error:
[chan_misdn.so]Aug 15 14:13:29 WARNING[4929]:
2003 Jul 01
3
picking up a ringing extension
Hello,
We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186
phones.
All sip entries have:
callgroup=1
pickupgroup=1
However I am unable to remotely pickup a ringing phone using *8#. I get
fast busy tone. Is there some flag to add in extensions.conf ?
Thanks in advance,
2014 Feb 12
1
how to selectively disable callerid block?
In Asterisk 1.8, I used the following line in extensions.conf to allow
me to pass "*82" in front of a dialed number, to disable the callerid
block that's normally on that POTS line:
; disable callerid block
exten => _*82.,1,Dial(${POTS}/${EXTEN})
But this seems to have stopped working when I upgraded to Asterisk
11.7. I get the following debug output, with a "no
2006 Feb 23
2
chan_capi-cm-0.6.4
Hello Armin, hello List
I'm trying to get chan_capi working with asterisk from debian stable
(asterisk 1.0.7, the debian version number is 1:1.0.7.dfsg.1-2).
I managed to get it compiled by providing my own version of
ast_copy_string.
This is an Austrian PTP line. I can do outgoing calls fine (no
comprehensive tests yet). For incoming calls, I'm getting "No answer"
on the
2004 Sep 13
1
agents and *8 pickupgroups
Hi folks,
Recently we assigned our users agent id's and switched to
having them use agentcallbacklogin instead of just ringing the phones
directly.... it's been going well for the most part.
Before we gave the users agent id's we had their sip configuration
set to incominglimit=1 for one line on their phone, to prevent
call waiting beeps.
If they wanted to pick up while already on a
2005 Mar 17
3
Phone ringing and not going to voicemail?
Hi,
I have one phone on my network that just keeps ringing (when I call
it) and does not go to voicemail.
If the person there is on the phone, and someone calls it they get the
busy message, but they never seem to get the 'unavailable' message...
instead it will just ring and ring and ring... any ideas?
They are setup with a voicemailbox, and it is set to transfer after 15
seconds of
2005 Sep 06
1
SIP Callgroups
Hi all,
at time i am trying to get a better idea of callgroups and pickupgroups
(especially within the SIP Channel)
A Pickupgroup is relative clear - everyone in the same pickupgroup may
pickup a call
And a callgroup does what ? - The same ?
I thought that a callgroup would act like the ZAP groups - so that you
then can dial SIP/g1 - and every SIP Client which is in the callgroup 1
does then
2006 Apr 04
6
Loading module chan_zap.so failed! PLZ help me!
Hi,
I' ve just connected a carte X100M to my asterisk
server running zaptel-1.2.5, libpri-1.2.2 and
asterisk-1.2.6 on SUSE 10.0.
When I make modprobe wcfxo and modprobe zaptel I
haven't any error, I have also chan_zap.so module
existing in /usr/lib/asterisk/modules.
But, when i run ztcfg, it shows me this:
Zaptel Configuration
======================
Channel map:
0 channels configured.
2003 Dec 17
5
ALL incoming Zap channel calls are getting picked up as FAX calls!
All,
I upgraded my asterisk setup from CVS on or about 12/15. Suddenly, *all*
of my incoming calls are coming up as FAXes. I had to disable my fax
extension because every call to my POTS line was getting redirected to my
FAX machine. After removing the FAX extension, if I call my POTS line from
my cell phone, I get the following:
*CLI> -- Starting simple switch on 'Zap/1-1'
2005 Oct 09
4
*8 and group pickup not working
Hello
I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom
IP300 phones.
My config files look like this:
features.conf
pickupextn = *8
zapata.conf
context=frompstnisdn
group=1
callgroup=1
pickupgroup=1
I also edited sip.conf like this:
group=1
callgroup=1
pickupgroup=1
But on internal and incoming calls if I dial *8 from any phone I cannot
pickup. Do I need to add
2007 Oct 18
4
Issues with making calls
Hi List,
I am from Peru, I have installed an asterisk server in my company with
digium card E1 TE120P, I am having issues when i make calls, here the
error from my server
[Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No
channel type registered for 'Zap'
[Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full:
Unable to create channel of type 'Zap' (cause