Displaying 20 results from an estimated 30000 matches similar to: "callprogress and queues"
2003 Sep 24
0
More on"Callprogress"
Here is some more stuff to add to the confusion about the "callprogress"
option. I currently have my * system operating with a T100P talking to
an ADTRAN TSU600 channel bank with 8 FXO ports connecting to the outside
world and Grandstream SIP phones as handset extensions. At first I
naively set "callprogress=yes" in zapata.conf. The results were typical
of what many
2005 Jul 12
0
TDM400P FXO callprogress doesn't detect remote answer
Location = US
asterisk/zaptel from CVS. Updated last week some time. Currently
rebuilding with todays checkout.
I have 2 fxo channels hooked up to outside standard Bell South phone lines.
If I configure as so
[channels]
context=pstn
group = 1
signalling = fxs_ks
callprogress = yes
channel => 4,3
Then any call routed from asterisk to the outside line will ring, and can be
picked up, but *
2004 Jul 25
1
X100P Inbound Issue
Hello,
After much searching of voip-info.org and google, I'm finally giving in and asking the list.
The setup I have is this:-
Single X100P card in a Debian system
Inbound/Outbound POTS line connects to the X100P
Sipura 2000 and Budgetone 100 on the LAN
1 Cordless and one conventional phone connected to the sipura
Account on Stanaphone.com for eitherbound SIP calls.
(I have other SIP
2003 May 20
0
busydetect=yes shows answered call on originating caller hangup
simple setup... two * boxen, X100P in each
pots(1) --> ZAP/1 --> *(1) --> IAX2 --> *(2) --> ZAP/1 --> pots(2)
with busydetect=yes and callprogress=yes in zapata.conf, calling from
pots(1) through to pots(2) we get the following
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
etc, and if nobody picks up, then it timeouts and hangups, good
however, if during the ringing,
2006 Feb 03
1
Re: delaying "answer" for a number of rings or an amount
Well in my setup I have a few IP phones connected to Asterisk as well as POTS phones on my analog line. When a call for my daughter comes in on the analog line (determined from callerID) I send it to her own voicemail after 20 seconds of ringing. It all works quite well.
Here's a step-by-step of what happens below:
1 - a call comes in and Asterisk rings SIP/Brian and SIP/joe for 30 seconds.
2006 Jun 14
2
Calls keep ringing after being picked up
Hi all, using * 1.2.9.1 and this week all of the sudden calls keep ringing
even after they've been picked up... Here's one users summary:
When I pick up the phone, I hear a dial tone and I am able to dial out.
But for some odd reason, the receiving line picks up while the outgoing line
is still ringing.
And the receiving line can hear everything while the phone is still ringing.
I tested
2005 Sep 12
1
Can't pickup inbound calls with TDM400P Fxo
Howdy,
1 x TDM400P card with 1 x fxo module.
1 x BT Pots line.
Location - UK
Calls work fine outbound but i'm unable to pickup the
inbound calls.
Asterisk debug:
Asterisk -vvvvvvvvvvcg
*CLI> -- Starting simple switch on 'Zap/1-1'
-- Executing Wait("Zap/1-1", "1") in new stack
-- Executing Answer("Zap/1-1", "") in new stack
2006 Jun 06
5
HELP!!!! Weird TDM2406E unable to bridge all outgoing calls.
Hi all,
I have TDM2406E with 24FXO ports connecting to 10 POTS line sitting in
my office. the out going calls symptom like when called party pickup the
phone but the calling party still hearing the ring tone from the IP phone.
Please light me up. it been many sleepless night by googling around
trying to get the right answers.
The digium card running on Intel 915G chipset. Below are my zaptel
2011 Jan 24
2
Outgoing FXO calls have no audio with callprogress=no
My outgoing FXO calls are answered but have no audio in either direction
if I have callprogress=no in chan_dahdi.conf. If I change to
callprogress=yes then the audio returns. My chan_dahdi.conf file is
listed below. Can anyone point-out why callprogress=no isn't working?
#cat /tmp/a
[trunkgroups]
[channels]
language=en
context=incoming
toneduration=40
;usedistinctiveringdetection=yes
2010 Apr 27
0
callprogress issue
I'm running Asterisk 1.6.2.1 with DAHDI 2.2.1 using a TDM410P. I have
callprogress=yes in chan_dahdi.conf because, from everything I've read, it
is needed when using call files over PSTN, which I DO use occasionally.
I know that callprogress=yes is "experimental" and causes some issues.
We've never experienced any problems when making local calls over PSTN with
callprogress
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
2003 Oct 22
2
X100P Manually Answer
I have an X100P used, at present, largely for outgoing calls. It shares the
single incoming POTS line with a number of analog phones. Is it possible to
talk the X100P (Zap/1) to answer a ringing call only if I ask it to? I'd
like to use only the SIP phone in my office, but let the analog phones
continue to work in the rest of the house (until I can afford FXS cards
anyway..)
I can force
2004 Sep 20
3
asterisk install in a home with regular phones and a x100p
Hello all,
I have a question. I have an asterisk box setup with an x100p card
installed. I have about 3 VoIP phones connected to it. The x100p is connected
to my pots line. I also have regular phones connected the the pots line as
well like most normal people would. When I get an incomming call all the
regular phones ring and my VoIP phones ring. This is good. My question is
this. Say
2003 Nov 15
0
Problem with call pickup -or- what stupid mistake have I made?
For some reason, I can't get call pickup to work between Sip phones or between
Sip and Zap phones. All phones are in the same call group and pickup group
(1). The source code was downloaded and built as of today 11/15/03.
Here's what's in sip.conf:
[general]
port=5060
bindaddr=192.168.17.2
tos=lowdelay
disallow=all
allow=ulaw
context=aliens
;
; SIP Entry for sipura line 1
; This
2011 May 05
1
Queues, pickup and transfers
Hi,
If my memory serves me right, up to Asterisk 1.6, Queue app internals kept
the application from working some other apps such as PickUp.
I wonder if such things are possible (and if possible, still keep useful
Queue Logs ie logs in which picked up or transfered calls are shown as
such):
1- a call enters a queue, a phone rings, and a non-Queue member dials some
digits and speaks with caller
2-
2007 Sep 19
0
Howto pickup call from queue?
Hi all,
how can I pickup a call from a queue? Which context parameter do I
have to specify? The context that calls the queues application is
ext-queues. This is what I tried so long (777 is the extension of the
queue I want to pickup from):
exten => _**ZXX,1,Noop(Attempt to Pickup ${EXTEN:2} by ${CALLERID(num)})
exten => _**ZXX,n,Pickup(${EXTEN:2}@from-did-direct)
exten =>
2004 Dec 01
0
extension and PSTN connection
I got two phones on an ATA-186 (601, 602) and two phones on the TDM22B
(603, 604). I have two lines on the TDM22B.
I cannot figure out some of the problems:
1. 601 dials via ZAP/3-1 to local phone number at PSTN:
ringing
pickup on PSTN (empty)
still ringing in the phone set 601
2. call from PSTN back:
601 picks up ... everything works !!!
No caller id shows up
3. For testing I have only one
2004 Nov 22
1
callprogress option
>From what I've been reading about the callprogress option, it seems like it
will work properly only with a T1 or PRI in the US. Is that correct or are
there still issues with call progress detection even if those qualifications
are met?
Thanks,
Shaun Tierney
2004 Jun 09
0
Call Pickup problem in Asterisk with SIP phones
I'm having a tough time getting call pickup to work on *. Here's my
configuration:
X100P with T-1, channels 1-4 voice <---> * <---CISCO 7960 with SIP 6.0 Image
A call comes in, and * picks up and presents a menu. Caller chooses
extension, (in this case ext 103, SIP/wsmith)
Wsmith is sitting in my office, hears his phone ringing, picks up my phone,
gets dial tone, and presses
2004 Jul 12
6
Asterisk crashing with no indication why.
I'm hoping someone might have seen this before because I'm just about
at a loss of what to do. I have an asterisk system setup in a call
center environment with multiple queues. After a random uptime asterisk
will suddenly come to a partial halt where I can connect to the cli but
issuing a command such as show channels gives no response, and calls
cannot be made in or out. Calls in