Displaying 20 results from an estimated 5000 matches similar to: "Attended transfer reconnect when goes to voicemail?"
2013 Feb 04
1
CallerID external call after Attended Transfer
Hello,
using Asterisk 1.8.12.2
case :
I call with my cellphone to our public telephone number
Our receptionist answers the incoming call and does an attended transfer
to my colleague ( A )
Colleague answers and the receptionist tells him that I am on the other
side.
Receptionist transfers the call and I am connected to my colleague ( B )
My question is about the CallerID that the
2005 Jan 18
14
Attended call transfer
Hi All,
Does any one know if attended call transfer has been added into the STABLE
release of asterisk yet? Potentially using a mix of phones would create
confusion in a user base, any ideas on attended transfer or how to achieve
this / mods to dial plan etc would be greatly appreciated.
I have been on an almost vertical learning curve with Asterisk and Linux for
6 months this is just
2006 Jan 06
3
Announcing a call transfer
With our current pbx system, a call comes in from the PSTN to the
receptionist. She then hits flash, which puts the caller on hold, calls
my extension, says "so and so is on the phone for you", I say "ok put
him through", she hangs up and I am connected to the caller.
With asterisk@home I can it # then the extension to transfer to and it
will ring there. But is there a
2006 Jan 25
1
ISDN D-channel disconnects for a minute every 5 minutes
I have a problem with Asterisk-bristuffed using a zaphfc card.
I am located in the Netherlands, so I have an ISDN line from KPN. When I
start Asterisk, and plug in the ISDN line, everything works perfectly for
about 5 minutes. And then the ISDN line is down for 1 minute, and after that
minute, the line comes back up and works for another 5 minutes. Every time
the line goes down I get the error
2009 Jun 11
3
SIP hacked connection?
Hi
Running 1.2.26 BRI stuffed. Calls made via PSTN via ISDN interface (Junghanns).
SIP ports mapped through firewall as we often connect from outside, but all SIP accounts have good passwords.
However our telecoms provider picked up a few suspicious calls to places we do not normally call at times we do not often call.
Looking at Asterisk logs it shows SIP session from the internet connected
2006 Feb 09
3
Corrupt CDR records in Asterisk 1.2.x
I have a problem with CDR recording in Asterisk 1.2.x. This is the
situation:
An Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1 with florz) machine with a single
HFC-S ISDN BRI card. I log the call records to both the Master.csv and
MySQL.
The problem is that when an incoming call from the ISDN line is logged to
the CDR, the "src" and the "clid" field show up as something like
2006 May 09
6
Bristuffed Asterisk: Hangup problems
Hello,
I have a problem with the Bristuffed version of Asterisk. I have tried
Bristuff-0.3.0-pre-1m,n,o,p (Asterisk 1.2.6 to 1.2.7.1) but they all have
the same problem it seems:
The setup:
A machine with a single hfc-s PCI BRI adapter running Gentoo 2.6.15.
Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1) installed and working perfectly.
Grandstream gxp-2000 as a SIP phone, and a normal mobile
2006 Nov 16
5
spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
Hi,
I'm using spandsp-0.0.3
[http://www.soft-switch.org/downloads/snapshots/spandsp/
spandsp-20061116.tar.gz]
on a bristuffed asterisk (1.2.13)
[http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0-
PRE-1v.tar.gz]
libtiff is at version 3.6.0
Running on: Linux router2 2.6.17-2-686 #1 SMP Wed Sep 13 16:34:10 UTC
2006 i686 GNU/Linux
Debian testing distro.
I've tried many
2006 Feb 14
3
Grandstream hold one way audio -URGENT
Hi all,
At our customer site i've installed one asterisk server with 20
Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the
customer, the receptionist picks up, and does an attended transfer (the
'grandstream way') to a collegue. Most of the times this goes ok, but
sometimes, when the receptionist puts the call on hold, and tries te
reconnect to the caller there's
2007 Sep 20
5
Horrible problem - calls losing sound
We're having a horrid problem with our asterisk setup.
Sometimes calls just go dead - we can't hear what the other end is
saying. (I think they can't hear us either). The call doesn't hang up
until one of the callers gets bored.
Internaly we use Thomson ST2030 SIP phones.
Externaly we have 3 ISDN BRI lines (6 channels total), connected to an
Eicon Diver server card (4BRI).
2006 Jan 23
5
Bug in attended transfer or as expected?
Hi all,
I have had quite a few customer complaints about attended transfer
cutting off callers.
The problem is when reception is busy she doesn't always wait for
someone to answer the call, however hanging up a ringing transfer on
attended also hangs up the caller.
I have checked the scripts I don't *think* this is a dial plan error but
if anyone has this working correctly on Asterisk
2006 Oct 26
1
chan_capi and bristuff
Hi,
Reading from www.voip-info.org, i can see that "Junghann's chan_capi is now
part of bristuff <http://www.voip-info.org/wiki/view/Bristuff> , as of
version 0.3.0-pre".
What does that really mean ?
Shall I understand I can share a Junghanns QuadBRI board between 2
CAPI-enabled software (like a 0.3.0-pre bristuffed Asterisk for instance) ?
If positive, what are the pros and
2008 Jan 28
2
IAX Calls - One Way Audio
Hello List,
I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up.
For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for
2007 May 11
2
Strange problem with asterisk
Situation such. There is an asterisk working as office pbx. 6 fxo - 18
fxs ports. All works perfectly, but some times in a week something
occurs. Could not catch what exactly yet. But symptoms such. The
asterisk infinitely writes the message of a type to broad gullies:
WARNING [20757] chan_zap.c: We're Zap/8-1, not ... <ZOMBIE>. Numbers
of channels can change. Because of that that broad
2007 Jul 02
2
Sip phones using the wrong context for an outbound call
Hi,
recently I changend a few things in the configuration of the Asterisk
1.2.17-BRIstuffed-0.3.0-PRE-1y-d of a customer. One demand was that
different groups of SIP-Phones are using different trunks to the outside
worls, so I moved some of them to a Support context.
However, dial out from this phones failes as they're still looking for
an extension in the default context, which doesn't
2006 Mar 20
2
ISDN Protocol Unknom Error with Junghanns OctoBRI
Hello,
I recently bought a Junghanns Octobri Card. I have some problems with
this card to make outbound calls but I can receive calls.
I have 3 lines to PSTN and 3 lines to my existing PBX
FRANCE TELECOM <-- OctoBRI --> Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h
<-- OctoBRI --> PABX e-Generis <----> ISDN Phones
|
2007 Dec 21
3
7970 CTLFile.tlv?
I've got a Cisco 7970 that's not completing its network registration to
Asterisk. The "Registering" message stays on the screen (with the moving
time wheel). After a few minutes, the onscreen message flashes "Updating
CTL" then "Loading...", then the status messages update with:
No valid CAPF server
File Not Found: CTLFile.tlv
No CTL installed
2004 Nov 20
6
SIP Phones-Receptionist Setup
I am looking at placing a system in an office with a central receptionist,
and phones for each individual employee thereafter. Could I use a Snom 220
with additional keypads to view if the lines are in use by the other
employees?
Fred is in sales... A call comes into the receptionist and they transfer the
call to Fred. The receptionist can tell Fred is still on the phone by
viewing the assigned
2006 Jan 14
4
Ugly echo cancel, with Bristuff/Zaphfc
I'm using bristuffed Asterisk with ISDN/ZAPHFC
I have VERY ugly (outgoing) sound through ISDN/HFC if echocancel=yes in
zapata.conf, but without echocancel I have bad (incoming) echo
Through PSTN/FXO sound is ok with or without echocancel.
I tried other echo cancellers (in zconfig.h) two times:
ECHO_CAN_KB1 (this was default)
ECHO_CAN_MARK2
ECHO_CAN_MG2
after any change I compiled (make
2007 Apr 16
1
Need some dialplan help for obscure user request
I have a customer who wants their receptionist to input the users' long
distance PINs for the because they use each others pins. I am having
trouble coming up with a way to do this because of creating a channel
between the user and receptionist, dropping the channel and its variables
and creating a new one for the actual long distance call. Any advice is
really needed.
1. User Dials Long