Displaying 20 results from an estimated 2000 matches similar to: "Attended Transfer question"
2011 Apr 12
0
No subject
[0004f2xxxxxx](poly650)
defaultuser=0004f2xxxxxx
callerid="Front Desk" <1600>
mailbox=1600
*setvar=callidnum=1234561600*
and from extensions.conf:
[outgoing]
; Outbound unrestricted domestic calls
exten => _1NXXXXXXXXX,1,Verbose(Outbound call from ${callidnum} to ${EXTEN}
on ${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)}.)
*exten =>
2011 Jun 25
1
Cisco IP Phones and Skinny in asterisk 1.8.4.2 "tooooooooooooooooo"
Hi All;
Again, the Cisco IP Phones 7942G and using Skinny:
I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file.
The phones are registering, but when we use them to place a call, we only hear tooooooooooo in the handset and we do not hear voice (even when we dial the digits, we only hear toooooooo .. but it dials and destination
2013 May 01
1
multiple provider for incoming
Matt,
At some point you need to consider how much is too much...
I run a call center with more then 125 commissioned phone sales reps and more than 60 customer service reps. We run dual servers, fiber from one provider and 6 bonded T1's from another provider. We purchase our so trunks from a wholesale company who is a major provider to resellers. Being so, their network is extremely
2013 Jan 16
1
Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start
I'm trying to decide if I need to open an issue for this or if it's just a
misconfiguration issue of some sort. Here's the situation - yesterday
morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS
5.8 installation and got a shell of a basic asterisk install setup (minimum
required configuration files, etc, with no dialplan or sip peers setup
yet). In the
2010 Jul 13
3
STRFTIME function declared in globals context
I'm trying to declare a few date-related global variables to ease my
dialplan. When I declare the following in the [globals] context of
extensions.conf, I get unexpected results:
YEAR = ${STRFTIME(${EPOCH},,%Y)}
MONTH = ${STRFTIME(${EPOCH},,%m)}
DAY = ${STRFTIME(${EPOCH},,%d)}
TIMESTAMP = ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}
If I evaluate these variables in the dialplan later, using
exten
2011 Oct 19
1
Asterisk call transfers not working
Hello:
We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0
running. Everything seems to be ok but call transfers. This is the issue:
*A, B, C and D are in FXS ports*.
1) A calls B. B anwers.
2) B tries to transfer the call to C dialing *2 (code for attended
transfer).
3) A hears MOH. B dials number C.
4) Asterisk says the dialed number is incorrect or non existing.
We tried
2011 Apr 12
0
No subject
be able to setup a SIP trunk. I've been able to successfully integrate a
Cisco CallManager 7.x system with Asterisk using SIP trunking, so I imagine
you should be able to do the same here.
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
--0016e651f0a6bbe47b04a303939e
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable
<div
2011 Aug 05
1
Ring delay problem
Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and
Celeron), and last days when I call from one extension to another of the
same PBX after I dial the number the rings sound after 20 seconds.
In the CLI log, when I debug the AGI, I see always goes good until
dialparties.agi, and after that there are 20 seconds without any log, and so
the ring sound.
I've read
2011 Jan 10
0
No subject
takes precedence over a queue's defined moh class.
--=20
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
--000e0ce0494051d402049b4247c1
Content-Type: text/html; charset=windows-1252
Content-Transfer-Encoding: quoted-printable
<div class=3D"gmail_quote">On Tue, Feb 1, 2011 at 10:20 AM, Danny Nicholas =
<span dir=3D"ltr"><<a
2011 Jan 10
0
No subject
Moh show files
This will show you if your class is set up correctly.
------=_NextPart_000_016C_01CBF83B.306A1A90
Content-Type: text/html;
charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable
<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
2010 May 26
1
Error compiling DAHDI...
I was at a client site tonight to install OSLEC on his machine running
asterisk 1.6.0.22 and DAHDI 2.2.1 installed via yum. I stopped asterisk and
DAHDI, downloaded the latest version of DAHDI 2.2.1
(dahdi-linux-complete-2.2.1.2+2.2.1.1) and made the necessary changes to
compile OSLEC with DAHDI, but I ran into compilation issues that I had never
seen before. So as a test I deleted my
2012 May 29
2
Fax Server for Asterisk
Hello,
For those customers with only analog lines, who ask for fax2email and
email2fax, whats the most reliable solution available and tested with
Asterisk?
Thanks
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2010 Feb 16
1
How does holdtime get calculated for queues
I had a customer ask me this question today, and I was surprised to say I
didn't have an exact answer for them. They have a relatively small support
queue for their business (three agents, and rarely more than one person in
line at any given time in the queue, if all agents are on a call), Their
support calls can range anywhere from 30 seconds to 30 minutes.
Occasionally, they'll get
2012 Feb 16
1
Park() ignores 'r' option which should disable music on hold in favour of ringing tone
When I receive a call, I want to automatically park it from the dialplan so that I can retrieve it later. However, I don't want callers to be aware that they are being parked, so I want to play a ringing tone to the caller. Park() is supposed to be able to do this:
Park([timeout][,return_context[,return_exten[,return_priority[,options[,parking_lot_name]]]]])
options
r: Send ringing
2010 Jul 26
1
VPMADT032 Failed! Unable to ping the DSP (2)!
Running Asterisk 1.6.2.9, DAHDI 2.3.0.1, CentOS 5.5 (update to date as of a
week ago), I've installed a Digium AEX800P with 2 X400M FXO Modules and 1
VPMADT032 Module, hooked up to 5 analog lines. I get the error message
referenced in the subject in my dmesg output everytime I load / reload DAHDI
using the command "system dahdi start/restart". When I make an outbound
call over
2011 Apr 05
5
IAS trunk error AES encryption disabled. Install OpenSSL.
Hey Guys!
I am getting this wired error when i am calling IAX trunk. Everything works! but i want to get rid on these RED WARNING messages.. what is wrong here ? I have func_aes.so module loaded. also i remove and test but still same error.
-Satish
== Using SIP RTP CoS mark 5
-- Executing [7623 at from-sip:1] Macro("SIP/7527-0000000d", "orasebcamdial,7623") in
2011 Apr 11
3
changing port 5060 to 5061
please send me the ways to change asterisk port from 5060 to 5061 i
need to configure it because we are already using 5060 port in router
then we cant use it again we have to configure other sip server so
please suggest me a way..........................
On 4/10/11, asterisk-users-request at lists.digium.com
<asterisk-users-request at lists.digium.com> wrote:
> Send asterisk-users mailing
2012 Jul 05
7
FreePBX: using context other than the default context and the generation for the configuration
Hi All;
If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal).
Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I
2010 Sep 24
3
should trixbox system hang when ISP drops connection?
NEWBIE alert: i'm a linux person, not an asterisk person so i'm
certainly capable of handling any linux-flavoured solution you can
suggest. here's a note i got from a local company i know (some proper
names removed):
===== start =====
Now and again our ISP goes down and when it does give us a hicup, the
Asterisk system shuts down (not very forgiving). When it shuts down
our phone
2011 Apr 08
2
MOH not working
I am using Elastix. Asterisk is used for PBX application in Elastix. I want
to make customize MOH. But not able to use MOH. I make simple extension in
asterisk conf file but no success :(
Below are the details of configuration files.
Even default MOH is also not working....
*Asterisk Version 1.6.2.17.2
*
*1) Extension.conf*
[incoming]
exten => 6000,1,Answer
exten =>