On Nov 20, 2007 10:16 AM, Kyriakos [EMAIL PROTECTED] wrote:
I have a question regarding ACD for queues. What happens when I have 2
or more queues with same weight and each queue has a call in? How will it
decide which call will be routed to the next available agent? Will it take
the
On 11/12/07, asterisk [EMAIL PROTECTED] wrote:
In my queue log I see that on the RINGNOANSWER Event I get different
content. Some events soe the ring timeout (15000). Other events show
0. Other yet show 1000 Doens anyone know what 0 means? Did it try to
ring the phone, but it was busy?
On 11/8/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
Is there any way to see the called number when a call gets redirected to
the 'a' extension from voicemail? Say x123 calls x456 and it rolls to
voicemail. x123 hits * and gets dumped into the 'a' extension in the
original context.
On 11/6/07, Stephen Bosch [EMAIL PROTECTED] wrote:
It survives if it goes to a Telus customer, but not if it crosses over
to Bell, Rogers, etc.
Well -- here's where you can help me, because our name info is not even
surviving on Telus' own network. I don't really care too much about Bell
On 11/6/07, Stephen Bosch [EMAIL PROTECTED] wrote:
We are trying to send caller ID NAME information over a Telus PRI in
Alberta.
The PRI tech says that he sees the NAME information, and for calls over
the same network, that NAME info should be reaching the receiving
station, but it is not.
On 11/5/07, Nick Brown [EMAIL PROTECTED] wrote:
Another quick question (Spending the day trying to get this project sorted
and tucked away) If I am dynamically adding queue members, they will not
abide to settings within agents.conf will they?
correct.
Ie. I need the equivalent of
On 11/2/07, Arpit Mehta [EMAIL PROTECTED] wrote:
Hello * users,
I know that passing variable in the AGI script is by
exten = _.,1,AGI(simple_c_prgm|123|789) ; 123, 789 are arguments being
passed and simple_c_prgm is C code
Now how will I receive these variables within C code ? Is it by
On 11/2/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Sadly those docs cover the situation of this:
exten = 666,1,Set(MY_VAR=fred)
exten = 666,n,AGI(simple_c_prgm)
Not this:
exten = 666,1,AGI(simple_c_prgm|123|789)
Looking back over it, you're right.
However, multiple
On 10/26/07, Matthew Fredrickson [EMAIL PROTECTED] wrote:
Is there some part of the debug output I need to tell the telco about?
When I was on to them earlier today, the engineer only seemed to know
how to turn bits of their network on and off, not much about settings
I need my end etc.
On 10/26/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
I know that you can set it up to where a user hits 0 from their mailbox
and goes to an operator, but can you set up other options as well?
Could I have 0 for an operator and 1 to go to another extension? I know
you can do this by
On 10/25/07, Adrian Marsh [EMAIL PROTECTED] wrote:
I'd like to know if anyone has figured out a way to be able to have
users logon/logoff manually from Cisco 79xx phones (with SIP firmware
loaded)?
Scenario is, user walks into office, sits at a random desk, and logs
onto the phone. The
On 10/24/07, Costa Dinoteli [EMAIL PROTECTED] wrote:
Most everytime Asterisk calls it thinks it is an Answering Machine and it
starts playing
the AMD message, instead of the delivering the 1st real message
Why is it thinking that it's a machine? If you're on the console at verbose
3 or
On 10/15/07, Pepo [EMAIL PROTECTED] wrote:
I am using Asterisk like voicemail of a great system with many users, How
do
I can get statistics of each box in the voicemail system? something like
space, number of messages, etc.
The only CLI commands are 'voicemail show zones' and 'voicemail
On 10/12/07, Yair Hakak [EMAIL PROTECTED] wrote:
you'll have to excuse the ignorance (i'm a software guy, not a telcom
guy..)
Is there any way to know if a channel has been answered by an automatic
system (like voicemail) rather than a human being?
Specifically, I want to use a .call to
On 10/11/07, James FitzGibbon [EMAIL PROTECTED] wrote:
What you do in between is up to you. Many people use something like
Wait(2) to give a comfort ring, since PRI-connected incoming calls can
often be set up nearly instantaneously. You'd want to limit the time
obviously, and have proper
On 10/11/07, Victor [EMAIL PROTECTED] wrote:
I need to process a number of lines of code in the dialplan before
answering a
call. Can standard ring back tones be played to the caller while this is
happening prior to answering the call. Which commands would facilitate
this?
You start
On 10/4/07, Arpit Mehta [EMAIL PROTECTED] wrote:
I was looking at a way to add the caller id to the outgoing calls (which are
made using .call files) using asterisk. Any ideas how to do this ?
Currently I get 'Unknown' number displayed on my phone when asterisk makes
an outgoing call.
Add a
On 9/25/07, Everton Goularth [EMAIL PROTECTED] wrote:
does anybody know any way that when it run reload app_queue in the
asterisk cli it don't lose the informations from show queue (queue
name) ?
A 'keepstats' option has been added to -trunk, and will show up when 1.6 is
released. Until
On 9/19/07, Alex Epshteyn [EMAIL PROTECTED] wrote:
Also, Asterisk restart results in all the watchers being lost. Is there a
way to force the phone to subscribe to notifications after restart (short
of
rebooting it) and is it phone specific?
Usually resubscribe-interval for extensions is
On 9/20/07, Marcus Franke [EMAIL PROTECTED] wrote:
Do you have any examples for these spec files?
I found a repository for installing Asterisk on Centos, but it
took a while before I discovered it. Ok, just checked the link
its for RHEL, but as Centos is just recompiled this won't matter.
On 9/19/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
here because we are actually specifying the IP Address of the Asterisk
server, but I am willing to try anything to fix this problem. The two user
pc's are setup on workgroups, so I do not believe that there is a domain
available that
On 9/17/07, Jim Canfield [EMAIL PROTECTED] wrote:
stuff useless. My real concern was the immediate '/ignore' for asking
about an issue with the *now ditro that actually had nothing to do with
the GUI itself. Truth be told, most of my time today was in the CLI
You may be taking what happened
On 9/17/07, Luís Palma [EMAIL PROTECTED] wrote:
Is there a way to enable the usage of UNIQUEID CDR field using a MySQL
database backend for storing CDRs without having to recompile
asterisk-addons as stated here
http://www.voip-info.org/wiki-Asterisk+cdr+mysql ?
After version 1.4 it is said
On 9/18/07, David Gomillion [EMAIL PROTECTED] wrote:
I've stayed out of this thread for a long time, and really didn't read the
past comments, so if I'm repeating someone, I'm sorry. I've been thinking
this for a while, and just have to say it. If you feel like you have to keep
people from
On 9/13/07, Kevin P. Fleming [EMAIL PROTECTED] wrote:
It shouldn't be that hard to translate the AEL example into traditional
dialplan language; in fact, Asterisk does that itself when you load the
AEL into memory, so if you load it yourself and then do a 'dialplan
show' you'll see the
On 9/14/07, Jeremy Wadhams [EMAIL PROTECTED] wrote:
In Asterisk 1.4, is there any way to force new users to configure their
mailbox? I'm thinking a simple IVR that holds a user's hand through
changing their PIN, recording their name, and setting up one or both
greetings, the very first time
On 9/13/07, Ken D'Ambrosio [EMAIL PROTECTED] wrote:
I got dragged away from Asterisk (somebody made me an offer I couldn't
refuse for system administration), but I'm thinking about seeing if I
can't get it deployed at my new employer. Regardless, there are two
things about older voicemail
On 9/11/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Part of a supervisor menu I'm writing requires that I allow the
supervisor to choose to ChanSpy a channel from the main menu then return
back to the menu (dialplan) to choose other options when she's done. Is
there a way to 'exit'
On 9/9/07, Barton Fisher [EMAIL PROTECTED] wrote:
Thanks, OK, a bit confused The cards are TE410P. I really don't
see how the set a codec for this, other than it might default to
something in code like ulaw. Any clue on how to verify codec in use
during a call?
If you absolutely want
On 9/7/07, Mark Michelson [EMAIL PROTECTED] wrote:
After talking in #asterisk-dev with those who wrote the joinempty
option, it appears I was mistaken about its use. joinempty=strict will
only not allow a caller to join the queue if the members are in a
permanently unavailable state. A member
On 9/7/07, Atis [EMAIL PROTECTED] wrote:
Well, for that case i have a RemoveQueueMember() after Dial, in case
of ${DIALSTATUS}!=ANSWERED. That works great, except for agent
complaints - that they are logged out from queue :D Would be a bit
better to be able to set agent's status to
On 9/7/07, Atis [EMAIL PROTECTED] wrote:
It doesn't work if you're using AddQueueMember with SIP channels,
because
the Dial() is implicit, so you have no control over what happens after
that
implicit Dial() finishes.
Nop, it works for Dial to SIP channels, if you set g option in Dial.
On 9/5/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
Or you might have two safe_asterisk processes trying to restart asterisk.
A symptom of this (when Asterisk is not actively crashing) is constant
remote UNIX connection messages on the console every few seconds (assuming
you have nothing that
On 9/5/07, Adrian Marsh [EMAIL PROTECTED] wrote:
Many thanks for that!! I didn't know that the order worked quite like
that but I see it now... Better go check the other contexts...
(the [56][0-9] worked fine).
You can also impose a finer level of control over the order extensions are
On 9/3/07, Arinze Izukanne [EMAIL PROTECTED] wrote:
Can you show me a sample fo config? The link schematic should look like
this:
E1 == TDMoE==E1.
Refer to the section Sample configs for setting up TDMoE between 2 servers
without TDM hardware, using ztdummy on this page:
On 8/29/07, BJ Weschke [EMAIL PROTECTED] wrote:
I think we will want to see what state chan_sip is sending into
app_queue for it to be called Uknown. What is the last state these
channels are in before they go to Unknown in app_queue?
Unfortunately, I don't know. This is in an active call
On 8/30/07, Adrian Marsh [EMAIL PROTECTED] wrote:
[outgoing-pstn-international]
exten = _+.,1,Set(EXTEN=00${EXTEN:+1})
exten = _+.,2,NoOp(test line: ${EXTEN})
Setting ${EXTEN} won't work, but Goto(context,00${EXTEN:1},priority) will:
[foo]
exten = 7997,1,Answer
exten =
On 8/29/07, Nitesh Divecha [EMAIL PROTECTED] wrote:
Basically, it would be a totally different system running Asterisk with
AGI scripts and monitoring other systems (Web Servers, FTP, SMTP). Not
specifically monitoring ports (80, 21, 25) but whole system. If system
timeouts then AGI scripts
On 8/29/07, Aubrey Wells [EMAIL PROTECTED] wrote:
I have a main Asterisk server, and a server at a branch location connected
via a IAX2 trunk. I want to have a queue at the main location that has
people from both locations as members. I got this working, but the trouble
comes when the
Does anyone know what can cause queue members to go into a status of
Unknown?
pbxtel-01*CLI queue show
cshas 2 calls (max unlimited) in 'rrmemory' strategy (24s holdtime),
W:0, C:447, A:20, SL:91.7% within 60s
Members:
SIP/1405 (dynamic) (Unknown) has taken no calls yet
On 8/28/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Realtime + MySQL does it. That needs some extra work but
it's possible.
Or DUNDi. JR just posted a quick tutorial on getting that up and running:
ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf
--
j.
On 8/24/07, Marlon Dutra [EMAIL PROTECTED] wrote:
Every queue has some status counters (completed, abandoned, hold
time...) that are very useful for statistics. The problem is that those
counters are reset every time Asterisk restarts.
Is there a way to keep those counters, maybe in astdb?
On 8/24/07, Joshua Colp [EMAIL PROTECTED] wrote:
I'm going to end this email with a question myself... how many people
have Asterisk on a development/staging server before deployment, test,
and isolate the issues they may have in their specific scenario?
I do, but many of the problems I have
On 8/21/07, Steve Edwards [EMAIL PROTECTED] wrote:
To control the tv in this room, press 1. To control a tv in another room,
press 2. To control the outside lights, press 3. To control the
sprinklers, press 4, ...
To control the power bar the Asterisk server is plugged into, press 5
click
On 8/19/07, Steve Turner [EMAIL PROTECTED] wrote:
I would like to send Multimedia Messaging (MMS) email (gateway) to my
cell
phone and have the from and subject be the callerid/calleridnam
information
from the voice mail message.
voicemail.conf lets you change the from and subject line,
I'm hoping people can suggest some ideas for debugging a problem that I'm
having with DTMF.
Unlike most of the DTMF problems reported here, it has nothing to do with
Asterisk interpreting DTMF. My problem is with the synthesis of DTMF tones
on outbound calls on a PRI connected to a TE412P card.
On 8/16/07, John Meksavan [EMAIL PROTECTED] wrote:
CLI. What am I doing wrong? Thanks in advance.
The channel spec you need to use is:
Dial(Zap/g0/${EXTEN:1})
not
Dial({Zap/g0/{EXTEN:1})
Though bear in mind that the :1 is removing the first char of your
extension, so if you dial '123' on
On 8/14/07, Atis [EMAIL PROTECTED] wrote:
That's possible, but i wouldn't recommend on large production system.
Using MySQL you would need to connect and disconnect all the time, and
it takes resources.. I would suggest to append that info to CDR
userfield (if you are storing your CDR in
On 8/14/07, Matt [EMAIL PROTECTED] wrote:
I have a 536i expansion module attached to a 57i-CT. The BLF lights
on the 536i will light up and work fine for a while... however after a
bit they seem to loose their ability to see if someone is on a phone.
They still work to dial, if I try to
On 8/14/07, Fabio Ardeola [EMAIL PROTECTED] wrote:
Let say that the user entry during the call is a
reference number of a house to rent. Would be possible
to check if the reference number is a valid entry on
the MySQL database and then base on its answer define
the next menu item on the IVR
On 8/10/07, Jason K. Carter [EMAIL PROTECTED] wrote:
Could everyone that has a working production Asterisk server that uses a
Digium telephony card as a BRI/PRI gateway let me know what
motherboard/processor your server uses?
Currently running a TE412P in a IBM x3650 Model 7979. I had some
On 8/10/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
That's great, now say you have 5 or 6 AA's and each one has 10 different
parts that you want to record (thank you for calling... for Steve
press 1 for dave press 2). Rather than having to record a long
message, I want to break it
agi_uniqueid: 1186667018.723
AGI Tx agi_callerid: 427
AGI Tx agi_calleridname: James FitzGibbon
AGI Tx agi_callingpres: 0
AGI Tx agi_callingani2: 0
AGI Tx agi_callington: 0
AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: 7993
AGI Tx agi_rdnis: unknown
AGI Tx agi_context: from-internal-admin
AGI Tx
On 8/9/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
After you run make menuselect, you'll have a file 'menuselect.makeopts'
in
your asterisk source dir. Copy that to /etc/asterisk.makeopts (or
~/.asterisk.makeopts) and it will be used for future builds. Once
you've
copied the file
On 8/9/07, Farooq Ahmed [EMAIL PROTECTED] wrote:
Asterisk has a lot of customizable voice prompt in /var/lib/asterisk/sound
but i want to change a very well known voice message which occurs when we
try to dail a number
against dial plan
beep beep beep The person you are calling is
On 8/8/07, arkda [EMAIL PROTECTED] wrote:
I've been digging around and I haven't found a way to do this, but I have
a feeling I'll feel like an idiot because it's something I'm over looking.
Normally if I need to specify an additional option (such as different
language sound files) or I'm
On 8/8/07, Mike [EMAIL PROTECTED] wrote:
I'd be most thankful for some link to a page that shows how to write such
a
function in Asterisk.
There is a test application in the source tree (not built by default I
believe), but it doesn't look like anyone has made an equivalent sample
function.
On 8/8/07, Mike [EMAIL PROTECTED] wrote:
So, I wrote (well, plagarized directly from the Web) a simple Perl program
that prints Hello World. I call it using this:
exten = 12345,1,AGI(agi-helloworld.agi)
Seems to work (I'm not expecting anything, really, just no Asterisk
error).
When I
On 8/5/07, James R. Stevens [EMAIL PROTECTED] wrote:
In the design of an Asterisk system using Cisco 7900 series SIP phones
we are struggling with giving the reception folks (3) hardware that can
tell them the status of everyone in the office (10 or so) (On the phone,
out of office etc)
On 8/3/07, Michael Munger [EMAIL PROTECTED] wrote:
Is there a way to setup an IAX bat phone (immediate=yes) or is this a
privilege only reserved for ZAP channels?
As I understand it, this would have to be supported by your specific
hard/soft phone.
It's the same with SIP - taking a handset
On 8/6/07, James R. Stevens [EMAIL PROTECTED] wrote:
I'm reading the PDF on the Cisco Expansion module and it says 'When used
as a DN key buttons are illuminated …'
Is that what we are doing within Asterisk or Trixbox when we configure an
extension? (A Directory Number??)
I suspect DN
On 8/3/07, bilal ghayyad [EMAIL PROTECTED] wrote:
At the extensions.conf file, at [demo] context, there
is a line:
exten = 1234,n,Macro(stdexten, 1234,
${GLOBAL(CONSOLE)})
In this line, I understand that it calls the macro
name stdexten [macro-stdexten] but about the other
variables, do
On 8/3/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
why if I call the Busy or Congestion extensions, the DIALSTATUS and
HANGUPCAUSE variables are not set ?
If I call (say) extension 1234 all things are set ok.
I think you've answered your own question there. The only asterisk
On 8/3/07, nik600 [EMAIL PROTECTED] wrote:
is it possible to spy (not record, spy) partially on a channel?
for exaple, i'd like to listen only the input or the output voice.
trunk has added an 'o' option to ChanSpy:
o - Only listen to audio coming from this channel.\n
You
On 8/1/07, Linux Lover [EMAIL PROTECTED] wrote:
This SOHO PBX box won't interop with Asterisk
because it doesn't speak any
of the protocols that Asterisk does. This box
I tend agree with your evaluation. Still, I was
thinking that since all these el-cheapo SOHO PBX boxes
support manual
On 8/2/07, Jay Moore [EMAIL PROTECTED] wrote:
With my current setup, I record all incoming calls to my queues. My
problem is that once a call is transferred out of a queue, recording
stops. How can I make it so recording continues even after a call is
transferred?
If you need me to post
On 7/31/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
span=1,1,0,ccs,hdb3,crc4
I was under the impression that ccs/hdb3 was more typical of E1 service than
T1.
I ran across this when looking up something on span syntax yesterday (from
On 8/1/07, Linux Lover [EMAIL PROTECTED] wrote:
But one thing that I forgot to mention is that my
business is only in its beginning stage and I need to
be as thrifty as possible. While $216 is a reasonable
price, I was wondering whether my (currently very
modest) goal can be achieved by
On 7/31/07, Benjamin Jacob [EMAIL PROTECTED] wrote:
Searched all over, but couldn't find anything conclusive.
Does an off-the-shelf version of Asterisk run without any issues on a
64-bit machine?
Does anyone have any 'conclusive' figures?
I've run 1.2.14 - 1.2.18 and 1.4.4 - 1.4.9 on CentOS
Another day, another apparant unexplained hardware incompatibility.
I have a TE412P and a TDM400B living quite happily in a whitebox using an
Intel motherboard:
http://www.intel.com/design/servers/boards/se7230nh1-e/index.htm
I tried to move to an IBM x3650 system. It uses a slightly newer
On 7/30/07, voiplist [EMAIL PROTECTED] wrote:
I noticed that if I have an agent logged in using AgentCallBackLogin
and that agent is unreachable for some reason (SIP phone unplugged)
calls to him/her will completely yack.
For example:
1-Agent 500 is the only one logged into queue number 1.
On 7/27/07, James FitzGibbon [EMAIL PROTECTED] wrote:
I'll go open a bug report.
http://bugs.digium.com/view.php?id=10320
For anyone who wants to track it.
--
j.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk
On 7/26/07, James FitzGibbon [EMAIL PROTECTED] wrote:
Is it possible for qe.parent-membercount to be set to zero in a queue
where all agents but one are on the phone and that one remaining agent lets
their phone ring without answering it?
I added some debug code to app_queue and ran a few
On 7/27/07, Mark Michelson [EMAIL PROTECTED] wrote:
Could you submit this as an issue on the bugtracker? The 'n' option was
mucked with just prior to the 1.4.9 release and so any problems
experienced with it should be reported there so they can be fixed as
quickly as possible.
It's been
I am experiencing a change in behaviour of my Queues in 1.4.9 vs 1.4.8.
I do not pass the 'n' option to any call to Queue() in my dialplan. Yet
since I upgraded to 1.4.9, I have occasionally seen this on my console:
-- Nobody picked up in 2 ms
-- Exiting on time-out cycle
That log
I just saw this on my console:
[Jul 26 11:36:30] WARNING[8667] file.c: File vm-duration does not exist in
any format
[Jul 26 11:36:30] WARNING[8667] file.c: Unable to open vm-duration (format
0x4 (ulaw)): No such file or directory
Thinking I might have lost a file during a fsck or something, I
On 7/24/07, Asterisk guy [EMAIL PROTECTED] wrote:
-- Attempting call on Local/[EMAIL PROTECTED] for application MusicOnHold()
(Retry 1)
Jul 24 08:23:17 NOTICE[21177]: chan_local.c:479 local_alloc: No such
extension/context [EMAIL PROTECTED] creating local channel
Jul 24 08:23:17
On 7/24/07, hugolivude [EMAIL PROTECTED] wrote:
Thanks or all your help!
I've posted the ./configure output below. I noticed that it says:
checking for mysql_init in -lmysqlclient... no
Presumably that's a problem, but I don't know how to fix it!! As I
mentioned, I have MySQL installed
On 7/23/07, Dovid B [EMAIL PROTECTED] wrote:
Can it be that asterisk does not have permission to copy the file over ?
Also check your date settings on the server.
Yes, it's interesting that the page intro includes the sentence Lots of
error checking to make sure its done correctly, but the
On 7/20/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Did you read UPGRADE.txt? Priority jumping was deprecated in 1.2. I
assume it was removed from 1.4.
According to UPGRADE.txt, the default in the absence of priorityjumping=
changed from yes in 1.2 to no in 1.4:
* In previous
On 7/16/07, The Asterisk Development Team [EMAIL PROTECTED] wrote:
fix various known issues. See the ChangeLog included in the releases
for a full list of changes. The ChangeLogs are also available
separately on the ftp site.
Is there any more information available on this change?
On 7/16/07, Adrian Marsh [EMAIL PROTECTED] wrote:
Anyone know if theres a way to share a Cisco 7940 between hot-desk
users?
My phones get their setup via SIP .cnf files, that load at boot via
tftp, so I'm assuming the configs a failry static. However if I want a
phone to be hot-desked, I could
On 7/12/07, Jared Smith [EMAIL PROTECTED] wrote:
It probably wouldn't hurt to open a bug for this... I've seen something
like this before, only it was manager events ending up inside of SIP
traffic. It definitely sounds like a pointer problem or maybe a locking
problem to me... which means
On 7/12/07, Stefan Reuter [EMAIL PROTECTED] wrote:
You might want to have a look at QueueMetrics:
http://queuemetrics.loway.it/
I am not sure if it supports all features you are looking for but it
should be a good start.
QueueMetrics is working well for me in a 75 seat call center, but it
On 7/6/07, James FitzGibbon [EMAIL PROTECTED] wrote:
Under 1.4.5 and 1.4.6, I've seen a few instances where the channel name in
queue log entries is replaced by a snippet of a manager event:
Nobody else seeing this? I'm at a loss - it's only one queue now that I go
and look at the history
On 7/9/07, Daniel Gradecak [EMAIL PROTECTED] wrote:
are you sure the monitor is started and sotoped via the dialplan ?
If you're using Monitor() or MixMonitor(), then just add a UserEvent() call
just before it in the dialplan.
If you're doing monitoring of queues, it's a bit trickier - you
Under 1.4.5 and 1.4.6, I've seen a few instances where the channel name in
queue log entries is replaced by a snippet of a manager event:
--START--
1183582823|1183582823.104763|queuename|SIP/|REMOVEMEMBER|
1183582828|1183582793.104744|queuename|
Context: macro-dialout
Extension: s
Priority:
Has anyone observed a problem where using Local channels with AddQueueMember
results in missing TRANSFER events?
Right now I'm using straight SIP channels when I call AddQueueMember(). I'm
contemplating moving to Local channels because the non-state-based
wrapuptime blows when you have a
Has anyone successfully run * 1.4 with the following configuration (or
something very similar)?
HP DL380 G5 (3Ghz Xeon)
CentOS 4.5 (kernel 2.6.9-55)
Asterisk 1.4.5 (or 1.4.4)
Zaptel 1.4.3 (or 1.4.2.1)
TE412P
TDM400B (2x FXO and 2x FXS modules)
I've had this rig running * 1.2.18 with Zaptel
On 6/26/07, Marty Mastera [EMAIL PROTECTED] wrote:
Any recommendations on an economical layer 3 switch? Preferably
something that you have hands on experience with connecting to IP phones
with attached PCs? Specifically I need the ability to set the VLAN in the
phone to tag voice packets and
On 6/17/07, Nick Seraphin [EMAIL PROTECTED] wrote:
Yes... 1.5 cents per dip... you prepay the fees... and they deduct from
the prepaid amount. You can start with $5.00 which seems like a low-risk
to check it out at least.
The CLEC I use is more expensive that that for CNAM, and they want
On 5/22/07, Axel Thimm [EMAIL PROTECTED] wrote:
Have you tried using the 1.4.x atrpms packages?
I did try the 1.4 packages from atrpms overnight yesterday, with similar
results. I was able to address the kernel panic when unloading by
commenting out ztcfg -s in the stop() function of the
I attempted an upgrade of our production system from Asterisk/Zaptel 1.2 to
1.4 this weekend. Intially everything looked like it was working properly,
but some time in the day following the upgrade, the system died to a kernel
panic. I wasn't able to catch the entire kernel dump on the console
On 5/15/07, lizhong zhu [EMAIL PROTECTED] wrote:
I compiled asterisk under arm-linux. i am using asterisk 1.4.2. i can run
./configure and menuselect with embedded modules. but running make comes out
errors:
ranlib libmxml.a
[...[
/usr/src/asterisk-1.4.2/include/asterisk/paths.h:23:
On 5/11/07, Morgan Gilroy [EMAIL PROTECTED] wrote:
At the moment to find the codecs used I have to look though the sip
trace or show channels/show channel (annoying when you have 50+
channels).
Im just trying to find an easier and quicker way to keep track of the
codecs used to help with debug
On 5/8/07, Pedro Silva [EMAIL PROTECTED] wrote:
Can i identify the sound files that are played in the asterisk
console? I defined the verbose to 100 but i can not see the sound
files that are played in some situations... :(
For example, I need to know what files are played for the message:
On 5/1/07, Savoy, Kevin - Williston, ND [EMAIL PROTECTED] wrote:
Not sure if this can be done or not, but I can't seem to find it anywhere
on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to
have the caller id of the person I am dialing displayed and not the number I
On 4/25/07, Mike Lynchfield [EMAIL PROTECTED] wrote:
may i add , eyebeams confnig file is xml and could be generated , BUT, the
password is hashed in some way.. any idea on that ? its a pretty long hash
Two options:
- type in the passwords manually, shut down eyeBeam, and then cut/paste the
On 4/20/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
I am not. The soft phone is not the only software on that computer that
needs cetral configuration.
How do you configure the networking on those computers? The mail
clients? How do you deploy updates?
The fundamental problem, as I
On 4/21/07, Senad Jordanovic [EMAIL PROTECTED] wrote:
What about creating a configuration file on server for each soft phone
extension automatically and then importing that file into the soft phone?
In another words, user receives a link to the setup program and the
configuration file in an
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