On 2014-08-19 23:56, Jeff LaCoursiere wrote:
Hello,
I wrote earlier today about a new PRI installation in the Caribbean, where all outbound
calls are functioning fine *except* calls to Sprint phone numbers, which get rejected
immediately as busy.
The telco has been working with their switch
On 2014-08-08 21:54, Jerry Geis wrote:
On Thu, Aug 7, 2014 at 2:53 PM, Jerry Geis ge...@pagestation.com
mailto:ge...@pagestation.com wrote:
I am using a cyberdata sip paging adapter and with the
Dial(MulticastRTP/basic/IP:port) and with
tshark I see the RTP data, my device looks like
On 2014-03-07 17:31, Paul Belanger wrote:
On Thu, Mar 6, 2014 at 3:33 PM, Markus unive...@truemetal.org wrote:
Hi Thorolf,
Am 06.03.2014 16:21, schrieb Thorolf Godawa:
Using (para-)virtualization with Xen could be an other option, on
systems with low load this works reliable, but what
On 2014-02-28 14:04, Tahir Almas wrote:
1) We do not perform any transcoding whatsoever. All recordings, and
voice mail are in G729,
and allow=g729 for all peers and in sip.conf. Is there anything else
we need to perform g729 passthrough. More importantly are we still
liable?
On 2013-09-25 09:22, Endri Stefani wrote:
Hi
Greeting to all you out there.
I am new at asterisk, I have been working with PLMN platforms telecommunication
for 5 years with NSN and Huawei.
We have recently built an asterisk PBX with Trixbox and connected it to our MSS
using Digium E1
Maybe you should open 11955 on you fw as well. This could be the rtcp port.
Regards
Hans
On 2013-09-13 11:49, Jonas Kellens wrote:
Hello,
and when I define 11500 - 11954 it should use a random port in this range.
Where is it stated that you MUST use 1-2 ???
Someone else please ?
Hello,
I 'm looking for a way to pass the '302 moved temporarily' received from the
SIP device
back to the SIP provider.
Here is the setup:
Some SIP phones are connected to an Asterisk System version 1.8.
External connection to the public network is also done via SIP to a VoIP
provider.
Phone
You did not show how the Nortel side is configured, especially LD 17 ADAN
configuration.
Regards
Hans
On 2013-05-03 11:27, Danilo Dionisi wrote:
I'm sorry, the mail is automatically send :p
However, I am for the Asterisk, there are other external consultants for Nortel
... according to you
On 2013-03-03 18:41, Olivier wrote:
hello,
In a machine I've got :
CLI pri set debug off
No such command 'pri set debug off' (type 'core show help pri set' for other
possible commands)
CLI core show help pri
pri intense debug span no description available
pri service disable
Please check out the scripts located in contrib/scripts
Regards
Hans
On 2012-05-23 11:42, Danny Dias wrote:
Hi, thanks for your answers...
Can i delete like this:
rm -rf /var/spool/asterisk/voicemail/voicemailcontextcustomer/300/INBOX/*.*
Is that ok? will this break something?
A little
Hello,
is there a way to disable a span for maintenance purpose (i.e. send yellow
alarm) ?
What would be the correct ioctl definition ? DAHDI_MAINT seems not to be the
right
candidate. Would DAHDI_SHUTDOWN send an alarm ?
Thanks
Hans
--
On 2012-03-13 18:38, Chris Bagnall wrote:
Greetings list,
I'm trying to source a very basic ISDN BRI - SIP gateway. Unfortunately,
everything I've seen seems to want to do lots of other things - registering
handsets, IVRs, voicemail, etc. I only want it to
present an ISDN BRI as a SIP account
, 2012 at 1:45 PM, Johann Steinwendtner
steinwendt...@gmx.net wrote:
On 2012-01-09 17:46, Alex Villacís Lasso wrote:
I am trying to collect information regarding a bug report for Elastix
(http://bugs.elastix.org/view.php?id=1146). In this bug, an user has
asterisk-1.8.7 and dahdi-2.4.1.2. He
On 2012-01-09 17:46, Alex Villacís Lasso wrote:
I am trying to collect information regarding a bug report for Elastix
(http://bugs.elastix.org/view.php?id=1146). In this bug, an user has
asterisk-1.8.7 and dahdi-2.4.1.2. He is trying to make an
outbound call through an ISDN trunk, by placing
of them is on
wikipedia (http://en.wikipedia.org/wiki/List_of_SMS_gateways). For the
Asterisk side, you have an extension that sends the email. I personally
use an AGI script for this part, but you could use a System() call as well.
--johann
Hello !
I 'm using a TE405P with a HW echocanceller module attached on it.
dahdi version is dahdi-linux-complete-2.2.0.2+2.2.0.
As far as I know, the fax tone detection is done on the FW board.
How can I verify that the echo canceller has been turned off ?
When I do a cat /proc/dahdi/1 for span
On 2010-06-22 12:36, Remco Bressers wrote:
Dear list,
I've got the following setup :
[FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP]
On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general].
The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the
On 2010-06-22 15:16, Remco Bressers wrote:
On 06/22/2010 02:51 PM, Johann Steinwendtner wrote:
On 2010-06-22 12:36, Remco Bressers wrote:
Dear list,
I've got the following setup :
[FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP]
On the PBX's we run Asterisk 1.4.33
this into a macro and tweak it further based
other patterns that you encounter.
--johann
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Zhang Shukun wrote:
hi, all
in my test,it shows Playback will answer the call automaticly, but i
don't want to so.
i will use answer function to answer the call. could you help me ?
core show application Playback
Regards
Hans
--
Magnus Benngård wrote:
Gentlemen,
I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk.
0851711201 and 0851711290 is on our WAN, no NAT.
0197673581 is outside our WAN and needs to be NAT'ed.
Sending a fax from 0851711201 to 0851711290, no problem, switches to T38
and fax
Kevin P. Fleming wrote:
David Gibbons wrote:
snip
This doesn't work?
Dial(SIP/*31#ww061234123412)
/snip
When I was browsing the sip debugs, it seemed that the 'w' was not being
honored for one reason or another. My thought at the time was maybe it
didn't work at all over SIP.
Does the
Olivier schrieb:
2010/1/7 David Backeberg dbackeb...@gmail.com
mailto:dbackeb...@gmail.com
On Wed, Jan 6, 2010 at 6:23 PM, Olivier oza-4...@myamail.com
mailto:oza-4...@myamail.com wrote:
The second time I'm dialing an internal extension attached to the
same
randulo schrieb:
2009/10/9 Juan E. Rodríguez jerdg...@gmail.com:
Does any one know about a SIP hard phone capable of sending SMS messages
(Or a SIP MESSAGE) that could be read from Asterisk dial plan??
The Gigaset S675IP series of DECT/SIP phone has SMS capability, but
not sure it can work
Mindaugas Kezys schrieb:
Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified
In the given example:
*ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000*
How do I interprete the jitter value ? Is the
Hello !
I 'm having a machine running asterisk 1.6.0.10 with IAX and dahdi.
The calls are going in and out from IAX2 to dahdi (chan_dahdi + libpri)
and vice versa.
After a period of time, I got the following scenario:
NOTICE[860] chan_iax2.c: I should never be called!
WARNING[752] channel.c:
Hello !
Are there any plans at Digium to include also german voice prompts ?
Thanks
regards
Hans
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Lee Howard schrieb:
fred wrote:
That’s being said, before going through the T38 Gateway tests, I’ve
tried first the Fax2mail and Mail2fax solution with (Hylafax +
Iaxmodem + Asterisk), to make a well-tested Asterisk solution working
and I’m already facing some problems. Receiving faxes is
made a solution with a simple web interface already?
Any suggestions would be welcome :-)
--
Andreas-Johann Ulvestad
Dagleg leiar, Unicornis
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Hello !
In order to chase after a problem I implemented the following dialplan to have
an
answertime of exactly one minute:
exten = xxx,1,NoOp(Test wait)
exten = xxx,n,Answer
exten = xxx,n,NoOp(Current timestamp:
${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)})
exten =
Hi, I have been trying to get a Wildcard TE122 card running here the
last couple of days.
libpri and zaptel are all installed and configured to E1 specs. The
jumper on the card is on, so configured for E1 (I'm in Norway).
When running zttool, I get 'Alarms: RED' on the single card installed.
Tony Mountifield wrote:
I have been asked by a potential customer whether we can connect an
Asterisk box to an Ericsson MD110 that has an E1 port with ISDN-USR.
They are unable or unwilling to upgrade their E1 port to QSIG.
Has anyone here had experience of successfully making such a
John Todd wrote:
Just a suggestion: have you tried more recent versions of Asterisk
with IAX2? I'm uncertain what version you're using, and if it's
1.2.4, that's getting to be quite old and the problems that you
reference may already be solved in more recent updates.
In addition, if
Hello !
I've upgraded our testsystem from asterisk 1.4.21 to asterisk 1.6.0.6.
We 've noticed that the log files are now in colour.
I could not find a note in the upgrade section about this.
Is this a feature or a bug ?
It might be usefull to have them not in colour.
best regards
Hans
Danny Nicholas wrote:
The log files themselves are not in color. It would be a style sheet change
on the GUI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Johann
Steinwendtner
Sent: Friday, March 06
Wolfgang Pichler wrote:
Hi all,
we have the following setup
PSTN 3 PRI Lines --- Asterisk (1.4.22) --- Siemens HiCom
--- Bosch Integral
The Asterisk Machine does play the man in the middle - and adds some
extra functionality to the system (SIP users...) - the normal calls
Gordon Henderson wrote:
On Mon, 29 Sep 2008, Andres wrote:
In other words, I'd really appreciate feedback from voip administrators
(not from resellers) who have had experience testing their devices and are
happy with them.
I would recommend the Linksys SPA8000 (8 port ATA). It is as
Kevin P. Fleming wrote:
Benoit Plessis wrote:
Is it possible on a TE220p to deactivate the hardware echo canceler at
will ? (With a function in the dialpan for example)
example for fax SDA ,beeing able to shutdown the echo canceler could
give better results don't you think ?
All echo
Florian Hackenberger wrote:
On Tuesday 13 May 2008, Steve Totaro wrote:
You can be shot several times and not die. I would try
resetinterval=never just to be able to to say Not the problem
rather than Probably not the problem.
I'll do that, although I'm pretty sure that the setting is not
Hi,
Thanks for your response. I've kinda worked around the situation, but
still need to do more testing on it.
Yuan LIU wrote:
From: Johann Hoehn [EMAIL PROTECTED]
Date: Wed, 28 Mar 2007 16:45:28 -0500
How do I clear a global variable for good? I have a situation of
needing to use global
status up the channel chain instead of down.
--
Johann Hoehn
Project Coordinator, Administration
Direct: 270-707-2040 x 4011
Ecommerce Corporation (www.ecommerce.com)
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Christoph Fürstaller schrieb:
Can someone explain what the parameters pridialplan and prilocaldialplan
are? What do they do and do I need them?
I've connected an asterisk box via E1 (sangoma) to an alcatel 4200 pbx.
The pbx technican complains about the format of the nr asterisk sends.
Hello !
I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal.
As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine.
Has anybody success with the HT486 as T.38 terminal ?
ATA as originator: I managed only onetimes a successfull T.38 fax
session. The other
Matthew Fredrickson schrieb:
On Oct 12, 2006, at 1:17 PM, Johann Steinwendtner wrote:
Hello !
I 've some questions how bridging of ISDN calls is done.
Assume an asterisk system with a TE405 card equipped.
(PRI1 - PRI4)
An incoming ISDN call on PRI1 is transfered back to
PRI3. Unless
Matthew Fredrickson schrieb:
On Oct 12, 2006, at 1:17 PM, Johann Steinwendtner wrote:
Hello !
I 've some questions how bridging of ISDN calls is done.
Assume an asterisk system with a TE405 card equipped.
(PRI1 - PRI4)
An incoming ISDN call on PRI1 is transfered back to
PRI3. Unless
Hello !
I 've some questions how bridging of ISDN calls is done.
Assume an asterisk system with a TE405 card equipped.
(PRI1 - PRI4)
An incoming ISDN call on PRI1 is transfered back to
PRI3. Unless there is DTMF detection or other things
involved, the bridging is done without Asterisk. Does
this
May be you can build an application which controls the background
terminal of the Meridian. (This would be a serial connection to the M1)
This application sends background commands like: se mw 3000.
This could be a try.
Best regards
Hans
Andrew Kohlsmith schrieb:
Please keep responses to the
Araklidas schrieb:
Yeah is true.but we have to sincronize this console command with
Asterisk SIP MWI
Regards.
Cris.
From: Johann Steinwendtner [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing
(it will silently unpause
them as well).
--johann
Douglas Garstang wrote:
I have a problem here, when an ACD agent is stuck in PAUSED mode.
As you can see from the outout of 'show queues' below, the agent 80014133 has
a status of paused.
Why is there a 'not in use' after the paused?
hestia
W - Waiting
C - Completed
A - Abandoned
SL - Service level(defined in queues.conf servicelevel value). Percentage of
calls answered within the time frame.
These numbers reset on reload or restart.
--johann
Douglas Garstang wrote:
Not documented anywhere that I can see. What are the W:, C
Sebastian,
This is possible and most likley the reason. To make sure, check the
location code of the cause IE in your ISDN disconnect message.
You have two options:
1) call your provider and describe your problem.
2) Change your provider
Best regards
Hans
Sebastian Reitenbach schrieb:
Hi,
Been running queues 24/7 and have 108 days uptime on the machine. However I did
restart asterisk two weeks ago due to a zap pseudo channel being stuck that was
created for app_meetme. We are using Asterisk 1.2.4.
--johann
Warren (mailing lists) wrote:
So let's cut to the chase here
that is misbehaving somewhere. Most of the SIP phones are Polycom
IP600/601s.
--johann
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play audio to the caller with the above.
--johann
Tristan wrote:
Hi List,
Just one more question that may sounds stupid to some people but I can't
find the solution for now,
I have the following dialplan:
exten = queue,n,Queue(myqueue)
exten = queue,n,NoOp(ENDQUEUE)
I don't
Are you using an idle webpage? If for some reason the phone can't reach the
page it will display an error and rebooting is about the only way to fix it.
--johann
Ken D'Ambrosio wrote:
Hi, all. Every now and then, some of my users get Error on their
phones. A reboot fixes it, but it's quite
will affect them regardless of them
being in multiple queues. The wrapuptime defined in queues.conf will only
affect their standing for a specific queue and won't have any effect on the call
distribution for any other queues they may be in.
--johann
by a separate extension. Unless the employee is aware of the no
answer == pauses, they may not know to unpause themselves later.
--johann
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I would have it invoke an AGI script.
[incoming_extensions]
exten = _X.,1,AGI(ManagerControl)
You could have the AGI script have it then jump out to some other
context,extension, or priority in the dialplan or have it handle the call itself.
---johann
Álvaro Palma wrote:
I'm developing
The BT guy should check LD 73 block LPTI and prompt AFF.
If it is crc then you need crc4 as well.
Best regards
Hans
Steve Totaro schrieb:
Andy Kirby wrote:
I am new to the group but have searched the doc's FAQ's etc before
posting here.
We are attempting tie our asterisk server/service
David,
You need to use the 'g' option with Dial().
g- Proceed with dialplan execution at the current extension if the
destination channel hangs up.
--johann
David L. West wrote:
In the following macro, a call is dialed and control branches according
to DIALSTATUS, much
.
Keep in mind that asterisk parses the contexts and extensions in a way you might
not expect. Use the CLI show dialplan context to see how the ordering ends
up. See
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting
for more info on it.
--johann
Álvaro
are for.
Keep in mind the UNIQUEID field will be the same for a caller as they go
through the queue. So the enterqueue, connect, complete actions will have the same.
--johann
Thermal Wetland wrote:
I am trying to figure out which one of our agents is answering the calls.
According to http
Yes, it is possible. I'm using PtP and TE mode at home with chan_misdn.
Hans
Ralf Mueller schrieb:
Hello,
can someone on the list confirm, that it is possible to connect a FritzCard to an
Anlagenschluss, when I use the mISDN driver?
I have read a number of posting and articles, that this is
I've noticed that when app_queue.so is reloaded(or just a reload command is
used) that all queue members that were paused are automatically unpaused. Is
there a workaround for this? (Note, I use statically defined callback agents).
--johann
Did you try rtpholdtimeout in sip.conf ?
Hans
Marco Mouta schrieb:
How do I report a Bug to Digium? or asterisk project?
On 4/19/06, *Doug Lytle* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Marco Mouta wrote:
I've tested maxexpirey=120 and even with this, asterisk didn't
How do you use the agents? Callback or on-hook? If callback you can direct the
calls to another context that doesn't have the fail over to voicemail.
--johann
Kyle Sexton wrote:
All,
I am experiencing an issue where if an agent is logged into the queue,
but has their client closed
Small update, I've been able to sort of work around the problem by making the
AgentcallbackLogin() direct to a context that in turn does another dial over a
local channel with the /n that gets around part of the problem. Still kinda
nasty seeing 5 channels around for 1 call...
--johann
])
exten = ,1,AgentCallbackLogin(1,s)
; join the queue
exten = ,1,Answer
exten = ,2,Queue(testing)
[queue]
exten = 1,1,Dial(Sip/4000||got)
exten = 1,2,Playback(beep)
exten = 1,3,Noop(Jump to the QA menu now)
Any ideas?
--johann
There isn't one that behaves the same. I resorted to storing stuff in astdb so
that I could use that to make the pause/unpause toggle.
--johann
Dov Bigio wrote:
Hi,
I wanted to use the same extensions for Pausing and UnPausing queue members.
Is that a variable that is set up
It's a unixtime stamp. It's the number of seconds since the epoch(Jan 1, 1970).
[EMAIL PROTECTED] wrote:
Hi,
How do I read (make sense of) the timestamp in the queue_log? I'm
probably just slow but I don't understand it.
Thanks!
Regards,
Jan
___
when that comes there will be
little if any chance for people calling in.
--johann
Gareth Blades wrote:
Cant you set the calleridname before putting the call into the queue?
On Thu, 2006-04-06 at 22:57, Shaun wrote:
I was wondering if it was possible to run a macro once the agent/member
picks
, Johann
/proc/zaptel/1:
---
Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS/CRC4
1 WCT1/0/1 Clear (In use)
2 WCT1/0/2 Clear (In use)
3 WCT1/0/3 Clear (In use)
4 WCT1/0/4 Clear (In use)
5 WCT1/0/5 Clear (In use)
6 WCT1/0
You could have something parse the output of /proc/zaptel/1 (depending on which
card you have you may have additional files there). If there is an alarm it
should be displayed there. This assumes you are using zaptel though.
--johann
Kevin P. Fleming wrote:
Wai Wu wrote:
Does Asterisk
Hi !
What is now the difference between a:
reload - (cli command reload).
restart - (I assume the application asterisk is restarted. o.k starting
from new)
sip reload - (cli command sip reload). Is sip reload part of the
reload command ?
Please confirm:
Which is the correct command when
callerid = asreceived
group = 1
channel = 1-15
channel = 17-31
--
Cheers, Johann
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That looks like the dialplan for Asterisk 1.0.x, The AstDB and other commands
have changed in Asterisk 1.2.x(and CVS HEAD). Check the UPGRADE.txt in the
source code directory of Asterisk to get the details on all the changes...
--johann
Andrew D Kirch wrote:
This is a follow/find me script
. Just keep in
mind if you have the agent default to both queues, they remove themselves from
one, then you reload Asterisk putting them back in both.
Reloading asterisk also undoes pause I've found...
--johann
nik600 wrote:
hi
if i have an agents that figure as a member in more than one queue
AgentMonitorOutgoing seems to do), just kinda flag them
so that they can be spyed upon. Any kind odd workaround using local channels?
--johann
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settings and
nothing is stored for ring volume. Polycom could add it in a future firmware
version if enough people requested it...
--johann
Anton Krall wrote:
Yep, that much I know but do you know which setting to use? Manual doesn't
mention anything.
|-Original Message-
|From
this from
within the dialplan.
Also it is possible to alter the format that ChanSpy() records in? It seems to
be hard coded to .raw (and lame/sox don't seem to like it for conversion).
--johann
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I can only guess, but I think I can remember that the creflen needs
to be 2 octets for qsig. Check what the Alcatel switch sends in the
setup message to *.
Anyway, why do use QSIG ? Does name display work on the * implementation ?
Best regards
Hans
P.S.: Schoene Gruesse an Kurt Krenn
Enable pass thru fax mode on the HT486, or enable ulaw in your SIP config.
Hans
Garth van Sittert schrieb:
Hi All
Is there any special configuration needed to send and receive faxes on
an ATA device?
I am using G711.a with a Grandstream Handytone 486. I can send faxes
from a fax machine on
ulaw was neccessary when pass through was disabled. What does a sip
debug tell you ?
Hans
Garth van Sittert schrieb:
I am using alaw and I have already enabled the pass through. Does alaw
and ulaw work?
I can fax out, but not receive faxes.
Garth
Johann Steinwendtner wrote:
Enable pass
with them...
--johann
Zeeshan A Zakaria
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Monday, February 06, 2006 12:52 PM
To: asterisk-users@lists.digium.com
Subject: SV: [Asterisk-Users] Help on queues
What kind of help do you need then?
Regards,
Jan
-1138045561.0.wav.
extension = 100,MeetMe(,r)
Is there something that I am missing to get this to work?
--johann
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Hello !
Asterisk 1.0.9 running on Linux 2.6.12.
I'm not able to call iax2 channels. There can be no translation path
found.
When I try to call from a ZAP PRI channel the following error occurs:
channel.c:1891 ast_request: No translator path exists for channel type
IAX2 (native 63488) to 72
on in the dialplan and just do an articial wait of XX seconds.
--johann
Adrian Carter wrote:
How would one go about incorporating that into the Dialplan ? I saw
those with the release of 1.2, but they seem to be more geared towards
'static' agents or the like... Not so much dynamic agents
But i
Dov Bigio wrote:
Hi all,
I have agents who are members of more than one queue.
When an agent is busy with queue A, he is not considered busy by queue
B, and receives call (since his EyeBeam Softphone has 6 channels).
Are you using the same AgentID for the person being on both queue A and
with an
external application using manager api to Pause agents on one queue when
they are busy on the other
That will likely work, however you may want to try one of the above first. I
can give you a better example of the above with more information off list.
--johann
Thank you
DOv
the only way to prevent the deadlock and in a
production PBX that accepts calls 24/7 that isn't acceptable...
--johann
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that
with the source I downloaded. Also show functions gets and error on
1.0.9.
That is a feature of the 1.2.x branch. You might want to upgrade :)
--johann
Sorry. It's in the doc directory. Thanks.
Thanks,
Michael
On 12/23/05, Michael Stearne [EMAIL PROTECTED] wrote:
From the console
with astdb and how it separates family and keys. We are running
Asterisk 1.2.1
currently.
--johann
Nicolás Gudiño wrote:
Is it possible from within the dialplan to determine if an Agent channel is
already a member of
a queue? Would like to use this as part of a check that will play a message
Checked the code and the queuename is not included regardless. I looked
at the public SVN and it appears to be the same there as well. So
I will have to come up with an alternative solution in the mean time.
--johann
Johann wrote:
I see what you mean and already have the option turned
connection to see, but this seems kinda overkill for something that should be
simple.
--johann
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. This breaks the blind transfers :(
Also tried putting, the below in sip.conf for the phones without success:
canreinvite=no
Any advice?
--johann
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...is there a way
to have Asterisk rotate the queue_log automatically or some otherway to
do it without losing data?
--johann
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*. The caller's hold time and the length of the call are both
recorded. The caller's original position in the queue is recorded in
origposition.
So, I take it the documentation is just wrong and the idea was never
implemented?
--johann
lenz wrote:
Hi Johann,
we engineered QueueMetrics out
in the wiki or documentation about it. Seems
like someone applied and patch and never documented what it did...
Anyone got any info to share? It doesn't appear to be applied to the
queue_log in anyway so it is of only use during realtime...
--johann
are running.
We are using callback agents. Here is an example log entry:
1133290480|1133290425.5|da_queue|Agent/1|COMPLETECALLER|35|20
Here is roughly what it should be:
1133290480|1133290425.5|da_queue|Agent/1|COMPLETECALLER|35|20|1
Any reason it doing what the documents say?
--johann
Make sure that you compile misdnuser with gcc3.x, gcc4 did
not work for me.
Hans
Yoann Le Bihan schrieb:
Jose,
I met so many problems these last 8 days that I don't remember exactly
which config was mine at that time, so I can't testify the answer...
(just for fun : my linux box is having 3
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