any transport, but it will be a while before most of
us are happy using the latest offering in production.
Alec Davis
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Definitely, 10 votes from me.
For the home user, 2xFXO + 6FXS, in a single slot small profile box is
ideal, but only able to offer 2xFXO + 4xFXS at the moment.
SIP phones don't exactly have the appropriate WIFE factor. A standard off
the shelve, no frills phone does the job.
Alec Davis
Having troble assigning rooms.
Kitchen
Lounge
Master Bedroom
Downstairs Family Room
Garage
Could easily get to 6, as we havent included the kids room or the home
office.
SIP phones don't work too well in a power cut, unless you've invested in an
8 port POE switch, as well as the UPS that we
Sorry not aastra, but the ideas in the following link may help, but the
specfic XML is for grandstream
http://www.voip-info.org/wiki/view/GXP-2000+XML+Idle+Screen
Follow the section titled 'Create indivudual GXP2000 Idle screens using
Apache2(mod_rewrite module), PHP and MySQL'
Alec Davis
We also need 'call deflection' and 'call redirection' using ISDN PRI (ETSI)
and currently using Asterisk 1.6.1.
If any one can point us in the right direction, or what is on the horizon
would be a good start.
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
is documented
by Richard Mudgett by his note dated 25/09/09, at the link mentioned above.
Regarding ISDN subadrress and associated costs, I can't speak for other
countries, but here in New Zealand, it's enabled by default, and doesn't
incur any extra costs that I'm aware of.
Alec Davis
countries are ISDN
subaddressesused ?
2009/9/26 Alec Davis siva...@paradise.net.nz
When did that happen? Added to libpri, someone beat me to it.
Hello Alec,
In this question, I was refering to your ongoing work
I'm sorry if this question let anyone believe this ISDN subaddresswas
already done
dialplan sees that CalledSubaddr was set to 8801 so dials 8801.
Regarding POTS it will not work, but some ISDN PABX's have the dialling
concept of dialling a '*' as the separator between Number and SubAddress and
'#' as end of dialling.
Alec Davis
-Original Message-
From: asterisk-users
Try 'pri intense debug span 1'
Used it last night.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, 1 October 2009 4:09 a.m.
To: Asterisk Users Mailing List - Non-Commercial
If your connected by ISDN at both sites, shortly the dialplan will support
'subaddress'.
The outgoing dialplan would look like
exten = 5145551212,Dial(DAHDI/g0/5145556000:7287)
For more https://issues.asterisk.org/view.php?id=15604
Alec
-Original Message-
From:
=13828#112881
If you find it cleans up you console from these messages please report back
your success or failure to the mantis bug.
Thanks Alec Davis
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Have you tried Digital calls over the B410P, using DAHDI and B410P as
bri_net_ptmp?
See https://issues.asterisk.org/view.php?id=16151
Useful when a business is E1/T1 connected, but need a BRI connection to a
remote warehouse, without needing a dedicated BRI line.
Alec Davis
-Original
at the following address,
http://www.voip-info.org/wiki/view/GXP-2000+XML+Idle+Screen
We do have some CISCO 7940's but never got around to doing the XML to play
the appropriate deck of cards.
Alec Davis
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
then do the equivalent ETSI methods to
reroute the call?
I may be able to test this over the weekend, in the mean time, I thought I'd
ask, if this was the correct way, or if mattf, rmudgett or others had 'team'
branch that is a work in progress that we can perhaps have a look at.
Alec Davis
It's now formerly a Mantis bug https://issues.asterisk.org/view.php?id=16339
John, can you please add to the bug.
I have just deployed this patch, so it needs a bit of time.
Alec Davis.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
://issues.asterisk.org/view.php?id=16409
If you are keen, please apply the patch and report back to either the list
or add a comment to the reported bug..
Alec Davis
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... announcement
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, 10 December 2009 9:30 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
... announcement
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, 10 December 2009 9:30 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
... announcement
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, 10 December 2009 9:30 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
On 1.6.1
Check out 'core show function QUEUE_MEMBER'
Don't have a 1.6.0 box anymore to check.
Alec
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Clark
Sent: Tuesday, 15 December 2009 12:59 p.m.
To:
straight from our 1.6.1 dialplan, don't know about 1.2.14.
exten = s,n,Set(LIMIT_WARNING_FILE=beep)
exten = s,n,Set(LIMIT_TIMEOUT_FILE=call-terminated)
;terminate after 1 hour, start beep warnings at 10 minutes, every 5 minutes
exten =
You may be able to use the SendText application, Conceptually and from
memory
exten = 971, 1, Answer()
exten = 971, n, SendText(Magnus 971)
exten = 971, n, Dial(SIP/971)
exten = 971, n,
exten = 975, 1, Answer()
exten = 975, n, SendText(Stefan 975)
exten = 975, n, Dial(SIP/975)
exten =
://issues.asterisk.org/view.php?id=15101, Tilghman published a
patch, that might also highlight other areas that are vulnerable.
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]--
Sent
from queue.conf
; UpdateCDR behavior.
;This option is implemented to mimic chan_agents behavior of populating
;CDR dstchannel field of a call with an agent name, which you can set
;at the login time with AddQueueMember membername parameter.
;
; updatecdr = no
I've never used it.
Did you get this resolved? And how if you did.
We've been have the same random PRI lockup issue for years now.
I've opened a mantis bug https://issues.asterisk.org/view.php?id=16713 and
hopefully we can get this issue resolved.
Alec
-Original Message-
From:
.
Could be related to more than just the above mentioned hardware.
Alec Davis
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To UNSUBSCRIBE or update options visit
, but the issue applies to all.
Alec Davis
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Search bugs.asterisk.org and enter 'digital' in the search field.
It probably will is my answer. I currently am not using it, so YMMV.
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Adams
voltage pulses, to light NEON bulb.
For the fully featured analog phones, that probably have CallerID
FSK MWI - the default mwisendtype.
SIP phones have their own subscription based VMWI.
And I'm sure more.
Alec Davis
I've been asked for recommendations for a small call centre, an ethernet SIP
deskphone with a wireless headset.
Similar approach would be a mobile phone with bluetooth head set.
Either I've not looked hard enough, or there isn't much on offer.
Alec Davis
period of the day.
Alec Davis
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Is the call successfull?
The 'Ignore polarity reversal on line seizure' may just be a warning.
What equipment, which Telco is the FXO card connected to?
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
gsm gw
On Apr 6, 2010, at 10:08 PM, Alec Davis wrote:
Is the call successfull?
The 'Ignore polarity reversal on line seizure' may just be a warning.
What equipment, which Telco is the FXO card connected to?
Alec Davis
-Original Message-
From: asterisk-users-boun
answeronpolarityswitch=no and hanguponpolarityswitch=no, this will give you
either dead air, or the caller will hear the CTU dialling out (which is
comforting).
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Many are having this problem.
goto http://issues.asterisk.org and search for 'bad magic number'
Notably, a few reports have come up in recent days.
Alec Davis
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Rose
I should have added, that if you havn't already, please report your senario
with example dialplan etc to one of the open bug reports related to you
problem, otherwise feel free to open a new one.
Also 'many' was a bit strong, should have said 'others'.
Alec Davis
_
From: asterisk
The following link may be a suitable workaround
How do I change the type of line from E1 to T1/J1 without using jumpers?
http://kb.digium.com/entry/121/
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
I filed the following bug on the 28th of May.
0017371: [patch] [regression] DAHDI analog FXS port segfaults after dialling
2nd DTMF digit
Please see https://issues.asterisk.org/view.php?id=17371
You problem sounds the same, if it is the same please report this on the
bug.
Alec Davis
Call progress (is only experimental), relies on defined ring tones, coloured
ring (music) messes this up.
in chan_dahdi.conf
callprogress=no
busydetect=yes
busycount=4
and possibly if your incoming analog lines support it.
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
_
was open.
In mysql.conf:
[general]
autoclear=yes
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder
Sent: Thursday, 16 September 2010 4:49 a.m.
To: asterisk-users@lists.digium.com
Subject: Re
Make sure you have allowguest=no in your sip.conf, the default is yes,
unless you really do want anonymous guests.
Also it might pay to consider
http://www.emergingthreats.net/index.php/rules-mainmenu-38.html
Alec Davis
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk
In another email I've just responded to, it might pay to consider
http://www.emergingthreats.net/index.php/rules-mainmenu-38.html
Alec Davis
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Sunday, 3 October
A DNS cache on your asterisk box may be the answer.
Google Asterisk DNS Cache, many hits.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Alejandro Imass
Sent: Tuesday, 23 November 2010 6:18 a.m.
To:
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?
I'm
, to find that the answer isn't there yet.
Note: Flaming is not an acceptable behaviour :)
Alec Davis
PS. Sorry to the asterisk-dev list that have seen this already, posted in
wrong forum.
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of the rules.
Nearly top posted again, it was hard not to... Until I found what causes
Outlook to mess up formatting replies.
If reply indent option is enabled, and if message is received in HTML
format, need to disable HTML format (Send Plain Text)
Alec Davis
through but always has some
lines missing.
The issues above could be asterisk's problem (1.8 SVN branch) or the
client's.
What I'm obviously seeking is a known good working free Windows XP - Windows
7 Fax add-in for Microsoft Fax and Scanning.
Alec Davis
Making an assumption here, I'm sure I cleared the remaining resequencing
issues up in 1.4 SVN and 1.6.2 SVN.
https://issues.asterisk.org/view.php?id=19032
The issues I uncovered and fixed were when a new voicemail is left, while a
mailbox is open for review and the user deletes a message.
Alec
.
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Leif Madsen
Sent: Friday, 29 April 2011 12:03 p.m.
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Discussion: Are we
Are you not seeing issues with *8 call pick up then ?
--
Thanks, Phil
https://reviewboard.asterisk.org/r/1185/ helps with *8 pickup issues,
particulary when you have pickupsounds enabled.
Alec
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I apply sig_pri.c patch for 1.8.3.2 manually. (the patch
can not apply to 1.8.3.2 or 1.8.4-rc3).
but the situation is the same. do I need to play with other
options with the patch? or I need newer asterisk versions to
solve the problem?
thanks a lot for information!!
What does
https://issues.asterisk.org/jira/browse/18998
https://issues.asterisk.org/jira/browse/18998 may have the answer,
particularly the patch bug18998-1.8.2.3.diff.txt
Alec
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
ringing extensions.
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Sebastian Arcus
Sent: Tuesday, 21 June 2011 11:10 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
IMHO, blind tranfer definition is to NOT connect A and B back
That is correct, and is why it's called a 'blind' transfer;
the transferring party does not know or care what happens to
the call after effecting the transfer.
That's not what users migrating from some legacy PBXs expect,
tripping over
a cable.
Alec Davis
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Most *8 pickup issues have been fixed in trunk. May have made it into 1.8.5,
I'm not sure.
https://issues.asterisk.org/view.php?id=18654 and others search mantis for
closed issues and 'pickup'.
Or newer https://issues.asterisk.org/jira/secure/Dashboard.jspa
Alec Davis
-Original Message
Beat me to it.
There are other commits that follow up from 18654 that may also help.
Check the blame's for changes to apps/app_directed_pickup.c and
main/features.c
Alec
-Original Message-
From: asterisk-users-boun...@lists.digium.com
That works for us with GXP2000's and GXP2010, but not the later HD series
GXP21XX.
Alec
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Mike Diehl
Sent: Friday, 22 July 2011 10:50 a.m.
To:
as phone1 and phone2.
Alec Davis
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Thanks for the confirmation. Too bad Dahdi doesn't provide
call supervision so that Asterisk knows if/when the callee
has answered.
I'll experiment and see how it goes.
DAHDI with an FXO card can support call answer/hangup supervison.
Check out chan_dahdi.conf options;
make sure the option '|60' is only included after the devices, IE. at the
end of the dial.
Dial(SIP/13365551212@8x8
mailto:SIP/13365551212@8x8SIP/13365541212@8x8SIP/13365531212@8x8|60|dgF(c
allFlo-in^3^1)M(record^39ff6274-c0f0-453d-aa05-402a7bd6d567
Did you try to make dahdi-tools before installing newt_devel ?
pre-requisits are on debian;
libncurses5-dev
libnewt-dev
I'm guessing as to what the problem might be, but I think that it now still
complains that libnewt-dev needs to be installed.
I see this regularly when building a
is the lastest.
Alec Davis
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Rojas
Sent: Saturday, 14 January 2012 3:37 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk problem sip
Hi everybody
I
This maybe not what you want.
Our solution was monitor a queue with a BLF, instead of a queue member
This review https://reviewboard.asterisk.org/r/1619/ allows a BLF lamp to
flash when a queue is ringing, then the queue can be picked up by the BLF
button.
Alec
-Original Message-
How can I test this solution on a 1.8.8.1 system ?
If I'm not mistaken, diff
https://reviewboard.asterisk.org/r/1619 do not apply to 1.8.8.1.
I've just checked out 1.8.8.1 and download my patch from
https://reviewboard.asterisk.org/r/1619/diff/raw/ and it applied clean,
using the following
-Commercial Discussion
Subject: Re: [asterisk-users] Pickup calls coming from queues
Am 23.01.2012 um 23:25 schrieb Alec Davis:
How can I test this solution on a 1.8.8.1 system ?
If I'm not mistaken, diff
https://reviewboard.asterisk.org/r/1619 do not apply to 1.8.8.1.
I've just checked
I want to make a new file of CLI log everyday. So I just make a
shell script in asterisk log directory. My file is working fine and making
new file with the name of full_2012-01-27. But strange I noticed that
asterisk is updating my newly crested files even i don't reload asterisk.
);
@@
Alec Davis
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Saturday, 28 January 2012 8:03 a.m.
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] TCP transport and BLF
I have some
Simply, without checking for BUSY, DND or TIMEOUT
I'm assuming each ring period is 3 seconds.
exten = 8512,1,Dial(SIP/8512,15)
exten = 8512,n,Dial(DAHDI/GO/101233456,15)
Or another way.
Maybe the FollowMe application, allow multiple numbers to be tried, each
after a configured timeout.
from
the system running at the moment with the previous kernel, so
nothing wrong with the B410P.
I'm hoping that 'tomorrow' that adding the line below to
/etc/modprobe.d/blacklist.conf will fix my lockups.
blacklist hfcmulti
Alec Davis
_
From: asterisk-users-boun...@lists.digium.com
Any one having this problem.
The Grandstream Firmware revision is 1.2.5.3.
We have the registration time set to 5 minutes, and every time after a
reboot, the BLF's will initially indicate the correct state, then stop
working a few minutes later.
The workaround has previously been to reboot
Create a field called 'dnid', this then is the original called number, no
matter now much you jump around with contexts.
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Daniel Knoll
Sent
The TDM410P doesn't support 'hvac', only the obsolete TDM400P supports that
option was for the old phones that have a neon light (or equivalent
LED+ZENER ciruit).
Are other phones off the TDM410P (other than the VTECH) working, or is the
Vtech the only model with VMWI available to you.
I'm not
Sorry, I didn't word my reply correctly, I wasn't trying to say 'hvac'
support on the TDM410P would fix the issue.
As I think about it more, having the option enabled there doesn't matter,
even for a TDM410P.
Shaun: Regarding neon patch (review 1144) for wctdm24xxp, I'll have to get
on to that.
-Original Message-
From: Niccolò Belli [mailto:darkba...@linuxsystems.it]
Sent: Monday, 16 April 2012 4:21 a.m.
To: asterisk-users@lists.digium.com
Cc: siva...@paradise.net.nz
Subject: Re: [asterisk-users] Pickup calls coming from queues
Il 20/01/2012 20:32, Alec Davis ha scritto
.
;
;hanguponpolarityswitch=yes
;
Alec Davis
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Have a look at the latest blacklist sample in dahdi trunk
http://svnview.digium.com/svn/dahdi/tools/trunk/blacklist.sample?view=log
file: blacklist.sample
...
# Some mISDN drivers may try to attach to cards supported by DAHDI. If you
# have a card which is *not* supported by DAHDI but supported
-responsive, 60 of them.
Removed the bindings, and left only one.
Reprovisioned the phones again, all 60 of them. All is well.
Have I tried again. No.
Did I have enough time to debug. No.
Sorry.
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
then allows a call via a BLF button to pickup a call from a
queue.
Alec Davis
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I've seen this post. That's why I thought it was possible.
I'm using 1.8.11
What is the difference between this post and asterisk 1.8.11 ?
The patch hasn't been accepted by the community, thus isn't in asterisk
trunk or any asterisk branches.
Alec
--
-Original Message-
From: Alec Davis [mailto:siva...@paradise.net.nz]
Sent: Saturday, 18 August 2012 10:36 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] BLF and Call Queues
I've seen this post. That's why I thought
Thank you Alec.
is there also some kind of manual on how I can use this
patch ? I don't want to mess up a perfect running system.
If you're asking how to apply this patch?
Download patch to your working src directory
Example:
cd /usr/src/asterisk
patch -p0 download.diff.txt
make
make
Do you also know why it hasn't been accepted ? Seems like this
functionality is asked for on different forums. Wanting
to watch a
queue for calls is not that strange.
Not sure why?
Maybe I didn't promote it enough.
Maybe my examples aren't simple enough.
So I'm just looking on how to make a BLF-button blink or turn
red, to show to my customer that there are still calls inside
the queue waiting.
Can I only apply on Asterisk 1.8.5 ? Or can I apply to my Asterisk
1.8.11 also ?
It's 4 lines, plus 2 debug statements.
I haven't had time to
So I can just add these 4 lines to app_queue.c and this will give me
the ability to use : exten = 566,hint,Queue:voipq1 ??
Yes, then I assume you know that you need to compile etc.
./configure
make menuselect
make
make install
Alec
--
returned, yeah.
But for an ordinary user (in my opinion) it is
cumbersome and unfriendly.
As a developer/committer, I also agree.
Alec Davis
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New
can I avoid this?
Incoming channel type is ISDN (mISDN).
Are you saying every digit twice, or some digits twice.
Where is the call originating from, GSM cell phone or landline?
Which version of asterisk are you using?
Alec Davis
: Friday, September 14, 2012 9:24:41 PM
Subject: Re: [asterisk-users] DTMF digits falsely detected
On 9/14/2012 6:04 PM, Alec Davis wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Vieri
Sent
[2012-09-15 22:36:44.489226] DTMF[1706] channel.c: DTMF end
'4' received on
SIP/alec-0009, duration 1660
Alec,
Interestingly in your log DTMF durations are even greater
than in my original sampling. Well, maybe my duration
theory is not
And just to make sure. In both scenarios, normal digit press
and prolonged digit press, you did not reproduce the problem
we are discussing with X-Lite. Is that correct?
Correct, everything with X-Lite 3.0 and asterisk 1.8.16.0 worked correctly
with short, normal and long key presses
calls coming from queues
Il 25/01/2012 22:52, Michael Keuter ha scritto:
Outcry! :-)
I'm outcrying too :)
--
Ok. Didn't make 1.8 or 10.
Hints for a ringing queue and queue available has been commited to
asterisk-11-beta2 and trunk.
Alec Davis
exist to allow pickup of a queue.
Queue hints code has now been submitted to asterisk-11-beta2 and trunk.
Alec Davis
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A long shot but how about 'campon' a queue, available on most old phones
systems but not asterisk.
Well maybe will still apply
https://issues.asterisk.org/jira/browse/ASTERISK-460
-Original Message-
From: asterisk-users-boun...@lists.digium.com
The options are;
dtmf_reverse_twist
dtmf_normal_twist
relax_dtmf_reverse_twist
relax_dtmf_normal_twist
Initally I'd set all to 100, you may get talkoff when on a call, but atleast
you'll know if CID is working.
Then set back to the appropraite standards of Ukraine, ETSI ATT etc.
Alec
are also the defaults.
dtmf_hits_to_begin=2
dtmf_misses_to_end=3
Alec Davis
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google function IAXVAR
It allows you to pass any variable you like between 2 boxes.
Alec
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Saturday, 22 December 2012 6:59 a.m.
To: Asterisk Users Mailing
)
exten = s,n,Set(penalty=0)
exten = s,n,Set(stateinterface=SIP/cisco1)
exten =
s,n(queue-add),AddQueueMember(${queuename},${interface},${penalty},options,,
${stateinterface})
And to remove the member;
...
exten = s,n(queue-remove),RemoveQueueMember(${queuename},${interface})
Alec Davis
I've not tried to publish device state with XMPP yet but I've
discovered this issue
https://issues.asterisk.org/jira/browse/ASTERISK-18078
I'm planning to install my XMPP server on the same machine as
one asterisk server so hopefully, I won't be hit by the issue
above but have you met
why they got the call - assuming they are in close proximinity.
Large queues, different story. We only have 5 members in a queue, so it
works for us.
Alec Davis
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