cationmap]
zapflash => *0,callee,flash,()
--
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blog.polybeacon.com
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On Fri, Jun 25, 2010 at 7:25 AM, Eyal Goltzman wrote:
> How can I trace\debug my dialplan?
>
*CLI> dialplan show 1...@context
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polyb
On Thu, Jun 24, 2010 at 2:56 PM, Zeeshan Zakaria wrote:
> Its possible but not easy. Search for n-way conferencing on voip-info.org,
> it has all the details on how to do it.
Or you could post the direct link:
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
--
Paul Belanger
w.php?id=15004
[patch] Add voicefile and dtmf options to res/res_agi.c
https://issues.asterisk.org/view.php?id=15531
[patch] MGCP Business Phone Packages patch
https://issues.asterisk.org/view.php?id=15159
[patch] chan_mgcp new feature: digitmaps definitions
https://issues.asterisk.org/view.php
the relevant information?
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blog.polybeacon.com
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On Tue, Jun 22, 2010 at 5:47 PM, wrote:
> Can i join 2 dahdi (meetme) channels from different servers?
>
No
--
Paul Belanger | dCAP
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Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeac
sk boxes together, but each conference will be
hosted on there respective server.
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Paul Belanger | dCAP
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blog.polybeacon.com
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On Mon, Jun 21, 2010 at 12:25 PM, RSCL Mumbai wrote:
> What is the simplest way to achieve this ??
>
Use the transfer button on your phone?
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On Sat, Jun 19, 2010 at 5:21 AM, michel freiha wrote:
> Waiting your reply
>
Reply: Do not cross-post to #asterisk-dev
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
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reat a performance factor, call arrival rates
are more relevant, the CPU time is spent setting up and tearing down
calls. Simply having calls in progress with no transcoding uses a tiny
amount of CPU in comparison to the work involved setting up and routing
--[ UxBoD ]-- wrote:
>>
> Would be nice if the VPN support could be back ported to the 360s.
Never going to happen, there isn't enough flash memory to store the
code. The Snom370 has had OpenVPN support for quite a while though.
Yeah sounds like you wanna use NVFaxDetect
it would allow you to add something like exten => fax,1,Swift("number
has changed"); to your inbound call part of your dialplan
On Jan 14, 2010, at 11:45 AM, Juan C. Villa wrote:
> Could you use NVFaxDetect?
>
> On Thu, 2010-01-14 at 17:35 +, --
i want to thank all the developers at asterisks .many many thanks to alain
chaipainos at netbricks /enea
work hard an dimplement the machine.
happy hanuca to all of you
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as
ore I just gave up.
I'm sure there is a better or proper way of handling this. I'm
interested to hear it.
Paul
On Nov 20, 2009, at 3:41 PM, Edwin Lam wrote:
> hi folks.
>
> we've experienced some weird problems lately. we have about 600
> SIP phone on a single
Have you considered putting an advertisement in the newspaper?
PaulH
On 05/11/09 06:43, Carlos Cuervo wrote:
> Hello,
>
> I've been tasked to look for ways to resell to others the service that
> one of a trunk provides.. In other words, i want to configure my
> current Asterisk (Ver. 1.4.26.1)
On 29/10/09 22:40, Matt Riddell wrote:
>
> :D
>
> I should hope not!!
>
> If everyone was as smart as me, how would I take over the world?
>
>
With violence, just like everyone else!
PaulH
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I have used both misdn and dahdi_bri over the last year, and would happy
take dahdi if for no other reason that it's much easier to install.
A patch is available to allow dahdi_bri to work with Asterisk 1.4, and I
have used that successfully.
PaulH
On 25/10/09 03:26, Olivier wrote:
Hello
On 24/10/09 00:59, Lyle Giese wrote:
PATRICK KANGETHE wrote:
I want to interface asterisk with a legacy pbx that has around 23
extensions through my 8 fxs card, how do i work around this?
Hint: I have already terminated 8 extensions from the legacy PBX, i
was thinking whether i can peer the ext
Jeff LaCoursiere wrote:
>> Steve Edwards wrote:
>>
>>
>>> Since I'm an "old-school" C programmer, I use emacs as my editor. I fire
>>> up gdb (the GNU C (amongst other languages) debugger) in a window, give it
>>> a command like "b main; r >> through my program line by line, examining and cha
The list will need to see your dialplan or a CLI dump to help you with this.
PaulH
B.Masoud @ SH wrote:
>
> I am not sure why I am getting this message,
>
> I have an outbound route that goes to asterisk gateway1 then asterisk
> gateway2
>
> When all lines on asterisk gateway1 are full, I get t
I have used the group function to limit the calls entering a queue for a
similar reason to yourself.
PaulH
Niccolò Belli wrote:
> Hi,
> I explain what I want to do..
> All the operators share their phones. The number of the operator isn't
> constant, so it's possible that two operators share al
Is inbound working?
Can you see action on the CLI when you send a call to the lines attached
to the card?
PaulH
B.Masoud @ SH wrote:
> Hi
> I installed TDM24 card, made ZAP (DAHDI) trunk, and set outbound all calls
> to that trunk, I am getting all circuits are busy now, do I have to do
> some
bump... Anybody using MeetMe hints for busy-lamps on phones? Anyone
seeing this issue?
On Mon, Sep 28, 2009 at 11:08 AM, Paul Dugas wrote:
> I suspect the issue I'm having is more specific to the MeetMe app but
> that's just a guess. app_meetme.c (in 1.6.1.6) calls
>
>
) was where it would look for
the "o" extension. Briefly looking through app_meetme.c frim 1.6.1.6,
it seems to fit. But you have "sip" for both the vm-context and the
dial-context. Weird.
P
--
Paul Dugas -- Computer Engineer -- Dugas Enterprises, LLC
522 Black Ca
Death to all sip users!
Paulh
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exten => *,1,NoOp(Attendant: Directory)
exten => *,n,Directory(default,attendant,eb)
exten => *,n,Goto(s,1)
exten => o,1,NoOp(Zero)
exten => o,n,Goto(0,1)
exten => a,1,NoOp(Star)
exten => a,n,Goto(0,1)
Paul
--
Paul Dugas -- Computer Engineer -- Dugas Ente
http://www.noojee.com.au/Page/NoojeeClick
Built for firefox.
PaulH
Stefan Schmidt wrote:
> Hello,
>
> iam searching for an Firefox plugin which can make an sip Invite and
> Redirect after 200 OK, so i dont have to use a softphone, just to
> initialise a call by clicking on a number
>
> i've fo
ded by http://www.api-digital.com --
>
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> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Paul Du
uot; },
> { AST_EXTENSION_ONHOLD, "Hold" },
> { AST_EXTENSION_INUSE | AST_EXTENSION_ONHOLD, "InUse&Hold" }
>
> So a line Is Unavailable on 3 conditions, but only Idle on one; unless you
> tweak to make UNAVAILABLE equivale
: MeetMe:${EXTEN} State:Unavailable
Watchers 0
I'm wondering why they're Unavailable instead of Idle. They go to
"State:InUse" when active but usually return to Unavailable when the
conference ends. Occasionally they end up in InUse but not
consistently.
Anybody know why?
Paul
--
Pa
ver if server_ip is of same server
>
> i have same kind of problem but still dont found proper solution
>
> in,fact i need dialing on IP base in which dialing by using IP address
> will send call to remote machine or same machine
>
> regards
> Dhaval
>
> On Fri, Sep
I have used the SIPPEER function to find if a phone is local and
available before.
PaulH
Asterisk User wrote:
> Hi,
>
> I have a generalized syntax for dial application in my dialplan where
> I send calls to particular server.
> Here is my dial sysntax...
> exten =>
> _x.,1,Dial(${Dial_techn
(${calleridn...@default)
exten => 300,1,MeetMe(100)
exten => 678,1,Goto(ivr,s,1)
[macro-extensions]
exten =>
s,1,MixMonitor(exten_${ARG1}-from_${CALLERIDNUM}-${TIMESTAMP}-${UNIQUEID
}.wav)
exten => s,2,Dial(Sip/${ARG1}|20)
exten => s,3,Voicemail(${ARG1}|u)
exten
It's easier to work with the closed hours then - use a goto just for
Sunday/Monday
PaulH
James Hankins wrote:
> Greetings folks, new to this, trying to get the syntax correct for a
> day of week routing.
>
>
> exten => 345,1,Answer()
> exten => 345,n,GotoIfTime(10:00-17:00|tue&thu&sat|*|*?op
Has anyone seen this one before?
full:[Sep 14 11:49:01] DEBUG[15771] chan_sip.c: SIP attended transfer:
Error: No target channel
It coincided with a failed attended transfer...
Ideas?
PaulH
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I would strongly suggest you browse:
http://www.asteriskdocs.org/
Kind regards,
PaulH
Songtao Yu wrote:
> Hi All,
> I am new to Asterisk. Now I got one question on the identifier
> parameter of the Dial() command. I saw as below:
> exten => 20,1,Dia(Zap/3/5551234).
> Would you please let me k
I have also seen:
PSTN asterisk legacy
Which also gives you a migration path
PaulH
Research wrote:
> Hello team;
> While am aware and active user of astersk monitor function for
> recording, i would like to know if i can use asterisk as a pure
> recording server(like nice or wit
Matt Riddell wrote:
> On 3/09/09 11:34 AM, Paul Hales wrote:
>
>> Hmmm.any idea how I can use hints to monitor their mobile phones?
>>
>
> Unless the call came in via Asterisk, you can't.
>
>
The calls will - so it should be able (at the very
n => s,n,hangup
>
> The AGI checks the hint for 100 or 101 and assigns CELLLINE to call the
> cell. If either is in use, LINESTAT is set to BUSY, otherwise set to AVAIL.
>
> -Original Message-----
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-bo
nsion set/reset the has_mobile property in the AstDB.
> You could then call Local/1...@gaents directkly or make it a member of
> the queue (with known issues on some version of *) :-)
> l.
> 2009/9/2 Paul Hales <mailto:pdha...@optusnet.com.au>>
>
>
> A situation where
to be queued, you create a queue with only one member, and
> have agents log on and log off as necessary; if you don't want callers
> to be queued, likely I would not use a queue but woul dial the agent
> straight.
> l.
> PS. this is quite an unusual requirement, what is it for
But they do taste similar.
PaulH
Darrick Hartman wrote:
> Polycom sip.cfg is not the same as the Asterisk sip.conf file
>
> hadi motamedi wrote:
>
>> Thank you for your reply . Please find attached my Asterisk sip.conf .
>> Can you please let me know what modifications are needed ?
>> Regar
Miguel Molina wrote:
> Paul Hales escribió:
>
>> I have a _very_ specific situation where I need queues to work in a very
>> specific manner - I need the queue to only accept one call at a time,
>> even though several phones are attached to it.
>>
>> My me
I have a _very_ specific situation where I need queues to work in a very
specific manner - I need the queue to only accept one call at a time,
even though several phones are attached to it.
My memory tells me that queues might have even worked this way in the
distant past (pre 1.0)...but I am wil
I couldn't find any information on this brand of phone on the internet
at all.
PaulH
hadi motamedi wrote:
> Sorry for lack of enough information . I mean my subscriber when goes
> off hook he will see his own number displayed on his phone . I need to
> disable this feature on my Asterisk .The p
Matt Riddell wrote:
>
> What is a subs?
>
>
A submarine. I think.
PaulH
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asterisk-users
Sticky Park sounds like somewhere you go late at night wearing a plastic
raincoat.
PaulH
Mat Murdock wrote:
> My company for various reasons has asked that I come up with a way to
> have previously parked calls be re-parked in the same parking slot. I
> have looked at setting up asterisk so
sterisk doesn't send RCTP keepalives while on
hold? Is this possibly a bug?
Cheers,
Paul Herman
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Use a standard network cable - but you have to activate the 'terminate'
jumper on the NT end.
- Also, the new BRI stuff in dahdi is much easier to work with than misdn.
PaulH
voip crazy wrote:
> Hello all,
>
> I'm trying to conect two asterisk servers using two B410p Digium
> cards. One card o
Can I assume that you meant to add that the person's phone would be used
to listen to the message?
PaulH
Olivier wrote:
> Hi,
>
> I've lastly read a Request For Quotation asking for a software option
> I've never heard about before.
> It's about a player plugin with which, when using an Outlook
I have an MP114 2fxs,2fxo which I would like to use with Trixbox, does
anyone have a setup file they can share to help me work this out.
Instructions or a link I can follow - thanks.
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In australia, I would usually suggest a mix of E1 and SIP for calls - it
doesn't cost any money to receive calls via E1, and redundancy is an
old, valuable friend of mine.
PaulH
Stephen Fierbaugh (PBT) wrote:
> I am a Linux sysadmin who has been tasked with developing the phone
> system for ou
Can I assume that your project has stalled?
PaulH
logan wrote:
> Thanks Paul. Your help is much appreciated here.
>
>
>> I don't really understand this question - Asterisk can make calls over
>> phone lines. And it does it well.
>>
>>
> Sure
Sadly, at the end of the day the answers will probably be no, no, no and no.
PaulH
logan wrote:
> Hi,
>
> I'm an absolute newbie and wanted to know the following.
>
> I want to have a setup where I have a PSTN line connected to my
> Asterisk box and want to know if it is possible to make more t
logan wrote:
> Thanks Paul. Your help is much appreciated here.
>
>
No problem - been working on telephone systems for about 12 years now -
which doesn't even make me an old hand...
> Surely, Asterisk does that well, but Asterisk needs to have multiple phone
> lines for th
Some thoughts inline:
logan wrote:
> Hi Paul,
>
> Thanks a lot for the response.
>
> I'm a novice so pardon me for the stupid questions. I thought that maybe the
> PSTN lines don't allow more than 1 simultaneous calls on a line, but on GSM
> it might be possibl
logan wrote:
>
> Hi Trevor,
>
> Thanks for the response.
>
> I was thinking if a GSM to VoIP gateway can do the job of multiple outgoing
> calls. I could be wrong but seems like cellphones do allow you to make
> multiple calls at a time (only one is active or one active conference). If
> we assu
Always a great readthanks.
PaulH
Alex Balashov wrote:
> Inspired by AG Projects' Adrian Georgescu's post of Eric S. Raymond's
> classic "How to Ask Questions the Smart Way" to the OpenSIPS-users
> mailing list[1], I'm going to repost it here:
>
> http://www.catb.org/~esr/faqs/smart-q
Yes, but not as your first Asterisk implementation.
PaulH
gergis.rasmy wrote:
> i am asked to implement a call center of 50 seats for my company , and
> i was wondering if Asterisk can fit this as a relaibale and low price
> system
>
> is it mature enough for this task??
>
> best regards
>
I have a problem with one way audio on Sip and I guess it may be a NAT
issue, in the example below 204 is rung by 208 (xlite external)
I dial perfectly but when I get to the answering of the Asterisk, I can
hear audio from the Asterisk but cannot get audio to the Asterisk, ie If
I ring the voi
'One touch park' was designed to work around this issue.
PaulH
Danny Nicholas wrote:
>
> Hi gang,
>
> When I try to park a call using blind-transfer (#1), the caller hears
> the lot instead of the transferring party. Attended transfer and blind
> transfer from the phone buttons (Polycom 501) wo
gt;
>
>> It appears that that option is set
>>
>> from queues.conf
>>
>>
>> [ops]
>> musicclass = default
>> strategy = leastrecent
>> timeout = 5
>> retry = 1
>> wrapuptime= 3
>> autofill = yes
>> autopause = no
>&g
The queue option
ringinuse = no
might be what you are looking for.
PaulH
Kev Szaszvari wrote:
> Hi All
>
> I am using asterisk 1.4.21.1
>
> Im not sure if this is a issue but it has become one for me :)
>
> When agents are logged in to a queue (AgentCallBackLogin) and they receive a
> direc
I can definitely recommend the 'sit down and play with it' website.
Worked for me.
PaulH
David @ULC wrote:
>
> What the best website and book to start learning asterisk ?
>
>
>
>
> _
>From memory, it is doable but this is a feature that Polycom never quite
finished writing.
PaulH
On Fri, 2009-06-19 at 10:58 +0200, Louis-David Mitterrand wrote:
> Hi,
>
> Is there a way on Polycom phones to show an agent whether he is logged
> in or not?
>
>
this could be happening?
Thank
Paul
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Farooq Hussain wrote:
> Dear All,
>
> Please help as I new to Asterisk. I want know something what is Trunk
> what it will do. And I want to create a Dialplan like bellow:
>
>1. It ask to dial a extension.
>2. User will dial a extension.
>3. User will be routed to that extension.
>
> I
Todd S wrote:
> What's the bets way to verify T.38 is being used on both incoming an
> outgoing transaction?
3 to 1 in favour of not working. ;)
PaulH
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Alex Samad wrote:
> Hi
>
> I have an account with mynetphone (australia), which gives me two voip
> (sip) accounts, which i used to have connected to a spa9000.
>
> this is behind a firewall, so on the spa9000 I would listen on another
> port apart from 5060. so on the firewall 5060 would go to vo
Digium PSTN cards seem to work.
PaulH
Manoj Panicker - FOES wrote:
>
> Hi
> Which is the best interface card to connect* PSTN* line with
> Asterisk. Can somebody please help. My intention is to route the
> incoming PSTN calls to internal IP Phones through Asterisk and Vice
> versa. The
Not true. I am always wrong.
(wait...is that a paradox?)
PaulH
ContactTel Business wrote:
>
> Niecly said.. hoeever, these list are not for astrix users, butt for
> bashing, didnot you realise this ?
>
> It had where 4 years more , know that this is fluent in this site.
>
>
>
> Translated as
Alex Samad wrote:
> On Thu, May 14, 2009 at 03:18:28PM +1000, Paul Hales wrote:
>
>
>> What happens if you make a call in from the old fax line and send that
>> over to the old PABX? Does that work OK?
>>
>
> not sure what you are asking here. I have check
Alex Samad wrote:
> On Thu, May 14, 2009 at 03:31:18PM +1000, Paul Hales wrote:
>
>> Have you tried plugging analog phones into the FXS ports in the Asterisk
>> box?
>>
>
> good ideal, but trying to find an old style phone the site has a
> commander P
Have you tried plugging analog phones into the FXS ports in the Asterisk
box?
That should let you know what the Asterisk is really doing with it's FXS
ports.
PaulH
Alex Samad wrote:
> On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote:
>
>> I think you have your li
Alex Samad wrote:
> On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote:
>
>> I think you have your line types mixed up - FXS is for phones, FXO is
>> for lines.
>>
>
> sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is
> that a
I think you have your line types mixed up - FXS is for phones, FXO is
for lines.
An analogue passthorugh setup _is_ doable, just not overly recommended.
PaulH
Alex Samad wrote:
> Hi
>
> I am in the middle of move a small business over from legacy PABX + PSTN
> lines to VOIP infrastructure.
>
>
I got very excited when I read the title of this email - I was hoping
someone had learnt to speak g729.
Ah well.
PaulH
Adrian Marsh wrote:
>
> Hi,
>
> I’m having problems with an asterisk server that’s not offering Codecs
> for ulaw and alaw as it should.
>
> I’ve three servers in total: a1, a
George Kwabenah Appiah wrote:
>
> Are there (??) instructions for people who are experienced at the
> Trixbox level but wish to move on?
>
>
> If you'd like to get a solid foundation on Asterisk and how the
> various pieces fit together, I suggest you invest a couple hours and
> go through
You don't need freepbx.
In all honestly, building a system from scratch isn't too bad if you
having some decent (or indecent?) linux karma.
If you don't, it's going to be a rather unfun time.
PaulH
John F. Ervin wrote:
> So, people have recommended building a system from scratch, start with
>
Steve Howes wrote:
> On 28 Apr 2009, at 16:49, Daniel - Asterisk wrote:
>
>
>> Dear all,
>>
>> I wanna know what can I do to get the PBX's clock from
>>
>
>
> You sir, are made of fail.
>
>
I had to admit, I laughed.
PaulH
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I am still not sure what you are askingis it something to do with NTP?
PaulH
Daniel - Asterisk wrote:
> I guess it was a problem with my connection, here the complete question..
>
> Dear all,
>
> I wanna know what can I do to get the PBX's clock from an external AMI
> server, especially wi
's been solid for me (just a customer, no connection to
the company).
-- Paul
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Is a drive image out of the question?
PaulH
David Klaverstyn wrote:
>
> Hi All,
>
> I’m in the process of writing an install script and I would like to
> change some settings for the install process but I don’t want the user
> to go into menuselect and make the changes manually.
>
> Is there a
I would upgrade to the latest 1.4, if stable is what is needed.
PaulH
Gabriel - IP Guys wrote:
>
> Dear All,
>
> I have a asterisk setup that is currently running on version 1.4.15 –
> I wish to upgrade or migrate this instance to the current asterisk
> stable, 1.6.0.6. It is my intention to bu
If there is an asterisk users group in your area, visit and ask lots of
questions.
PaulH
Roland Roland wrote:
> Hi all,
> a few month ago I got the task of setting up asterisk for my company.
> I had 94 employee to set this up for ...
> I never heard of asterisk before to b honest, so after res
Karl Fife wrote:
> On the landing page of the Polycom web site there's a "We're
> listening" nanosurvey, asking what is the one thing Polycom can do to
> improve their products. The link points here:
> http://polycom.zuberance.com/survey.htm
>
> I wrote a sentence about tweaking the user interfa
products_IP02.htm
None have USB or wifi, as far as I know. If you can elaborate on what
the USB is needed for, maybe someone can suggest other ways to achieve
the same end.
-- Paul
Robin Rodriguez wrote:
> what about http://www.rowetel.com/ucasterisk/ip04.html seems like what
> you might
Are both of your agents logged in?
What does the CLI show?
PaulH
edw...@web.de wrote:
> Hi,
>
>
> I have a problem with queue strategy. But only 1 of my agents ring,
> when someone call.
>
>
> my queue.conf:
> [MyQueue]
> strategy=ringall
> member => Agent/201
> member => Agent/202
> announce-
The Asterisk console is pretty goodbut there was a text version of
one of the softphones once (sjphone, if I remember correctly)
PaulH
Hans Witvliet wrote:
> While reading the thread about recommending usb-phones...
>
> Once in a while, i'm in a data-centre, no normal phones, and too much
>
http://www.geonames.org/ is a great free resource, though not sure if
it's what you're looking for. The maintainer is pretty approachable, try
the forums if you don't see what you need.
-- Paul
Dean Collins wrote:
>
> I need a country, state, city database
Joseph wrote:
> I'm using Asterisk 1.4.22.1
> When I'm on active call it happens many times the call gets interrupted by
> music-on-hold without my pressing any button.
> MOH just kicks in and int erupt the call and I have no way of getting the
> call back.
>
> Did anybody experienced anything l
Any idea what this means? And why they are different?
->
Extension Changed 22142[default] new state Idle for Notify User 31001
(queued)
Extension Changed 22142[default] new state Idle for Notify User 30060
->
I have googled and searched, and can't find anything on this subject.
Does anyone
SIP wrote:
> I believe SNOM 300s do PoE (might have to check that, though) and are
> around $100. We've little experience with them, but we use an office
> full of Snom 320s, and we're nothing but pleased with them. Good
> speaker, good handset, lots of excellent options. And reasonably priced.
>
> I am probably missing something, being a newbie. I have a 4 port
> fxs/fxo (2/2) card.
>
> My land line is going to one of the FXO port and my home phone is connected
> to one of the FXS port.
>
> I want to be able to call my phone number from external phone (cell phone)
> and have my home pho
> I can't force them to use star codes to set DND in astdb).
>
>
Once again, someone who underestimates the power of physical violence.
PaulH
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I wish it was available too - I have just had to back dahdi out of a
system and revert to misdn after a whole day of testing.
PaulH
Andrew Thomas wrote:
> I have LibPri installed and working (.../wPRI).
>
> So, if I understand Tzafrir correctly - DAHDI support for the B410P isn't
> available i
Noojeeclick?
http://www.noojee.com.au/Page/NoojeeClick
ADM? (asterisk desktop manager?)
PaulH
Alan Lord (News) wrote:
> Dean Collins wrote:
>
>> ADA Forums: http://forums.digium.com/index.php?c=8
>> ADA Download: http://dl1.digium.com/ADA/ADAInstall.exe
>> ADA Administrators Guide: http:/
pproximate physical location of an employee in
the building(s). Kinda hard to do that with a mini-browser :)
Mini-browsers certainly have their uses, don't get me wrong, but not
everything can/should be done 'at arm's length'. A bit of local
intelligence can go a long way, ev
Outcall moved to http://code.google.com/p/outcall/
There's also Camrivox's Flexor (Snom and Asterisk versions).
Googling 'outlook click-to-call' will also show a bunch of related info
and tools.
Paul
Godson Gera wrote:
> http://outcall.sourceforge.net/
>
>
Can I assume that you want this only for blind transfers?
I have done this previously, but I lost my copy of the work (and it was
a proof of concept only)
It involved the ${BLINDTRANSFER} variable, which catches the number that
made the blind transfer and making macro-stdexten (or your equivalen
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