.
Thanks!
-erik
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of them, audio quality-wise or stabilty-wise.
-Erik
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On 9/30/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
I don't know about you, but I've had nothing but very good results with
VOIPSupply. I didnt do huge business with them, but I have purchased new and
refurb polycoms from them without so much as an ounce of pain.
Ditto - I've never had a
the subdirs of /var/spool/asterisk that
apply to your install as well.
-Erik
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://www.digium.com/en/products/hardware/digitalcards.php
You need to choose how many T1 spans you need and whether you want a
hardware EC chip on the card. I'm not sure if Digium sells a PCIe
version of their single-port card.
HTH-
Erik
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On 9/27/07, Doug [EMAIL PROTECTED] wrote:
http://www.atacomm.com/
Heh - yah I pulled up their website earlier today with the hopes of
purchasing a Polycom SIP conference phone. Oh well...
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idea?
Kind Regards,
Erik
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Hi,
Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a
business graded installation (with really traffic on not 3 calls a
day ;-) )
Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall)
Thanks!
Kind Regards,
Erik
On 9/24/07, Steve Davies [EMAIL PROTECTED] wrote:
The phones can send a parameter to the provisioning server to indicate
that they want an Update if they do this, and you send no network or
other major config parameters, the phone does not reboot.
Look at the Linksys provisioning PDF for
/reservation-b5c322c8 is busy
so the phone seems to stuck in calls and is not reachable anymore
Any idea?
Kind Regards,
Erik
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need to go to a
lower-level tool than asterisk. For sangoma cards, you can use the
`wanpipemon` command to do a packet dump. I'm not sure what the
equivalent for Digium cards is, but I'm sure it's possible.
-erik
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(one of the
above options or otherwise) is best to keep your sip.conf sane?
Thanks!
-Erik
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On 9/18/07, C F [EMAIL PROTECTED] wrote:
Use the extension, and use grep to determine which account uses which
phone. For example I provision my spa9xx phones from a subdirectory on
apache called spa which on slackware is at: /var/www/htdocs/spa/
doing:
grep 123 /var/www/htdocs/spa/* will
On 9/7/07, Mike Hammett [EMAIL PROTECTED] wrote:
If it has nothing to do with Asterisk, then why does every other device work
as its supposed to?
You never answered as to whether or not you're able to get out past
your gateway with any other network applications on your asterisk
server. Fire
On 9/6/07, Mike Hammett [EMAIL PROTECTED] wrote:
I have multiple upstreams in my office. The primary upstream is having some
issues with latency\jitter. I want to move the VoIP traffic to another
interface.
I have the router set to send all traffic destined for local networks out
the
.
Is there currently any script out there that would facilitate this
sort of testing?
Here's my current config:
linux-2.6.21
asterisk-1.4.10
zaptel-1.4.4
wanpipe-3.1.3
libpri-1.4.1
Thanks!
-Erik
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cables are connected between ports 1-2 and
ports 3-4. I'd like to generate a bunch of calls over those spans.
-erik
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On 8/28/07, Steve Totaro [EMAIL PROTECTED] wrote:
Another more creative tool would be to place an ad in the Penny Saver or
whatever your local equivalent is for a free 42 inch LCD TV, you haul
and list your number. I bet that would generate alot of calls. You could
put them through and IVR,
On 8/28/07, Steve Totaro [EMAIL PROTECTED] wrote:
Also, you seemed to miss Brian's main point, keeping calls up is not
going to tax your box or prove anything really, you want to create as
many short calls as possible. Run BOINC in the background for a CPU
burn-in test.
Well - with this
Off the cuff, I can't recall if asterisk can listen for (in this case
I assume) SIP on multiple ports. It would be quite easy to do this
redirection with iptables, though.
On 8/15/07, Walter Willis [EMAIL PROTECTED] wrote:
hot to asterisk multiport...???
example 5060, 5061, 5080
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Erik
-TRUNK=Zap/g0
[outbound]
exten = _9NXXNXX,1,Dial(${OUTBOUND-TRUNK}/${EXTEN:1})
-Erik
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On 8/6/07, Erik Anderson [EMAIL PROTECTED] wrote:
I've been going back and forth with my telco for several days, trying
different configurations to get a new PRI to come up. The bchannels
are all up and the T1 is not in alarm status. The dchannel refuses to
come up however. We've tried ni2
, put it into production and be done
with it.
-erik
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On 8/9/07, MOSBAH ABDELKADER [EMAIL PROTECTED] wrote:
Hello,
Have i to install OpenVPN in each Asterisk server or it is enough to install
it in one side only?.
Both.
You best take any further questions to the OpenVPN mailing lists.
You'll get much better information and help there.
On 8/7/07, Andrew Joakimsen [EMAIL PROTECTED] wrote:
In your wanpipe1.conf see if you have
TDMV_DCHAN = 0
Nope. I have it set to 24.
-erik
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On 8/7/07, Jeremy Mann [EMAIL PROTECTED] wrote:
In Zapata.conf, if my PRI is NI-2 configured, do I still use
switchtype=national ?
Yup:
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf#ISDNPRISwitchConfiguration
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On 8/7/07, fateme fatah [EMAIL PROTECTED] wrote:
Hi:
I want to have conference call(meetme) service with asterisk and 30
users.Now do I use 1E1 or 30 analog lines with due attention to high price
of E1 line?And which interface card do I use?
I'm not sure what analog prices are in your area,
firmware
and I'm running wanpipe-2.3.4-12 for the sangoma drivers.
Any ideas?
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out of the office today or at
least he forgot to start up MSN this morning, as he's showing offline.
Hopefully he's not the only tech support guy there.
-erik
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A102 port 1 [slot:10 bus:2 span: 1]
switchtype=dms100
context=from-pstn
group=1
signalling=pri_net
channel = 1-8
There you go.
As an aside, turns out that it's a national holiday in CA, so the
Sangoma support guys are on vacation for the day.
-erik
and get Sangoma and
the telco tech on the horn at the same time tomorrow.
-Erik
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On 8/6/07, Anthony Francis [EMAIL PROTECTED] wrote:
also in asterisk do:
pri intense debug span 1
Then you should see UA's and SABME's, If you don't, your not talking to
them.
I see plenty of SABMEs, but nothing else:
[ 02 01 7f ]
Unnumbered frame:
SAPI: 00 C/R: 1 EA: 0
TEI: 000
On 8/6/07, Anthony Francis [EMAIL PROTECTED] wrote:
You should never be the signaling source, you are always a slave to the
provider, go with pri_cpe and see if things go better.
That's what I've experienced in the past, but they were adamant about
me being the network end. I tried switching
tomorrow when I can get
both Sangoma and the telco on the phone.
-erik
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On 8/2/07, Don Kelly [EMAIL PROTECTED] wrote:
Hi, Erik,
Never heard of call-by-call trunking.
Are you in Minnesota? What carrier are you using?
Yes I am...this is for one of our branch offices, though, outside of Boston, MA.
-Erik
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anything.
Thoughts?
This is a Sangoma A102 card, by the way. In this case, though, I
don't think that's of any relevance.
-Erik
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On 8/1/07, C F [EMAIL PROTECTED] wrote:
what channel are they putting the Dchannel on?
Post your zapata.conf and zaptel.conf
The D channel is on 24.
zaptel.conf:
loadzone=us
defaultzone=us
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
zapata.conf
lpdlnx04 asterisk # cat zapata.conf
;autogenerated
.
That's not the case in my situation, as I need to be syncing with the
telco's clock.
That said, in the interest of troubleshooting, I did try setting it to
zero - this didn't fix the problem.
-erik
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comment on the quality (or lack
therof) of any of the offerrings. I believe FreePBX is the most
popular by far, though, so you may want to check that out first.
-erik
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- we've all been wasting our
precious money on CSUs this whole time. We're all idiots!
/sarcasm
Seriously - if you're so sure about your card not having a CSU, what
is the make/model? Pony up, man.
-Erik
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On 6/12/07, Olivier [EMAIL PROTECTED] wrote:
Hello,
Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone.
Did I miss something ?
I don't know of any other GE phones.
However...
Why in the world would you ever need GigE sip phones?
-Erik
) is causing some delay.
-erik
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Hi,
Im sure somebody out there had the same problem before.
IF a call comes in with suppressed caller id (Call Centers, etc.) 'asterisk'
is shown as CallerID. Can I change somewhere this behaviour to display like '
Unknown' ?
Thanks!
Kind Regards,
Erik
of ports and blades. Anyone know if such a device exists?
Have you seen these?
http://xorcom.com/products/astribank_32
I haven't used them personally, but have seen several good reviews.
I'll most likely be picking up one of their smaller models to evaluate
in the next month or so.
-Erik
). That was working pretty fine before
I switched of RTP encryption on the SNOM 360 after a hint from SNOM here on
the list, but it doesn't work.
Finaly I think such a * version switch must be really carefully done and also
expecting some troubles.
Kind Regards,
Erik
My first lines
can't find it again. Thanks much.
Use the search. The page you're looking for will return as the first
link if you search for softphones.
-erik
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On 5/9/07, Adam Moffett [EMAIL PROTECTED] wrote:
Try this:
http://puck.nether.net/npa-nxx/
This probably goes without saying, but this data is, at best,
marginally useful due to LNP.
-erik
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You need to set the following value in the sip.conf section for this phone:
mailbox = vm context@vm box
-Erik
On 5/7/07, Ray Chen [EMAIL PROTECTED] wrote:
Hi, I am configuring an asterisk PBX system. I had everything working but
MWI. Can someone provides me hints on how to set up MWI? I have
On 5/3/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Ugh. This is a Win32 app, isn't it?
Yup.
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dialplan(s)
and call routing manually, or would it be worth it to set up dundi for
extension discovery?
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like it would be at least worth
giving Dundi a try. I've never touched it before now, but I can't
imagine a configuration like mine would be too complex.
We'll see how this goes...
-erik
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.
With 1.2 it was working well.
Any suggestions?
Thanks
Erik
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a call outside (and
powered of course over the POTS net), in normal operation it should be
connected to Asterisk and the Server. Some suggestions or running
environments?
Thanks
Kind Regards
Erik
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Is there a reboot for the phone neccessary? If no then it didn`t work. I
tested to call a Linksys phone with deactivated RTP encryption, no audio
transmission.
Erik
Am Donnerstag, 26. April 2007 10:28 schrieb Tim Koehler:
Hi Erik,
have tried to switch of RTP Encryption on the snom
Thanks that's the device i was looking for. I like the Linksys PAP2T as well
so Im happy thats a device from a well known vendor.
Erik
Am Donnerstag, 26. April 2007 11:37 schrieb Cosmin Prund:
How about connecting the analog phone in the elevator to an ATA gateway
that provides PSTN FallThrow
,
Erik
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used a headset that came with
one of my cell phones, and it worked great w/ my SPA-941.
-erik
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for commercial purposes. This hasn't been a good
day for you, has it? I'm guessing I'm not the only one on the list
that has added astawerks to my banned sellers list.
-erik
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of the ports, and will be providing VoIP service to a
bunch of SIP deskphones.
So - with that usage scenario in mind, is it worth the extra $800 for
the echo cancellation?
As a sidenote, I assume these cards are PRI-compatible, correct?
Thanks!
-erik
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http://andersonfam.org
, and will only be handling ~20 calls at a time or so, so
I don't envision CPU being a problem, but offloading the EC never hurt
anyone.
-Erik
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-info.org/wiki/index.php?page=Asterisk
There is also quite a bit of good documenation available here:
http://asteriskguru.com/
Forgive me for pulling an RTFM on you, but that's all I can offer
right now with the details you've provided.
-Erik
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for the advice everyone.
-erik
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On 4/13/07, Forum [EMAIL PROTECTED] wrote:
Can anyone tell me what the capacity is of 2 E1's in minutes. Ie how many
minutes can 2 E1's take.
42.
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(using Linksys SPA-941s), all I had to do was to add
the following to each SIP account definition:
mailbox=mailbox_number@mailbox_context
-erik
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for orginization
and future flexibilty, as well as perhaps needing some code to set CID
information correctly (I'm not sure if that's passed through
automatically).
Am I heading down the right path?
Thanks!
-erik
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http://andersonfam.org
services from many people, once the
site is back online, there will be a number of read-only mirrors of
the site available as alternate access.
Thanks for using voip-info.org!
[EMAIL PROTECTED]
-Erik
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reducing some
of the load on the main server.
For some reason, I'm doubtful this will ever happen, but hey, we can
always hope, right?
Here's to v-i.org coming back soon.
-Erik Anderson
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try to make a call out, I get this error:
Jan 16 13:17:28] WARNING[18084]: app_dial.c:1081 dial_exec_full:
Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel
congestion)
Also got the same SABME errors as you do.
Best regards,
Erik Haider Forsén
/linux/
wanpipe_includes.h file wanpipe is actually looking for. Is there a
patch or a newer version of wanpipe that has this issue solved?
Erik
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file completes, I get this:
-- Playing 'logic-main' (language 'en')
== Auto fallthrough, channel 'SIP/445-0815e1d0' status is 'UNKNOWN'
Any ideas? This seemed like it should be simple, but it's getting the
best of me.
Thanks-
Erik
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On 1/9/07, Doug Crompton [EMAIL PROTECTED] wrote:
You need a 'waitexten()' after the background command.
Gah! That worked perfectly. Thanks Doug.
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to G729 transcoding calls
this setup will be able to handle?
* Which of these two would be the better?
* Would this configuration be enough to serve a TE412P to 120 channels G729 ?
Kind regards,
Erik
Erik wrote:
Hello list,
I'm currently looking into building a new Asterisk server, due to some
or dual Opteron configurations, which of these
platforms would perform better?
And how much power would be needed to transcode 120 Channels PRI to G729 (for
example Digium TE412P)?
Erik
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a 8 kbit/s stream :)
So in order to use 5 simultaneous calls you would need a 1:1 DSL line of at
least 5*42400=212000 bps so a 256/256 DSL would do, however if
you need 20 calls that would be 20*42400=848000 bps so that would be a 1M/1M
line (and some bandwidth to spare)
Erik Versaevel
you able to revive them?
Thanks!
-Erik
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Erik Versaevel
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-default-server
user myusername
works for me (note that this is a modified Patton setup, so you might have to
tweak the language a bit.)
rgds,
Erik
C F wrote:
Erik, I have tried it and it did NOT work, can you tell me where to
enter that info? Have done it and it worked?
On 9/19/06, Erik
QuadBRI PCI
Regards,
Erik
PS.:
My incoming context:
[incoming]
;
; Startup settings.
;
exten = s,1,Answer ; Answer the line
exten = s,2,Wait,1 ; Wait a second, just
fo
r fun
exten = s,3
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as well.
Thanks!
-Erik
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On 9/19/06, Erik Anderson [EMAIL PROTECTED] wrote:
Hello - I'll be heading out to the Boston area next week to start up a
branch office for my company. I'll be implementing an Asterisk box as
part of their network infrastructure...so...does anyone have any
recommendations on a good reliable SIP
packet queueing algorithms as it is easier
to shape a constant stream of packets.
Erik Versaevel
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uses about 108 kbit on
DSL
Erik
Avi Miller wrote:
Hey guys,
I need some assistance in tracking down the cause of audio problems that
are occurring at two of my sites:
Both sites run Asterisk 1.2.10 and use Sangoma A101u PRI cards. Both
sites are reporting that audio in calls is dropping
the
overhead in half :)
Erik Versaevel
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Chris,
Take in account that 64k on the RTP layer is about 108 kbit on DSL when sending
50 packets/s (20 ms samples)
ERik
Chris Blunt wrote:
Hi List,
I need to put an Asterisk server in a remote office where only ADSL is
available. Maximum of 8meg downstream 646k upstream.
I need
, sera.
Sorry that my first post is a huge plop, but it's an interesting situation
that I've been going back and forth about for a while.
Plus, Asterisk sure beats a $20k Altigen setup.
Erik Jacobs
Project Engineer
[EMAIL PROTECTED]
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UA there are 2 different call legs, so 2 different SIP-CallerID's
Erik
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if there's only one participant in the
conference.
All phones are Supura/Linksys SPA-941 phones. Everything is working
fine (users can talk to each other, voicemail is working, etc), exept
for meetme.
In meetme.conf I've got audiobuffers=32, which doesn't help.
Any clue?
--
Erik Hensema
On Thursday 20 April 2006 19:07, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Erik Hensema [EMAIL PROTECTED] wrote:
One of the servers is hosting MeetMe. It's working fine as long
as only SIP phones connected to the meetme server participate in
the conference. As soon
Kevin P. Fleming wrote:
Erik wrote:
IMHO Asterisk B should change it SSRC to tell Asterisk A the RTP source has
changed (and fix this timestamp gap)?
That's an interesting question; since Asterisk is not actually a proxy,
in point of fact the SSRC has _not_ changed, since Asterisk B
sequence 502 timestamp 500060 SSRC 5678 from ip 1.2.3.4 sequence 6
timestamp 500060 SSRC 1234 from ip Asterisk B
IMHO Asterisk B should change it SSRC to tell Asterisk A the RTP source has
changed (and fix this timestamp gap)?
Erik
to littleabout digital telephony and
such.
erik
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"or you can use hardware devices to connect to
traditional phone lines"
That can be a Digium card, Sangoma card, Linsys
SPA3000, Mediatrix 1204, and several other
devices.
Ok, thanks for the
hints...
erik
___
-
On 3/22/06, Andrew D Kirch [EMAIL PROTECTED] wrote:
Andrew D Kirch
Indianapolis, United States
snip
Well if that isn't one of the most bizarre emails I've seen come
across this list.
--
Erik Anderson
http://andersonfam.org
___
--Bandwidth
On 3/22/06, Alexander Burke [EMAIL PROTECTED] wrote:
It's a spoof of a typical Nigerian 419 scam email. Rather well done, too. :)
Thanks for the laugh, Andrew!
I should have figured as such :-) I wasn't quite sure if I should
take it seriously, though, first time 'round.
Nice work!
interface
[logic]
exten = 270,1,Dial(IAX2/erika_idefisk)
exten = 271,1,Dial(IAX2/erika_idefisk)
exten = _ZZZ,1,Dial(${TRUNK}/${EXTEN}})
iax.conf:
[erika_idefisk]
type=friend
username=erika
secret=foobar
host=dynamic
context=logic
Any ideas? Thanks!
-Erik
--
Erik Anderson
http://andersonfam.org
this all have to be so difficult??? :-)
Any sage bits of advice would be greatly appreciated. Thanks!
-Erik Anderson
--
Erik Anderson
http://andersonfam.org
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that at this point, it's a configuration
issue on the intertel PBX, and that this list is probably not the
place for support :-(
Oh well - I'll get it eventually!
thanks-
erik
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is unavailable and
agent/1 should only get calls from queue_2 when all other agents of
queue_2 are unavailable
Erik Versaevel
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