IRC log for #asterisk on 20150311

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05:29.57jaxxanhi
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05:48.12clebHello all, I am using Asterisk with a DCS SIP Trunk, and have 3 DID's. I would like the Call ID on outgoing calls to be set as a certain DID for certain extensions. Some are local extensions on this PBX, and some are coming over an IAX trunk from another Asterisk box.  I've done a bit of research and haven't seen a clean and simple way to implement this.
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05:54.58[TK]D-Fendercleb, have the dialplan they use set it.  Either hard-coded from their specific dialplan, or based on looking at who the caller is before dialing out
05:55.13[TK]D-FenderOr based on a variable set if the caller is a sip peer
05:56.03jaxxanexten => _9NXXNXXXXXX,1,Set(CALLERID(all)=XXXXXXXXXX)
05:56.30cleb[TK]D-Fender: OK, thats what I'm doing in the outbound context for the first/default DID, how do I scale that to multiple DID's?
05:56.52[TK]D-FenderUp to you
05:57.07[TK]D-FenderDepends how much you need to scale, how you intend to update them,etc
05:57.18[TK]D-FenderYOU have look at the options and judge your own scale
05:58.03jaxxanSpan 1: Channel 1/1 got hangup, cause 100
05:58.11jaxxandahdi
05:58.21jaxxani get that for inbound and outbound calls
05:58.43jaxxani see the calls coming and leaving asterisk over a PRI
05:58.57jaxxanbut they just immediately hangup
05:59.50cleb[TK]D-Fender: OK. And for my other IAX trunks with more extensions hanging off of them, for the outbound do I set the call ID on the seconday PBX, or the primary with the DCS trunk
06:00.35[TK]D-Fenderyou should be setting it on the system actually dialing out
06:03.06clebOK.
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06:25.55[TK]D-Fenderheads to bed
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07:10.46[Consultant]C-Asdoes asterisk 13 still use res_config_mysql.conf  for realtime
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09:28.30yun1989hello all
09:28.40yun1989good morning :)
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09:29.25yun1989I receive this packect in asterisk but i don't understand your format
09:29.25yun1989http://pastebin.com/eHDZTQzg
09:30.34yun1989because i receive this "anonymous@anonymous.invalid" ?
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10:05.38roxI have a set of PRI lines connected to Asterisk through ZAP, and for inbound calls those lines are not sending out the ALERTING signal, so callers don't get to hear the ringing tone
10:05.52roxdoes anybody have an idea, what configuration I am missing?
10:07.13roxCalls go (in PRI signaling):
10:07.14rox< SETUP
10:07.14rox> CALL PROCEEDING
10:07.14rox> CONNECT
10:07.14rox< CONNECT ACKNOWLEDGE
10:07.14rox... DISCONNECT
10:07.47roxthere should be an ALERTING signal between CALL PROCEEDING and CONNECT
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10:55.32WIMPyrox: Do you really still use zaptel? If so a simple upgrade should help.
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11:11.28casdudehi all, I have written a macro that is activated when caller hits *4, the macro playsback a sound file. This works well however it only playsback on one side of the call. Is it possible to playback on both sides of the call at the same time?
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11:36.05fireglowHello. Asterisk 11.16.0. I use DAHDI to receive calls. When the caller hangs up, my phones continue to ring for about 5 seconds afterwards. How can I shorten this time, and improve detection of the hangup event?
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12:13.59xphereshi
12:14.23xphereshi guys
12:14.24xpherescould someone explain me the deny and permit mask?
12:14.24xpheres<PROTECTED>
12:14.24xpheresmy extension is in the local address 192.168.178.23
12:14.24xpheresI wrote permit 192.168.178.0/255.255.255.0 and deny 0.0.0.0/0.0.0.0
12:14.24xpheresis that correct?
12:16.01Demon_VoIPacl.conf?
12:17.22xphereswhat?
12:17.49xpheresI set up everything with freepbx gui
12:18.04xpheresbut I can also modify files if needed
12:18.52Demon_VoIPxpheres, your mask is correct. But first will be better "deny", second is "permit"
12:19.22xpheresyes
12:19.24xpheresdeny is first
12:19.42xpheresI just reinstalled asterisk, I'm going to test if it works now
12:20.18xpheresbut it seems that my local phone tried to connect from outside, the rejected acl is my wan
12:22.12Demon_VoIPxpheres, delete acl, when phone connected type in console "sip show peer ???" and look IP address..
12:22.26xpheresok
12:22.58xpheresI forgot the command to connect to asterisk sorry
12:23.07xpheresI dont use much the console
12:24.05Demon_VoIPxpheres, try "asterisk -r"
12:24.45xpheresok
12:24.56xpheresit keeps trying to register but it always reject it
12:25.37Demon_VoIPacl exists? What the reason of rejection?
12:25.44xphereshow can I stop the show peer ?
12:25.48xpheresI can not copy
12:25.58xpheresbecause it moves and it does not allow me to copy
12:26.07Demon_VoIPexit :)
12:26.12xpheres[2015-03-11 12:26:05] NOTICE[1425]: acl.c:748 ast_apply_acl: SIP Peer ACL: Rejecting '88.73.52.99' due to a failure to pass ACL '(BASELINE)'
12:26.12xpheres[2015-03-11 12:26:05] NOTICE[1425]: chan_sip.c:28104 handle_request_register: Registration from '<sip:100@xpheresserver.hopto.org>' failed for '88.73.52.99:53008' - Device does not match ACL
12:26.56Demon_VoIPok. IP Address is: 88.73.52.99. External :)
12:27.13yun1989hello
12:27.32yun1989it's possible when one user leave the conference close the conference ?
12:27.36yun1989in confbridge ?
12:27.38xpheresyes demon
12:27.42xphereswhy is this happening?
12:27.56xphereswhy a local cisco phone with extension 100 does not connect from inside the network?
12:28.10xpheresit's my phone and is here
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12:29.01Demon_VoIPyun1989, you can look toward "marked"
12:30.06Demon_VoIPxpheres, what address of sip server in configuration of phone? Local? WAN?
12:30.47xpheresah maybe that's the problem
12:30.49xpheresthe wan
12:31.32xpheresyes you are right
12:31.34xpheresthat's it
12:32.16yun1989@Demon_VoIP but if you have two users with "marked = yes"
12:32.25yun1989it not works correctly
12:32.34xpheresI changed this to the local address: <natAddress>192.168.178.25</natAddress>
12:32.39xpheresin my cisco
12:32.48xphereslet's see if it works
12:33.33Demon_VoIPyun1989, yes. I don't know any other way. Except of intercept AMI events of ConfBridge and...
12:34.55yun1989@Demon_VoIP how intercept AMI events in confbridge ?
12:34.57yun1989?
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12:38.29nokowhat the variable have user sip number? I wanna use GotoIF(user from 11X to 12X...)
12:38.57nokoto match users from 110, 111, 112..
12:39.06nokoi just need variable name
12:39.12[TK]D-Fendernoko: WHAT "SIP NUMBER"?
12:39.51noko[TK]D-Fender, local pbx users numbers that stored in sip.conf
12:40.51WIMPyUsers shouldn't be numbers. that's impolite.
12:40.53WIMPyAnd insecure.
12:43.13[TK]D-Fendernoko: "core show application SIPPEER"
12:45.19casdudehey
12:45.38casdudedoes anyone know if its possible to playback to callee rather than a caller?
12:46.19WIMPyWhen?
12:46.56[TK]D-FenderSeveral ways.
12:47.05wjeanneau_Hi, anyone has experience with ARI ? I mean I use ARI in order to make an outgoing call and start a specific AGI (like call file). I'm looking a way to pass external parameter into my AGI but I didn't find any option in the ARI doc except the option related to spasis app but I don't use it. I use asterisk 13.
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12:49.06casdudeI am running a macro to playback a sound based from received DTMF
12:49.37[TK]D-Fendercasdude: You're now bringing the term "macro" into this which also doesn't answer WIMPy 's question.  WHEN?
12:49.51[TK]D-FenderAh, DTMF
12:49.55[TK]D-Fender. features.conf <-
12:50.07casdudehowever the playback only plays to the caller rather while on inbound calls i would like it to play to the callee
12:51.04casdudehttp://pastebin.com/1i79Ba9v
12:51.12casdudeher are my settings
12:53.33casdude*here
12:54.22casdudedoes that help at all
12:54.29casdudelet me know if you need more info
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12:56.16[TK]D-Fendercasdude: Whole features.conf....
12:56.42[TK]D-Fendercasdude: And a failed call attempt
12:56.51casdudethe call does not fail
12:57.03casdudeit is the playback
12:57.11casdudeof the de-activated
12:57.45casdudeis it possible to playback to the callee rather than the caller
12:58.14[TK]D-FenderShow us the call
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13:00.49casdudehttp://pastebin.com/yhJTFJ3P
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13:02.14[TK]D-FenderI don't seee it executing at all
13:02.18[TK]D-FenderFor either channel
13:02.35[TK]D-Fender68-69.  Answer then hangup
13:03.11casdudehttp://pastebin.com/Edf1Fd1t
13:03.22casdudeok
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13:04.44casdudei'm sorry i must have gotten the wrong call
13:05.04casdudethe site is very busy so is hard for me to grab the entire call
13:06.20[TK]D-FenderAs long as we get something good to follow up on.
13:07.50casdudehttp://pastebin.com/Xg5V862g at line 84 the call is paused
13:12.35[TK]D-Fendercasdude: Yup, that looks like it's executing against the wrong channel...
13:13.04[TK]D-Fendercasdude: If you've done mods to the filee I don't thinkg a simple reload would apply them.  IIRC you have to reload the module entirely.
13:13.14[TK]D-Fenderwhich I'm not certain if it would disrupt calls
13:19.06[TK]D-Fendercasdude: Verify "features show"
13:20.01[TK]D-Fenderin case you flipped these around during testing
13:25.46casdudekk
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13:28.13casdudehttp://pastebin.com/2HAe0qbX
13:28.53casdudeummm
13:28.55casdudeok
13:29.23casdudei think the features look the same
13:32.21[TK]D-FenderYeah, it doesn't list the activateon/activateby though...
13:32.32[TK]D-FenderThink you might have done changes sice last restart?
13:32.50[TK]D-FenderBecause based on the name se see executed it should be running on the othere channel...
13:34.32casdudehttp://pastebin.com/1i79Ba9v
13:34.48casdudethis is the original settings posted, which have not been changed
13:35.11[TK]D-FenderThis LOOKS like a bug based on that....
13:35.30casdudeyeh
13:35.31[TK]D-FenderWhat versi0on are you running?
13:35.52casdude1.8.20.0
13:36.27casdudeit works its just the playback does not go to the callee
13:36.35casdudeit only plays to the caller
13:37.07[TK]D-FenderLook at the channel it's executing on directly though.. that's what seems to be the issue
13:37.20[TK]D-Fender<PROTECTED>
13:37.36[TK]D-Fender--  Feature Found: in-pauseMonitor exten: in-pauseMonitor
13:37.37casdudeumm
13:37.38[TK]D-Fender-- Executing [s@macro-pause-record:1] Playback("DAHDI/i1/01633508781-9441", "de-activated") in new stack
13:37.52casdudeyeh
13:38.10[TK]D-Fenderin-pauseMonitor is supposed to be ActivateOn = peer.
13:38.20[TK]D-Fenderbut it's runnign against the caller channel....
13:38.34casdudeyeh
13:38.41[TK]D-Fenderupgrade to the latest in your branch and retest.  If it continues, post it on the tracker
13:38.49casdudedamn it
13:38.51casdudeok
13:38.52file1.8 is no longer supported
13:38.59fileit's security fixes only
13:39.03casdudesure
13:39.06[TK]D-Fenderfile: When did that happen?
13:39.13file[TK]D-Fender, end of last year?
13:39.16[TK]D-Fenderfile: isn't that early for "LTS"?
13:39.28filenope
13:39.51filehttps://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
13:40.09casdudethanks for your help
13:40.23casdudeTK
13:42.18[TK]D-Fenderfile: ah, 4 years and just crossed it
13:42.24fileyup
13:43.02[TK]D-Fenderfile: I thought 12 was EOL'd completely already...
13:43.37[TK]D-FenderI remember 10 got shit-canned faster than usual
13:45.57malcolmd10 and 12 had the same schedule.  10 went away a little faster than 1.6.2 did, but it was the same as 1.6.1 and 1.6.0.  1.6.2 drug a little bit as we transitioned back away form the 1.6.x experiment.
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14:07.14pEYEdwhat is a decent gui to use with raw asterisk? nothing modular freepbx
14:07.56pEYEdlike freepbx   o.0
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15:01.37ThomasKellercan somebody please advise how to get rid of the following error?
15:01.38ThomasKellerERROR: config_options.c: Unable to load config file 'udptl.conf'
15:01.55ThomasKellerI am not using fax
15:02.09ThomasKellerand i am not loading any of the fax-related modules
15:02.43ThomasKellerwhy does asterisk still keep looking for the file ?
15:03.10filebecause UDPTL support is core
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15:04.11ThomasKelleris UDPTL used for anything other than fax ?
15:06.23coppicethankfully, no
15:08.03ThomasKellercan I disable UDPTL ?
15:10.51filelack of that file is not fatal, it's safe to ignore it if you aren't using UDPTL
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15:14.04ThomasKellerbut how can I get rid of the errors in my log ?
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15:43.34mjordanprovide a udptl.conf
15:47.07ThomasKellerI have created an empty file udptl.conf and the errors went away
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16:24.45rolerI am running Elastix (asterisk) on an NLX4000. We have been using this phone system for a few weeks now and everything is great, however we tried to dial out to a 3rd party conference line (hosted elsewhere) and when it says to enter the conference code, it is not registeering our key presses. Is there a setting on our end to make it send DTMF (or whatever it is) a different way?
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17:27.42jaxxanhello
17:27.47jaxxananyone awake ?
17:28.10linociscojaxxan, yes. We are GMT+
17:28.23linociscojaxxan, but still awake
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17:29.16jaxxando you have experience passing calls via PRI through a wcte43x+ Wildcard TE435/235 to an adtran 908?
17:29.32jaxxan(=
17:29.45jaxxani'm getting error 100 on the PRI side
17:29.56jaxxanwhenever i pass calls in either direction
17:30.04jaxxanit's a pretty vague telco error code
17:30.32linociscojaxxan, I am sorry . it is beyond my experience
17:30.56jaxxanno worries
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17:31.21jaxxani havent' really used PRI's in asterisk since it used to be zaptel
17:31.40jaxxani think i got the dahdi thing figured out though
17:32.05jaxxani'm just missing some pertinent configuration option somewhere
17:33.19jaxxanyou guys still use pastebin a lot in here ?
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17:35.05linociscojaxxan, for long paste, IRC users love to see pastebin or something like that
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17:49.00[Consultant]C-Ashi,  incomming calls are comming with a plus..  is there a way freepbx can change that in trunk or routes before sending to a destination app?
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18:22.14WIMPyjaxxan: What is vague about that error?
18:23.02WIMPy[Consultant]C-As: You need to ask in #freepbx
18:23.29jaxxanthe fact that i have no clue as to how to fix it. so i'm reinstalling asterisk and dahdi from scratch
18:23.29[Consultant]C-Asopps ,  thought i was in there lol
18:23.46jaxxanjust trying to ensure i didn't miss something
18:24.01jaxxanbefore i call someone for help (=
18:24.38WIMPyReinstalling is not going to help. Reconfiguring might.
18:25.21WIMPyBut sounds interesting that you get the same in both directions.
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19:01.23jaxxanin chan_dahdi.conf i dont need to worry about trunk groups or spanmaps if i'm only connecting a single PRI right? i can just leave that stuff commented out?
19:01.59WIMPyYou will need one group.
19:02.32jaxxanso just uncomment ;spanmap => 1,1,1 ?
19:02.46jaxxanand trunkgroup => 1
19:03.59WIMPycan't remember all those parameters...
19:04.49WIMPyThey can all be left commented out.
19:06.37jaxxank
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19:24.05jaxxanok
19:24.27jaxxanoutbound calls are working
19:24.32jaxxaninbound, i have no audio
19:24.38jaxxanyay. progress (=
19:29.49jaxxanhrm
19:46.08jaxxanhrm
19:46.34jaxxani'm not seeing any errors for why i'm not hearing audio from my cell phone calling into the PRI
19:47.25WIMPyWell, you can't, that kind of error doesn't exist.
19:47.38jaxxani dial in, i see the call hit asterisk over the dahdi/i1 and transfer to my x-lite extension. my x-lite rings and I can answer the call, but no audio
19:47.45WIMPyUnless the call was never answered.
19:47.50jaxxanoh
19:48.17jaxxani had it going to:
19:48.28WIMPyMissing audio is a SIP thing.
19:48.28jaxxanexten => 4848,1,Answer()
19:48.28jaxxanexten => 4848,n,Wait(1)
19:48.28jaxxanexten => 4848,n,Playback(tt-monkeys)
19:48.30jaxxanexten => 4848,n,Hangup
19:50.03jaxxanit shouldn't be a nat issue since it's coming through the PRI
19:50.21Synthase_You can use dahdi_monitor to visually confirm you have audio at the PRI leg. But yeah, missing audio is usually a RTP issue.
19:50.28WIMPyNo IP, no NAT.
19:50.50jaxxan[Mar 11 15:47:21] NOTICE[18648][C-00000006]: res_rtp_asterisk.c:4367 ast_rtp_read: Unknown RTP codec 126 received from '172.22.10.122:53566'
19:51.03jaxxanthat's probably it there then
19:51.33filethat's a keep alive from X-Lite
19:52.25WIMPySo theobvious question is: Did you enable your microphone?
19:55.35Synthase_You may need Progress() before Answer(), if I am remembering correctly, when using Playback(). Someone correct me if wrong.
19:56.06WIMPyNo need.
19:56.23WIMPyUnless you want to use Playback with the ,noanswer option.
19:56.47Synthase_That's it, dealing with early media on PRI.
20:00.57jaxxanmy x-lite client can call via sip to asterisk and out the PRI to my cell phone and I get audio in both directions. the call is clean.
20:01.21jaxxanthe other directions, from my cell phone down the PRI to asterisk to the SIP client results in no audio in either direction
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20:10.31bkruseHey guys - is there a way to check if a sound file exists, through Asterisk? Was thinking of adding this functionality in ARI, as it's important for me to know that information before I originate a call to playback a file. I could be overthinking, but I think it'd be neat
20:12.26jaxxangot it
20:12.54jaxxanit helps when you dont fat finger: siganling=pri_cpe
20:14.34mjordanbkruse: there is a sounds resource
20:14.59SamDaManjaxxan, I know how you feel I make stupid typos that cause me headaches all the time
20:15.24jaxxanif you tried to play it with dialplan and it resulted in a non-zero, then you could go to different dialplan?
20:19.12jaxxanwell i'm all excited now
20:19.33jaxxantime to prank call people @ work
20:23.05SamDaManuse pitch_shift and disguise your voice
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20:53.40[Consultant]C-Asis this the correct way AGI(a2billing.php,1,callback) then this  AGI(a2billing.php|1|callback)  ?
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21:29.36pjensen00Is there a transcript or something for all the core asterisk sounds?
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21:48.01malcolmdthere's a .txt file in the package that has a transcript
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22:04.04pjensen00ok, I'll look for that
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22:04.55pjensen00malcolmd: found it thanks
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23:28.47oli-workhey guys/gals
23:28.58oli-worki've got a pbxiaf install running in deviceanduser mode
23:29.10oli-worktrying to work out if there's a way to set the device displayname
23:29.15oli-workto be the currently logged in extension
23:29.27oli-workrather than the static device name
23:29.34oli-work(polycom 330ip handsets)
23:29.37oli-workany ideas?
23:35.20jaxxanno. i just gave them physical extensions and let users log in and out. each user had their own extension that followed them to whatever phone they happened to be logged into.
23:35.59jaxxanif it was permanent station then i put it in the phone tftp config
23:36.25jaxxani haven't used pbxiaf
23:37.00[TK]D-Fender<oli-work> rather than the static device name <- no, but you could work out a MicroBrowser script to show it on the Idle page instead.  You'd then leave the line-keys ambiguous as to "ID"
23:37.54[TK]D-Fenderoli-work, This is all on you to create of course
23:38.12jaxxanyeah, he's right. it's doable, it just depends on how much time you have to develop it yourself
23:38.27jaxxanthere is nothing in the world like custom tailored bro
23:38.52oli-work:)
23:38.53oli-workthanks
23:40.28[TK]D-Fendernot the easiest, but far from challenging
23:40.58jaxxani developed an asterisk based pbx for the business office of a telecom quite a few years ago. it was fantastic.
23:42.10jaxxanwhen i left though. no one else currently employed by them knew anything about asterisk and they couldn't maintain it. and then voip hackers became predominant and they apparently didn't know how to firewall either.
23:43.37jaxxani see voip hackers trying to pass international calls day in and day out. all dey evry dey
23:45.35jaxxantoday i find myself delivering PRI's to businesses with traditional PBX's using Adtrans. i love that i can use asterisk to test all of these service delivery technologies
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23:50.30jaxxanyou know what i dont like? dealing with those nortel admins that sit there doing nothing except ticking the clock, getting paid by the hour.
23:51.25jaxxanthe most unhelpful bunch of <explicative>
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