irclog2html for #asterisk on 20060116

00:02.06*** join/#asterisk dily (n=dily@ip-85-108.sn2.eutelia.it)
00:02.21*** join/#asterisk ZeMMaD (n=ZeMMaD@209.59.105.69)
00:03.07*** join/#asterisk GarryH (n=guangyao@S0106009027bbc526.ed.shawcable.net)
00:03.47effapeit's kinda big - AMP
00:05.12*** part/#asterisk GarryH (n=guangyao@S0106009027bbc526.ed.shawcable.net)
00:05.51*** join/#asterisk GarryH (n=guangyao@S0106009027bbc526.ed.shawcable.net)
00:06.44argentasokay, AMP is nasty, gonna make diagnosing the issue nigh on impossible..
00:07.00Ariel_argentas, it's not nasty
00:07.00justinuheh
00:07.29Ariel_he needs to get his country codes and setup for his location in the zapata.conf correct.
00:07.49argentasAriel_: it is from the point of view of trying to figure out what the hell is going on in the dialplan
00:08.04Ariel_argentas, I have no problem with the dial plans
00:09.03argentasyes, I agree, problem is *probably* the zap setup, but polarity switch is usually correct for the UK, but with that his SIP phone is not ringing
00:09.30Ariel_his sip rings just when he pickup the line hangs up
00:09.58TrazzAriel,
00:10.01Ariel_he can change the dial sec's from the default 15 to something like 30 or more
00:10.08TrazzDo you recommend any good softphones ?
00:10.18effapewell here it is http://pastebin.com/507463
00:10.24franckTrazz: xten
00:10.38franckTrazz: ekiga (aka gnomemeeting)
00:10.41Ariel_Trazz, xten makes a good one for most use. I use xlite for testing and most setups
00:10.57argentas24 ; when polarity is used the phones doesn't rign and the asterisk doesn't pick up.
00:11.18effapeisn't it polarity in the uk though?
00:12.16argentaseffape: I thought so, yes, although looking again at what you have said, it does seem you are coming closer to it working with hist
00:12.22effapetook a while for me to get to the point when asterisk didn't need to pickup the call to divert it to the sip. Now it rings on the incomming call end (and doesn't get picked up)
00:12.48effapeyeah it's just that it drops when i pick it up
00:13.02argentasis that with polarity or hist?
00:13.03effapeusehist works better than polarity but it just then get's hungup
00:13.35effapepolarity doesn't do anything usehist get's to the sip but then hangup on pickup of the sip
00:13.43justinuanalog telephony.... what a pain in the ass
00:13.49effapeyeah tell me about it.
00:13.55shmaltzanybody tried this:
00:13.57shmaltzhttp://www.voipsupply.com/product_info.php?products_id=1009
00:14.01effapeI tried to get them to use isdn at least ;)
00:14.10shmaltzits an aastra 9112
00:14.33justinui've used aastra 480i
00:14.35justinuthat's it
00:14.37*** join/#asterisk rstandy (n=rastandy@d83-176-4-141.cust.tele2.it)
00:14.58effapei'm wondering if it's something to do with caller id not coming in. I dunno if it's setup or not on the line but i don't know how to check.
00:15.07argentasone sec, i'm gonna see if i've got a backup of the config from the analogue setup i had connected to BT (I'm not using only PRI)
00:15.15franckis it possible to use asterisk to look in enum first if the phone is available and then dial via PSTN if not?
00:15.15argentass/not/now
00:15.16Ariel_shmaltz, it's an ok phone. But I would recommend the Sipura 941 better
00:15.46shmaltzAriel_, comparing those 2, Audio who wins?
00:15.46Ariel_franck, yes
00:15.56Ariel_941
00:15.58franckAriel_: pointer on config?
00:16.09effapeok cool cheers
00:16.13shmaltzARiel_, nat support who wins?
00:16.17Ariel_franck, the enum web site has samples rules
00:16.22Ariel_sipura
00:16.27Ariel_hands down
00:16.32shmaltzAriel_, ease of use, who wins?
00:16.40shmaltzAriel_, feel and look? who wins?
00:16.41franckAriel_: e168.org?
00:17.05*** join/#asterisk websae (n=websae@CPE-24-167-204-30.wi.res.rr.com)
00:17.07Ariel_sipura all around.  But the 9112 looks more like a phone. But the sipura 941 has it beat.
00:17.33websaeAriel_: you like sipura?
00:17.33Ariel_the sipura looks like a Cisco phone
00:17.36websaeI have the 841
00:17.40shmaltzAriel_, i have been having massive problems lately with sipura, so I'm not sure I want to try them again, what do you think?
00:17.47Ariel_841 is ok but it not like the 941
00:17.57websaewhat's good about the 941?
00:18.00effapehmm does this indicate i don't have it dialparties.agi: callerid = unknown /
00:18.03Ariel_lots
00:18.24*** part/#asterisk korihor (n=humberto@200.35.210.134)
00:18.40Ariel_shmaltz, what problems?  the 841 was in my view there first try and was not up to part. Better then the GS but not by much
00:18.51websaewhat's a good phone for a main call center location (secretary) with multiple lines...inexpensively speaking..?
00:19.02Ariel_the 941 is a different phone all together the speaker even works great on it.
00:19.15Ariel_Polycom 501 best all around phone for the price
00:19.23justinuyeah
00:19.25websaemulti-line?
00:19.26argentaseffape: does it work if you say usecallerid=no ?
00:19.29justinuthat's still basically true
00:19.32mog_homepolycom rocks
00:19.35Ariel_Polycom, polycom
00:19.35mog_homei love that phone
00:19.36effapelet me give it a go
00:19.45justinui love my ip601
00:19.45mog_homei am gonna get a 501 for home
00:19.50justinuget a 601
00:19.56websaei need one with 6 lines
00:20.03justinu601 = 6 lines
00:20.11Ariel_601 are great as well but for the overall price the 501 is better value
00:20.28EquinoxHow are the 301's?
00:20.56Ariel_301 and 300 from polycom are cheap and work great but are really cheap
00:20.56justinuariel: can you use 501 with CDP?
00:21.11Ariel_OK what is cdp
00:21.19mog_homewhat does 601 have that 501 doesnt
00:21.24justinucisco's verion of PoE
00:21.29Ariel_poe, web screen
00:21.40Ariel_601 can add the side cars
00:21.42mog_home501 has poe i thought
00:21.52Ariel_501 yes but only with there cable
00:21.55justinumog_home: you need some stupid cable that costs 40 bucks
00:22.02wunderkinhas anyone here worked with sphinx?
00:22.05mog_homemeh
00:22.09justinuyeah, meh
00:22.11mog_homei thought 501 could do xml
00:22.12justinuthat's why I got a 601
00:22.21Ariel_only for a directory
00:22.22justinui have a 24 port dlink PoE switch at home
00:22.29mog_homehow much is 601?
00:22.30Ariel_nice justinu
00:22.34justinu250, i think
00:22.36Ariel_less the 300
00:22.39mog_homedamn
00:22.42mog_homethats 100 more
00:22.50wunderkinwere thoses mehs to me? heh
00:23.00Ariel_yes it's more that is why I say 501 is best for the price
00:23.01justinuif you can get a 501 for 150, that's a smokin deal
00:23.08mog_homeebay
00:23.12mog_homei saw some a while ago
00:23.15mog_homeat that price
00:23.16Ariel_new for 171
00:23.27*** join/#asterisk santiago (n=santiago@208.195.215.222)
00:23.48EquinoxNew from voipsupply mine was about 200
00:24.03justinuvoipsupply is a rip
00:24.09justinutry atacomm.com
00:24.09mog_homeyeah
00:24.19mog_homei need to find a friend at polycom
00:24.23mog_homesee if one can fall of a truck
00:24.26mog_homearound my house
00:24.36justinui found the phone to be worth the money
00:24.38mswthose trucks
00:24.39Ariel_voipsupply is not a rip. give them a call
00:24.52justinuvoipsupply's prices are high......
00:24.56mswalways things falling out of them
00:24.58justinuthats fact
00:25.02Ariel_justinu, on the web give them a call
00:25.06mog_homeyup msw
00:25.14justinui'd rather not talk to anyone :P
00:25.17justinui'm in telecom
00:25.36Ariel_ok so your in telecom and don't want to talk...
00:25.52Ariel_justinu, they can set you up with var prices on the phone then you can order via the web.
00:25.57justinuok
00:26.00justinui'm just teasing
00:26.23Ariel_ask cory he is always on list
00:26.30wunderkinbut if one falls off of a truck thats another story
00:26.40mog_homewell i hate talking them
00:28.35*** join/#asterisk iq (n=iq@71-38-78-98.omah.qwest.net)
00:28.50*** join/#asterisk _ke4qqq (n=ke4qqq@srv.fgp.com)
00:29.16GarryHAny one using Snom 360? Is it a good phone for business use?
00:29.28justinuit's ok
00:29.43justinuit has it's strong points and weak points
00:30.01mog_homei loved mine
00:30.05mog_hometill the screen broke
00:30.07Ariel_the 360 is a good phone but I feel it's over priced.
00:30.09GarryHJustinu: What are the weak points?
00:30.13mog_homebut it was a "truck phone"
00:30.22mog_homebut its a good phone
00:30.34justinuweak points: sound quality, funky interface, odd design most people don't seem to like
00:30.50*** join/#asterisk effape (n=nick@81.5.150.54)
00:30.57Ariel_in my view for buz go with Polycom.... can't go wrong with them 501 and above
00:30.59effapesorry connection issues again
00:31.05effapeit still disconnects
00:31.16mog_homeyeah 501 is cool as shit
00:31.16justinui'd have to agree with Ariel_
00:31.20mog_homei liked snom 190
00:31.23mog_homeand 320
00:31.24mog_homeas well
00:31.31effapeException on 23, channel 3 / Got event Polarity Reversal(17) on channel 3 (index 0) / Hangup due to Reverse Polarity on channel 3
00:31.38Ariel_too light feel like plastic hell but they work
00:31.41mog_homei still wish someone made a phone equal in ergonamics to 7960
00:31.43effapeWhich appears to be caused by picking up the sip phone
00:31.55argentaseffape: mind if i pm with you?
00:32.12effapesure if you tell me how (new to irc)
00:32.16Ariel_mog_home, do you know any settings for tdm400 and UK???
00:32.29*** join/#asterisk Aughey (n=jha@ns1.washucsc.org)
00:32.29justinuwhat kind of trunk?
00:32.30mog_homethere is a setting for hangup detect with bt
00:32.35mog_homebut it shouldnt matter
00:32.37justinusome BT stuff is DASS2!
00:32.39mog_homeand there is some cid stuff
00:32.45GarryHJustinu: We are going to upgrade all office phones from Nortel Meridian to new SIP phones. We are considering cisco 7960, polycom601 and snom 360. How your guys think?
00:32.57justinugarryh: polycom 601, no doubt.
00:32.57mog_homeid go polycom 601
00:33.06Ariel_601 polycom, polycom
00:33.08mog_homebut 7960 has best ergonamic feel
00:33.14mog_homepolycom close second
00:33.17effapehow do i reply?
00:33.18Ariel_7960G only not plain
00:33.20mog_homeand polycom has most features
00:33.36Ariel_there are two 7960 and 7960g
00:33.49argentaseffape: don't worry, just try the suggested config..
00:33.59shmaltzGarryH, have you looked at citel.com?
00:34.01effapehehe yeah that's what i had
00:34.03justinupolycom is just gonna cost you 100 bucks a phone more
00:34.10argentasah, ok
00:34.12Ariel_G is best.. But I still like the Polycom better due to you get to upgrade the sip for free and don't have to spend extra for sip
00:34.16justinus/polycom/cisco/
00:34.27effapei'll give it a go with all the other stuff out
00:34.34mog_homehow is that true?
00:34.45shmaltzPlycom is the best LAN phone out there
00:34.52mog_homehow much is a 7960g with sip?
00:34.57justinu300
00:35.05justinuat the lowest
00:35.14mog_homeso how is that 100 bucks more for a polycom?
00:35.15shmaltzmog_home, if you have to ask you cant afford it
00:35.21mog_homelol
00:35.28justinuthen you need a 802.3af -> CDP converter
00:35.30mog_homei only want 2 phones at home
00:35.31justinuif you want poE
00:35.33mog_homei want a polycom
00:35.34justinuthose are 20 bucks each
00:35.36mog_homeand a wisip
00:35.48mog_homebut really waiting on the wiax
00:35.54mog_homebut no one is making one yet
00:35.57Ariel_you have to pay 150 for the sip lic
00:36.22GarryHMog_home: The Meridian system one good feature: Each phone has 5 line buttons bridged to all five pstn lines. Is this feature supported by *?
00:36.28Ariel_Cisco don't come with sip licence, it's extra
00:36.35mog_homenot really GarryH
00:36.40mog_homeasterisk is not a keysystem
00:36.43Ariel_gambolputty, no
00:36.47mog_homebut you can make it like one if you want
00:36.55mog_homeyou can have line apperances
00:37.05Ariel_hint
00:37.15argentaseffape: did you say it was a TDM400 you were using?
00:37.19mog_homeand there is stuff in bug tracker to do really cool stuff with hints
00:37.22effapeyeah
00:37.23mog_homethat should be in tree soon
00:37.30GarryHmog_home: How? what's line apperances?
00:37.49mog_homeits a way for you to see the status of a device
00:37.58mog_homeso if bob is on the phone
00:38.02mog_homea polycom 601 can have a light light up
00:38.06franckmog_home: look at www.planet.com.tw in their Internet Telephony products
00:38.08mog_hometo tell you
00:38.10*** join/#asterisk Lurr (n=pr0ph3t@adsl-156-168-59.mia.bellsouth.net)
00:38.15GarryHshmaltz: The Citel is too expensive. Not worth
00:38.23effapebrb
00:38.29*** part/#asterisk Lurr (n=pr0ph3t@adsl-156-168-59.mia.bellsouth.net)
00:38.37shmaltzGarryH, I'm not so sure you are right
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00:39.11shmaltzGarryH, consider this:
00:39.13shmaltz~$200 * 24 phones for a polycom 501, vs $125 * 24 for citel
00:39.25mog_homewhat you want me to see franck
00:39.53franckmog_home: 2FXS to sip adapters with PoE
00:39.57GarryHshmaltz: We only have 17 meridian phones.
00:40.17*** part/#asterisk websae (n=websae@CPE-24-167-204-30.wi.res.rr.com)
00:40.24mog_homei just want to put stuff in my house
00:40.35Ariel_it's best if you can replace all phones instead of using part from one type then from others.
00:40.36shmaltzGarryH, well one citel box = $3000, 17 * 200 = $3400
00:40.43*** join/#asterisk gaupe (i=rmo@slogen.sunnmore.net)
00:41.16Ariel_wow missed extra point.. Game is getting good...
00:41.19hugo-v6gd morning
00:41.38Ariel_morning wow, I am sure glad it's not morning yet
00:41.52GarryHmog_home: Are you talking about the hint feature in *?
00:42.10mog_homeyes hints are how you trigger a line appearence
00:42.24hugo-v6Ariel_: 1h42 am here ;)=
00:42.35Ariel_hugo-v6, nice
00:43.31*** join/#asterisk a1fa (n=a1fa@24.144.50.173)
00:43.34a1fayo yo yo
00:43.38hugo-v6Ariel_: well then u still have a few hours
00:43.44GarryHmog_home: As I know hint can only indicate the status of a channel by button led, but if I press that button, can I pick up the call?
00:43.47a1fawhat is running on port: 2727?
00:44.13a1faudp: 2727, 4520, and 4569?
00:44.21a1faper asterisk conf?
00:45.23ravenpiHow do I have an call ring multiple extensions simultaneously?
00:45.37hugo-v6q: if i call someone which is busy, i get forbidden on my sip-phone. is there a way to tell the phone that the other side is busy?
00:46.00*** join/#asterisk Soul (n=Soul@87-196-39-131.net.novis.pt)
00:46.18hugo-v6ravenpi: dial(foo|bar|bam works here)
00:46.26ravenpihugo-v6: is this a local (SIP to SIP) call, or are you going through the PSTN?
00:46.44a1fahas anybody tried voice-changer?
00:46.45hugo-v6ravenpi: through pstn
00:47.05*** join/#asterisk effape (n=nick@81.5.150.54)
00:47.58effapestill doesn't work. When i take out hanguponpolarityswitch=yes it won't disconnect on pickup (but still registers a polarity switch) however it won't disconnect at all now.
00:48.23mog_homethat part you config on phone GarryH
00:48.30effapeSo if i call and don't answer it will still run though the rules and won't hangup
00:49.47a1fai dont want to fuck with my production asterisk
00:49.54a1fahas anybody tried the voice changer yet?
00:50.59effapethere must be some way to make this work ;)
00:51.25effapeIt also takes about 5 rings before it gets to the sip
00:51.35effapewhich is kinda anoying too
00:52.16effapedon't suppose you have anymore insight?
00:53.02argentasone sec, just having another look for my old configs..
00:53.09effapeexcellent ta
00:53.11TrazzAriel_, i installed the servercd.. is there a config command to run through the servername, dns, iip adress, routing still?
00:54.28*** join/#asterisk ke4qqq (n=chatzill@srv.fgp.com)
00:55.52ke4qqqanyone have experience integrating toshiba strata systems with asterisk??
00:55.56Ariel_Trazz, ifconfig
00:56.14shmaltzke4qqq, yes I do
00:56.22Ariel_ke4qqq, take toshiba sell on ebay works for me
00:56.50justinuke4qqq: what's happening?
00:56.54shmaltzke4qqq, you take the toshiba and use it as a door stopeer until you leave the room where you install asterisk, on the way out you take the toshiba with you and dump it
00:57.03TrazzAriel_ ok i can do that.. do you have a cheat sheet you use now to get asterisk on the box and up and running now?
00:57.19ke4qqqlol.....I wish I could....however they don't want to replace all at once.....want to slowly migrate....
00:57.45ke4qqqjustinu: just having problems deciphering what little documentation there is with this system...
00:57.50shmaltzke4qqq, so you can do the simple
00:58.01shmaltzdoes it support a T1 card?
00:58.02mswke4qqq: that your callsign?
00:58.09ke4qqqyes
00:58.28shmaltzke4qqq, then you buy a dual span t1 card for your asterisk box,
00:58.35mswke4qqq: suppose you asked for it? (they didn't do that when I got my ticket)
00:58.36shmaltzke4qqq, and a channel bank
00:59.00shmaltzput asterisk like this:
00:59.02shmaltzpstn <channel bank> t1 < asterisk > t1<toshib>
00:59.12shmaltzunlesss they use a PRI in which case it would look like this:
00:59.23ke4qqqschmaltz. already have a t1 communicating across...my big problem is telling the toshiba which ports are tie lines and then that certain extension are on asterisk....
00:59.24*** join/#asterisk burtonez (i=mimx@w201.ljudmila.org)
00:59.25shmaltzPSTN/PRI <asteirsk> t1 <toshiba>
00:59.58shmaltzke4qqq, it's not a problem at all, configure the moved extensions as remote extensions and have asterisk capture it
01:00.06*** join/#asterisk Rez (i=lorez@freenode/staff/lorez)
01:00.12shmaltzremote meaning off net
01:00.22shmaltzevery pbx I have worked with supports this
01:00.23argentaseffape: don't seem to have a copy of my old configs anymore - sorry. This thread might help: http://www.voipuser.org/forum_topic_2743.html
01:00.35shmaltzso I'm sure toshiba does as well, unless it's not a bpx
01:01.12shmaltzke4qqq, you got the idea?
01:01.15ke4qqqyeah it does....at least my interpretation of their weird documentation says so....
01:02.12shmaltzgtg
01:02.14shmaltzc yas
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01:18.59effapehi ho, me again :0
01:19.12effapeI'm banging my head against the wall now :)
01:20.19*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
01:20.23effapeI just don't understand why it's getting the polarity reversal - that's what seems to be causing the problem.... Why would SIP/104-4c94 answered Zap/3-1 ... Took Zap/3-1 off hook ... Exception on 21, channel 3 ... Got event Polarity Reversal(17) on channel 3 (index 0) ... Hangup due to Reverse Polarity on channel 3
01:20.25effapehappen?
01:21.03*** part/#asterisk Ironmask (n=joe@n120s119.bbr1.shentel.net)
01:21.16effapeisn't the polarity to do with the zap channel? Why would the sip answer cause it to happen
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01:29.58argentaseffape: I can only assume that the SIP answer is causing the zap channel to be answered, and perhaps the telco are then reversing the polarity at this point (causing the channel to be disconnected)
01:30.22*** part/#asterisk rabar (n=putteepu@cpe-68-175-40-207.nyc.res.rr.com)
01:31.09argentasdid you see the URL I posted for you?
01:31.33gambolputty3anyone use the IAXPEER function?
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01:32.42harry8does callerid not work on FXO ports?
01:32.47harry8analog that is
01:33.03harry8i have a TDM400P
01:33.26brookshireit should :)
01:33.46harry8hmm
01:33.49effapehmm
01:33.58harry8http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
01:34.10harry8it says it doesn't for analog lines but that doesn't make sense
01:34.23harry8i have callerid analog phones and that receives caller id
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01:41.07bn-7bcpleace excuse all the bs i bosted her earlyer, it seem rhat asterisk only captures the i\first key in the keymap (found that out the hard way) so ehen i har * for atxfer and *2 for atomon non of them worked, is tis a known issue in v1.2.1?
01:41.20bn-7bcand is ther a warkaround
01:42.00bn-7bcthat was posted not bosted
01:43.40mog_home????
01:43.53mog_homeis your keyboard jammed?
01:45.31bn-7bcno fingers in a tangle
01:45.48bn-7bc2:45 i shold be a sleap now
01:46.53JunK-Ysleep(x);
01:47.30Ikarusbn-7bc.sleep(60*60*8);
01:47.31bn-7bcit seem that asterisk only captures the first key in the keymap (found that out the hard way) so when I hare  * for atxfer and *2 for atomon non of them worked, is this a known issue in v1.2.1?
01:47.49bn-7bcthat shold be mouch clearer
01:48.49justinule vrai moyen n'a pas besoin de sommeil
01:48.54bn-7bcIkarus: yep in a few minutes
01:50.40bn-7bcwell is it a known problem or is my setup just scrowed?
01:51.05*** join/#asterisk p0g0 (n=p0g0@madwifi/support/p0g0)
01:51.49bn-7bcok it's late I'll ask again in aprox 5hrs
01:55.46*** join/#asterisk drumkill1 (n=russell@host-12-179-65-65.nctv.com)
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01:56.46*** mode/#asterisk [+o drumkilla] by ChanServ
01:57.18dilyplease, can anyone explane me what is this warning
01:57.19dilyJan 16 02:53:17 WARNING[4013]: chan_sip.c:2523 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 8/8)
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02:00.34harry8has anybody here had problems with Callerid for TDM400P FXO analog?
02:00.48Qwellharry8: only when the line doesn't support cid
02:01.15harry8hmm
02:01.23harry8is there anything special that you have to setup?
02:01.32Qwellcallerid=asreceived
02:01.42harry8in zapata.conf?
02:02.00QwellI always get the two confused.  whichever is in /etc/asterisk/, I believe
02:02.56harry8yes
02:03.01harry8zaptel.conf is in /etc
02:03.04harry8:0
02:05.16justinuzaptel.conf is just for phsyical line stuff
02:05.28justinuzapata.conf is where you put most of the signalling related things
02:06.16justinui've taken a crash cource in PRI/CAS circuit turnup with zaptel hardware in the past week
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02:11.17GD_hello... could someone tell me what groups the asterisk user should participate in (apart from dialout), in order to run asterisk as asterisk:asterisk?
02:11.42*** join/#asterisk aditya-bdg^_^ (i=EvilInLo@ws1.bratatex.melsa.net.id)
02:12.28brookshirethat should work
02:12.54GD_there's something wrong with starting asterisk from init.d and I believe it must have sth to do with permissions and stuff...
02:12.56brookshirejust have to make sure all the right directories have the correct permissions.
02:13.18GD_well i've done a chown -R asterisk:asterisk /etc/asterisk... what else should I do?
02:13.24brookshirelike /var/log/asterisk
02:13.30GD_the log files' permissions are ok as well
02:13.41*** join/#asterisk dily (n=dily@ip-85-108.sn2.eutelia.it)
02:13.46brookshirewhat's the error? can you pastebin it?
02:13.49GD_well..
02:13.59dilycan onyone explane me this WARNING:   Jan 16 03:13:18 WARNING[4491]: chan_sip.c:2523 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 8/8)
02:14.00GD_there are no errors if i start asterisk from the command line
02:14.13GD_but if I start it from init.d
02:14.14dilyPLEASE
02:14.31GD_my sip phone won't ring on incoming calls
02:14.39GD_although it can dialout...
02:14.44brookshireeven if you run asterisk as that user?
02:14.51brookshirelike su - asterisk
02:15.00brookshireasterisk -vvvvvvvvvcg
02:15.07GD_yes.. i've got the same problems if I do asterisk -U asterisk -G asterisk -vvvvvvvvvvvvvvvvvc
02:15.15GD_only asterisk -vvvvvvvvvvvvc works fine
02:15.46ast_new-Bhi all
02:16.13GD_su-ing is the same are using the -U and -G options right?
02:16.20GD_are=as
02:16.24ast_new-Bcan anyone tellme howto setup call limit by duration talk, extention, local & long distance call in dialplan ?
02:18.16GD_local&long distance is relatively easy... use a pattern matching scheme to limit a user... i believe that's fairly well documented at www.voip-info.org
02:19.42GD_brookshire can you think of anyway to debug this? the only thing I find different as far as asterisk's startup messages is concerned is a WARNING[7111]: db.c:47 dbinit: Unable to open Asterisk database
02:19.54GD_as well as a WARNING[7111]: pbx_wilcalu.c:70 autodial: Autodial: Unable to open file message
02:20.02h3xexcept you need an area code shitlist
02:20.21GD_i googled a bit about the asterisk database thing but to no avail...
02:20.58brookshirei dunno.. i got it working once
02:21.05brookshirebut i haven't played with it since
02:21.12brookshireit must be something simple
02:21.38brookshirebut if you can't get it working.. you can always try chroot :)
02:21.38GD_thanks anyway.. i'll let u know if i find anything :-)
02:21.54ast_new-Bthanks GD_, but how can grouped extentions eg : 110,111,112 they can call to long distance and another limited ?
02:22.02ast_new-Band how about call durations ?
02:22.30GD_i can't be of much help for the call duration thing
02:23.14GD_I thought you wanted to totally ban long distance calls...
02:23.15h3xshow application dial  tells you how to limit duration
02:23.35h3xrtfclh
02:24.00Luke-Jrwtf :(
02:24.03Luke-Jrmy PAP2-NA died ?
02:24.09ast_new-Bok h3x and All thanks, i'll try it
02:27.27Luke-JrAny idea on recovering a deadish PAP2-NA? ethernet/blue & red lights are dead
02:28.57Luke-Jrargh... ditto for router now too, this is not my night -.-
02:29.08AugheyAnyone have recommendations on dealing with needing 8 FXO ports.  Options are two TDM04B cards or a 8-port FXO Analog VoIP gateway.  (or any other options)
02:29.19h3xyou shouldnt have connected that lightning rod on your roof to your ethernet network :D
02:31.30Luke-JrI didn't
02:31.38*** join/#asterisk alf (n=alfredwo@dsl-202-173-191-109.qld.westnet.com.au)
02:32.10Luke-Jrso any ideas on fixing a PAP2-NA?
02:32.34h3xheh. you are screwed.
02:33.11Luke-Jr... -.-
02:33.15Luke-Jrwrong answer
02:33.27h3xno blinkey lights, no workie
02:33.38GD_brookshire: I followed instructions on http://www.voip-info.org/wiki-Asterisk+non-root and it works like a charm now! it was easy indeed but I didn't know there were so many directories asterisk needs write access to...
02:33.42alfanyone knows a way to make the snom phone leds to show you how many busy lines you have at any moment? thanks!
02:35.00brookshiregd: awesome :)
02:35.13alfanyone knows a way to make the snom phone leds to show you how many busy lines you have at any moment? thanks!
02:35.24Luke-Jrh3x: as I said, two LEDs are lit
02:35.55GD_now if only I could get voipbuster to work with IAX :-P
02:36.32ast_new-Bi try to put some function like this exten => 9, GotoIf(${$(Callerid=110,111,112)100?101})
02:36.48ast_new-Bbut why it only match for ext 110
02:40.57*** join/#asterisk khemir (i=khemir@200.56.189.30)
02:41.05ravenpiAnyone know how to turn off the occasional MWI "warble" on Polycom?  (And, no, not the stutter dialtone -- I know how to nix that.)  Thanks!
02:44.43khemirhow start a message when the user pick up the phone?
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03:05.21strtokhi
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03:09.05SkramXHiya
03:09.33warthoghaving trouble with udev after yum on centos, zaptel devices not there, readded the rules and permissions but still do not have /dev/zap dir, all channels in asterisk do not work after yum update of udev.  anyone have any ideas on this?
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03:10.55Demo_GHello, every body
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03:11.31warthognot to active at the moment.
03:11.34wunderkinany sphinx gurus yet? hehe. :D
03:11.40SkramXoh god
03:15.18warthogif I follow the instruction in README.udev for zaptel on a 2.6 kernel distro like centos 4.2, will the /dev/zap directory exist on reboot?
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03:18.15zavalagreeings all...
03:18.46zavalaI've come for a bit of wisdom from those who know all things SCCP... anyone?
03:19.32warthogis that skinny?
03:19.39zavalayeah...
03:20.02zavalaI've got the damnded problem... and it's getting the best of me..
03:20.04warthogI heard that cisco was now supporting SIP with more vigour
03:20.29zavalatrue, and I can make Asterisk sing with SIP... the problem is that I need to use the 7914 sidecar
03:20.46zavalaand that can only be used right now with Skinny (SCCP)
03:21.20zavalathat's not the issue though... it's the blasted phone... I've got a 7960 phone with the SCCP image on it and it registers with * just fine
03:21.34warthogsorry, I know nothing about skinny, but noone is talking right now, so I though I would....
03:21.49warthogyou know much about udev issues?
03:22.04zavalawhat issue are you having?
03:23.42warthogafter yum update in centos 4.2, none of my channels work in head.   looks like udev, I did the README.udev in /usr/src/zaptel but I still get an error about /dev/zap during startup and all sterisk channels do not work still.
03:24.24zavalahave you done a modprobe zaptel to see if it's there?
03:24.42warthogyeah, driver and 24xxx card load no problem
03:25.17warthogwhen I dial and extention via sip, iax or zap, I get nothing...
03:26.13zavalaby extention you mean an external interface on like the PRI?
03:26.23zavalaor do you mean even SIP -> SIP calls fail?
03:26.35warthogeven SIP -> SIP call fail.
03:27.16TrazzAriel, still there?
03:27.39zavalahmm that goes beyond the /dev/zap driver... that looks like a configuration issue..
03:28.11zavalazap is only needed if you're going to be using PRI based cards (or zaptel based card).... or for timing for things like meetme and moh
03:28.20zavalain which case you can compile ztdummy
03:28.21*** join/#asterisk welles (n=welles@222.90.141.49)
03:28.22Trazzzavala, i just installed fedora core 4 from scratch and want to install asterisk properly. which files do i need to go grab and install?
03:28.30*** join/#asterisk wellng (n=welles@222.90.141.49)
03:28.31warthogI was not sure if timer would do all this.
03:28.40*** part/#asterisk wellng (n=welles@222.90.141.49)
03:28.43zavalano, timer will only effect meetme
03:28.43argentasanyone using backports.org zaptel on Debian sarge?
03:28.49welleshi all
03:29.15warthoglooks like I will be doing a reinstall....
03:29.21zavalatrazz, you'll need the base package for sure....
03:29.37zavalahello wellls
03:29.52*** join/#asterisk goh (n=goh@60.49.6.190)
03:30.07Trazzi am going to ftp this one up then asterisk-1.2.1.tar.gz
03:30.20zavalathe zaptel package is for FXO/FXS based cards and the libpri is for... well, pri..
03:30.36Trazzok i dont have any cards in it
03:30.56zavalathen you will just need the base, the sounds, and addons if you want to play..
03:31.15Trazzok do i need to have all of those at the time i compile?
03:31.31zavalaif you plan on wanting to do conference with the 'meetme' package you'll need to download the zaptel package and compile ztdummy
03:31.43CoffeeIVI am looking for recommendations for a VoIP origination service to so that I can recieve faxes over VoIP.  Any experiences ?
03:32.05Trazzok great.. let me get started :) thanks
03:33.02zavalatrazz, you'll just need to have the base package to start withi
03:33.08warthogzavala, if I want to reinstall all asterisk packages, can I get away with just removing all asterisk stuff from /usr/src  and rm the /var/lib/asterisk and the spool/asterisk dirs? then download and compile again!
03:33.26Trazzi have the base now and just gunzip and untard
03:33.34zavalawarthog, that will work
03:33.42gohI would like to implement voip tht can bridge existing PBX in multiple branch. what hardware are recommented?
03:33.54warthogthanks, sorry I don't know about skinny to be of assistance
03:34.02zavalawarthog, np
03:34.16zavalagoh, how are the remote sites connected back today? private PRI/TI?
03:34.38zavalacoffeeIV, you want to fax over voip?
03:34.39gohbroadband
03:34.39Trazzzavala, is there a good cheat sheet to use to get this compiled, configured and up and running?
03:34.56zavalagoh, so they are VoIP pbx's already?
03:35.33gohthere are got PSTN line I just want to have a few trunk that go to IP bridge another PBX.
03:36.06CoffeeIVzavala: yes, just receiving for now
03:36.09gohso that branch call branch is free
03:36.15zavalaTrazz, if you want to install asterisk all in one directory  (other then the rood of your filesystem) then you'll need to edit Makefile and set INSTALL_PREFIX to something
03:36.25gohno they dun have IP-PBX yet.
03:36.26welleshi zavala
03:36.46Trazzyes i would prefer that..
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03:37.18gohzavala, do you think ATA device work able in this situation?
03:37.20zavalajust make a directory like /opt/voip or something like that and then set your INSTALL_PREFIX to be that directory in the Makefile..
03:37.31Trazzokay thats easy enough
03:37.37zavalagoh, ATA devices will only trunk one call at a time...
03:37.44Luke-JrAny idea on recovering a deadish PAP2-NA? ethernet/blue & red lights are dead
03:37.58zavalaif you need multiple calls trunked together you're looking at real hardware on both ends
03:38.12gohic
03:38.37zavalayou could have miltiple ATA devices, but then you reach a point where you might as well just impliment a server
03:38.42welleshi zavala ,
03:38.46zavalagoh, do your remote sites have a PBX today?
03:38.50zavalahe wells
03:39.00zavalasorry I missed your first hello
03:39.01wellesi have problems
03:39.03gohgot pbx already.
03:39.21zavalado the PBX's have T1 cards?
03:39.25gohthat why I think use back exsiting pbx.
03:39.25zavalawhat's up wells?
03:39.42gohI the HQ got T1 but not the branch
03:39.43welleszavala, it seems that there is a bug in asterisk1.2.1 's meetme
03:40.01zavalawhat are you experiencing?
03:40.05Corydon76-homeHave you reported it on the bugtracker, yet?
03:40.15welleszavala, i can not use ilbc as the codec
03:40.38*** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com)
03:40.41welleszavala, the voice became very bad
03:40.45drumkillawelles: try the 1.2 branch from svn
03:40.47gohif implement IP-PBX then they will got tow PBX in the office.
03:40.48drumkillait should be fixed there
03:41.04gohdo all branch also need IP-PBX in this situation?
03:41.53zavalawelles, some good advice from Corydon and drumkilla... look through the bug list already... and try the SVN 1.2.1 branch
03:41.58ast_new-Bcan anyone tell me what best hardware [CPU] recomendation for about 100 phone line extentions
03:42.10welleszavala, ok
03:42.17zavalagoh, it depends on what you want to do... if you're looking for toll bypass, then yes
03:42.28Corydon76-homeNot 1.2.1.... 1.2...
03:42.36zavalaahh... point taken
03:42.48Corydon76-homeIt's the tree that will become 1.2.2... and 1.2.3 after that, etc...
03:43.13zavalaast_new-B... good question... I know that some of the Dell hardware has some conflicts with chipsets... other then that, it depends on what you're going to be doing..
03:43.26gohzavala, they just want to make internal call though VOIP.
03:43.33*** part/#asterisk khemir (i=khemir@200.56.189.30)
03:44.21zavalagoh, unless each branch location has a large volume of people... this is going to be expensive all for the sake of someone being able to say "my branch calls are over VoIP"
03:44.23gohWhat the diff between IP-PBX with the VOIP gatewap?
03:45.10ast_new-Bmy scenario is replacing our PBX
03:45.25zavalaand IP based PBX generally means that the PRI gets terminated into a hard(or soft) PBX and then from the PBX to the handset is all done over TCP/IP
03:45.30ast_new-Band what;s good hardware are recomended
03:46.05zavalarock solid with room to grow, do Dell 2850 or Compaq DL380...
03:46.25a1fahas anybody tried the voice changer yet?
03:46.30a1fagrr
03:46.37a1factrl+v
03:46.38*** join/#asterisk jaike (n=a@203.131.137.76)
03:46.40CoffeeIVanyone have any experience using asterisk with voipxpress.com, particularly with receiving faxes ?
03:46.49zavalaa VoIP Gateway generally means a device that's going to take calls destined for a certain route and convert them to VoIP...
03:47.11zavalaa VoIP gateway can also be used to do backhaul voice between two sites...
03:47.12*** join/#asterisk mina (n=mina@modemcable175.94-70-69.mc.videotron.ca)
03:48.04a1faVoIP Gateway =~ ATA
03:48.18ast_new-Bzavala : how about bandwidth if it;s will connect to another asterisk server ?
03:48.46gohactually I got 37 branch tat need to call back the HQ everyday. That is costly. so how many IP-PBX need to location? do I IP PBX at HQ will work able?
03:49.02a1fa37 voips
03:49.08a1fa37 sip or iax accounts 1 pbx
03:49.12zavalakeep in mind though that routing calls over the internet is a 'best effort' approach...
03:50.07a1fagoh: pay me 3k in cash
03:50.08ast_new-Byes,  but how about bandwidth consuming for all codecs ?
03:50.11a1faand i got you hooked up :P
03:50.18zavalatheir is no quality of service for commodity internet... so if a path is long, or someone backhoes through a line... forget about phone calls
03:50.33*** join/#asterisk bmg505 (n=leon@c1-175-7.rndf.isadsl.co.za)
03:50.42Equinoxzavala - If you go with same provider at both ends you can get a QoS guarantee.
03:50.43goha1fa, why?
03:50.54zavalabandwith for codes depends on what you want and what you think people can live with others sounding like..
03:50.55EquinoxXO Communications gives me 60ms nationwide full rate node to node.
03:51.01zavaladefault is 64K per channel
03:51.12jaikeg729 works excellent for us
03:51.29jaikemake sure your internet connection is ok though
03:51.41zavalayou can get latecy garantee's... which is what XO is promissing...
03:51.58gohinternet should be ADSL
03:52.00zavalayou can't get QoS guarantees unless  you're doing MPLS or ATM
03:52.04Trazzzavala, should i use the make samples ?
03:52.20zavalayes, make samples is your sample config files to learn from
03:52.22EquinoxNever much used MPLS.
03:52.27gohit's QOS is workable if all terminal implement it?
03:52.45Equinoxgoh- Problem is you generally don't control the middle of the equation ;)
03:53.04zavalaequinox is right...
03:53.14EquinoxBut I've had good luck with most providers.
03:53.21zavalawith XO you have some favor if your XO all over as all of your traffic will be on XO's private lines
03:53.29EquinoxIt's when you hop provider to provider and end up going through a peering point 2 states away you get owned.
03:53.32zavalasame with Sprint
03:53.46EquinoxCogent is good too if it's available.
03:53.48ast_new-Bso 64K in my country costly 200$ in a month
03:54.11zavalawow... you in latin america?
03:54.30ast_new-Bso if 20 simulatanous phone will consume 20X64K
03:54.35jaikeast: thats expensive
03:54.40jaikewhere u at?
03:54.44ast_new-Bindonesia
03:54.45Math`I've a client who uses voip over VSAT, works pretty fine
03:54.58zavaladrumkilla... Corydon... you still on?
03:54.59jaikeast: im in the philippines..asterisk works great for us
03:55.08Math`there's delay, but no echo and no cutting-sound
03:55.09ast_new-Bthat;s very expensive ....
03:55.16zavalaanyone with SCCP Skinny knowledge?
03:55.45jaikewe have an E1 line...$950..can push 50-60 g729 channels through it
03:56.09zavalalike I said... 64K is the top limit... you can go down to as low as 7k
03:56.26ast_new-Bmaybe i have to use g729 with 8L
03:56.30ast_new-Bmaybe i have to use g729 with 8K BW
03:56.44jaikeast: make sure latency is 250ms and below...and number hops to a minimum
03:57.10Trazzzavala, got that all done.. where should i dive in next ?
03:57.21Trazzi didnt start any processes yet ..
03:57.24ast_new-Bjaike: how about quality if there's 20 simultan in/out phone ?
03:57.25zavalado you have phones?
03:57.35Trazzsoftphone
03:58.06ast_new-Bjaike: is there any delay or echo tail ?
03:58.10zavalathen you should start with the sip.conf and the extentions.conf
03:58.24jaikemake sure you have a good iax provider in the US....on an athlon 64 2800 with 2Gb ram, even 50 channels is no sweat
03:58.30zavalaecho isn't a result of latency on the line
03:58.41zavalayou get echo the closer you are in physical locality, etc
03:58.50jaikethe biggest problem we had was choppy lines, which was due to latency and packet losst
03:58.53jaikeloss
03:59.04jaikeonce we solved that part, everything went smooth
03:59.37zavalafor an IAX provider in the US I'd look at Jeremy's NuFone... or VoicePulse..
03:59.50jaikewe have multiple asterisk servers running
04:00.18jaikealso dont limit yourself to iax providers..asterisk does sip pretty well too..as long as youve public ips
04:00.19Math`packet loss is the big problem, latency doesnt really matter as long as there is no echo
04:00.22ast_new-Bjaike:  for how many users your asterisk serve ?
04:00.29zavalaanybody have exp. with Skinny (SCCP) and Cisco phones?
04:00.36jaikewere a call center...150 seats
04:00.38Math`zavala just ask
04:00.41Luke-JrAny idea on recovering a deadish PAP2-NA? ethernet/blue & red lights are dead
04:00.53jaikeexpanding to 500 by the end of the year...all on asterisk
04:00.54jaike:)
04:01.42Trazzzavala, i reviewed those two files and they have tons of stuff in them already enabled
04:01.44zavalaI've building a SCCP setup and the 7960 will register with *, but when I try to make a phone call it only takes the first number dialed and then returns a fast busy
04:02.15zavalafrom * I can issue 'dial 3000' and it rings the SCCP 7960...
04:02.26zavalabut I can't call anything from the 7960
04:02.27ast_new-Bjaike: that's heavy asterisk server
04:02.36jaikeast: actually we have multiple asterisk servers
04:02.46jaikeall interconnected
04:02.55zavalaI figured it was in a dialplan somehwere... but I've been snooping and SCCP doesn't pickup the dialplan.xml and beyond that I have nowhere left to look...
04:03.09zavalaMath' any ideas?
04:03.24Math`whats the output ob the cli
04:03.28Math`on
04:03.49zavalawith debug on it shows that the phone only transmitted one number..
04:03.55zavalaso I know it's not * dropping it
04:04.16zavalano debug it looks like:
04:04.22zavala<PROTECTED>
04:04.27zavala<PROTECTED>
04:04.30ast_new-Bjaike: i'm really in newbie, so i have a lot of questions but i'll first reading about voip
04:04.46ast_new-Band asterisk the miracle
04:04.46zavalawhen in reality I dialed 8500 and hit 'dial'
04:05.09jaikeast: took me a while to get the hang of it...been doing asterisk for almost a year now
04:05.18jaikebut its exciting
04:05.50*** join/#asterisk centrix (n=centrix@dhcp148.wireless.fiber.dcdi.net)
04:06.03ast_new-Bjaike: is there any problem with asterisk running now ?
04:06.35Kattyhi.
04:06.39zavalahello
04:07.19jaikeast: 1.0.* was buggy before..but 1.2.* seems pretty stable
04:07.35Math`zavala: did you set the proper context in skinny.conf?
04:08.23jaikezavala: experienced the same thing with our polycoms...send the number before the complete number was dialed
04:08.38Math`with polycoms?
04:08.38zavalaI'm using chan_sccp
04:08.46jaikehad to disable digitmapping...dunno if 7960 has something like it
04:08.47zavalajaike: what as the resolution?
04:09.11mswwhen you set the digitmap properly in the polycom, it works great
04:09.12Math`jaike: skinny has its name for a reason, its very very lite
04:09.13jaikenow...you have to press send to dial..so you can key in as many numbers as you like
04:09.19mswthe default digitmap is no good though
04:09.32Math`no digit map, nothing, just sends digits 1 by 1 to the sccp server
04:09.57jaikeive not experience with skinny..sorry
04:10.01zavalatrazz.... cant pm, not registered
04:10.01jaikeno
04:10.33zavalayeah... math' .... if I lead off with an * or a # it will take multi-digits (even though they go nowhere)... if I lead of with a number, any number, I only get to dial one
04:10.37Trazzzavala, simple do /msg nickserv resgister somepassword
04:10.53welleszavala, thanks very much
04:11.04zavalano problem wells
04:11.12welleszavala, the problems fixed
04:11.22*** join/#asterisk _SwM_ (n=admin@digitaldatabits.net)
04:12.38a1faomfg
04:12.39a1fa;)
04:12.43a1faeverybody stop for a second
04:13.35zavalaso what's the rush a1fa?
04:13.44jaike?
04:14.52a1fa:P
04:14.55a1fatwo things
04:15.07a1faanybody tried the voice changer patch on asterisk 1.2.1?
04:15.10a1fa2nd thing,
04:15.19*** join/#asterisk Jammy (i=jammy@CPE0008740429bc-CM001404df6f46.cpe.net.cable.rogers.com)
04:15.19a1faudp: 2727, 4520, and 4569?
04:15.29a1fawhats on 2727 4520, 4569
04:16.21chet~docs
04:16.25jboti heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
04:16.50jaikeisnt 4569 iax2
04:16.51jaike?
04:19.01Luke-JrAny idea on recovering a deadish PAP2-NA? ethernet/blue & red lights are dead
04:19.22zavalalukee-jr: I have no idea what  you're talking about
04:19.37a1fayeah
04:19.41a1fa4569 is iax
04:19.46a1fabut, what is 2727?
04:19.47Jammyhey guys, got a quick question, I always get this error parsing a certain section of my zapata.conf file 'Ouch ... error while writing audio data: : Broken pipe' cant seem to figure out what keeps causing it to break at that point.. from the point on, it tries to configure the internal zap fxs
04:19.52a1fa~2727
04:19.56a1fa~4520
04:20.01a1fai am trying to patch this server down
04:20.18jaikefirewalls huh
04:20.26Jammyhmmm..heh.. nvr mind
04:20.28Jammyfixed itself
04:20.51jaike2727 i think is mgcp
04:21.12jaikenot sure though
04:22.15a1facan i disable mgcp?
04:22.28a1fai had somebody trying to bruteforce my asterisk
04:22.39Math`unload chan_mgcp.so
04:23.01Math`or add a noload directive for chan_mgcp.so in modules.conf
04:23.23zavalabingo noload=chan_mgcp
04:24.35a1fawhat does mgcp do?
04:24.42remmortmf
04:24.53a1faufcs
04:24.59*** part/#asterisk warthog (n=nvadekar@69.17.198.58)
04:25.00a1faufcsdttrtmf
04:25.03a1faget it?
04:25.09remmoyes
04:25.11remmodid you?
04:25.16a1fayes
04:25.22zavalano, seriously... some level of rtfm has to be done here..
04:25.26remmoso then you worked it out
04:25.44a1faah
04:25.46remmocan someone tie my shoe laces? my arms are too short
04:25.51a1fano dude
04:25.57a1fayour iq is below your shoe size
04:26.07a1faits ok, daddy will hook you up
04:26.10a1falay back
04:26.19remmomaybe i should bend over for you?
04:26.30a1fano, i aint that kind of daddy
04:26.54zavalaoh give me a break... mgcp -- media gateway control protocol
04:27.11a1fayeah
04:27.14a1fai just read it
04:27.18a1fai dont need that
04:27.35zavalaif you don't know what it's for, your clearly can't miss it when you turn it of then can you?
04:27.40remmoanyone using vierling gsm gateways here?
04:28.09a1fawhat is on port 4520?
04:28.24a1fathere is nothing in documentation about it
04:28.59remmomaybe dundi
04:29.13a1fa2727 is mgcp
04:29.24remmoDUNDI
04:30.01gohif using leased line connection do I able connect 37 branch together with any lake?
04:30.08a1fayup
04:30.10a1faits dundi
04:30.22a1faso chan_dundi?
04:31.12zavalait's not a channel driver..
04:31.24zavalajust comment it out in the dundi.conf
04:31.41a1fa:*
04:32.18zavalahowever, after having just looked at the conf.. it's on port 4520 by default
04:32.25a1faeverything is commented out
04:32.34rajivexten => 500,2,VoiceMailMain(s${CALLERIDNUM})  results in callers hearing 'goodbye' then hangup. why might that be?
04:32.36a1fabut it still binds to the port
04:33.00a1farajiv: what version? ${CALLERIDNUM} is not used anymore
04:33.10a1fa${CALLERIDNUM} is obsolete
04:33.12rajiv1.0.10
04:33.17a1fawell maybe not
04:33.26a1fawhen did they take off ${CALLERIDNUM}?
04:33.29a1fa1.2.0 tree?
04:34.05rajivthe variable is working correctly as i see in the console debug: Executing VoiceMailMain("SIP/110-38ff", "s110")
04:34.17a1facool
04:34.27a1faits obsolete in 1.2.x
04:34.36a1fado you have voicemail.conf configured
04:35.04Trazzzavala, can i force the zaptel to get installed in /opt/voip too ?
04:35.31zavalayes you can... however it's best to just let it install in the usual places as it needs to modify the kernel, etc
04:35.43rajiva1fa: http://pastebin.com/507712 is the logs ... voicemailmain() with no args works just fine so i doubt it is a voicemail.conf issue
04:36.48a1fai see
04:37.02a1fabeats me.. it could be a bug in 1.0.10
04:38.03Trazzi would rather have it install in that same directory .. what do i need to modify for that zavala?
04:38.35zavalaone last request.. and then I'll stop beating the horse and leave.... anyone have any working knowledge of the Cisco SCCP image on cisco phones?
04:39.20a1fazavala: fyi : pbx_dundi is the module
04:39.47zavalaa1fa, thanks
04:39.54a1fanp :p
04:40.05a1fazavala: run netstat -nap | grep asterisk
04:40.16a1faand paste to pastebin.ca
04:41.15zavalatcp        0      0 10.40.6.40:2000         10.40.15.51:51241       ESTABLISHED18716/asterisk
04:41.15zavalaudp        0      0 0.0.0.0:2727            0.0.0.0:*                          18716/asterisk
04:41.15zavalaudp        0      0 0.0.0.0:4520            0.0.0.0:*                          18716/asterisk
04:41.16zavalaudp        0      0 0.0.0.0:5060            0.0.0.0:*                          18716/asterisk
04:41.18zavalaudp        0      0 0.0.0.0:4569            0.0.0.0:*                          18716/asterisk
04:41.20Math`pastebin!!!
04:41.20zavalaunix  2      [ ACC ]     STREAM     LISTENING     46606    18716/asterisk      /opt/voip/var/run/asterisk.ctl
04:41.21Math`pastebin!!!
04:41.23zavaladoh
04:41.28a1fayeah man
04:41.31Math`:P
04:41.38a1fado you really need that much exposure
04:42.02*** join/#asterisk rhousand (n=rhousand@rrcs-24-199-246-10.midsouth.biz.rr.com)
04:42.03a1faJan 10 09:57:21 NOTICE[29465] chan_sip.c: Registration from '<sip:90093@213.249.97.123>' failed for '202.164.44.75' - Username/auth name misma
04:42.11a1fasome guy kept trying to brake in
04:42.45a1fathere is like half a million entries
04:43.07a1fatoo bad my passwords are uber hashed + md5 stored
04:43.09Math`iptables -A INPUT -s 213.249.97.123 -j DROP
04:43.13a1fano
04:43.22a1faiptables -A INPUT -j DROP
04:43.25Math`lol
04:43.44Luke-JrAny idea on recovering a PAP2-NA? ethernet/blue & power/red lights are lit solid
04:43.49rajivwacky. if i put a wait(1) in between answer() and voicemailmain(s${CALLERIDNUM}) then things works properly
04:43.49a1fa$IPTABLES -A INPUT -p tcp -j REJECT --reject-with tcp-reset
04:43.49a1fa$IPTABLES -A INPUT -j REJECT
04:44.02a1fathats the way to go man
04:44.16Math`yeah except -j REJECT uses your bandwidth if you get dos'd
04:44.24a1fawell true
04:44.36a1fagotta keep it clean tho
04:44.52a1faso it dont show up in port scanning as "ports filtred"
04:45.11a1fa:P
04:45.30a1fabtw, his ip changed 10 times
04:45.36a1fadifferent sources/subnets
04:45.42a1faso i only allowed my subnet
04:47.17a1fasmart way to do it
04:47.21a1faalso allowedguest=no
04:47.30*** join/#asterisk GarryH (n=guangyao@S0106009027bbc526.ed.shawcable.net)
04:47.34a1fai suggest everybody set that to no
04:48.34zavalaa1fa if you want to get technical about it... stand the SER server up infront of asterisk and then make SIP users authenticate to an LDAP or Kerberos backend
04:49.09a1fazavala: not a bad idea
04:49.16a1fasLDAP rather
04:49.19zavalayeah
04:49.21zavala:_D
04:49.39a1fai am running my pbx from internet colo
04:50.02a1fatime=35.8 ms
04:50.05zavalaahhh... there is a recipe out there somewhere to stand up SER infront of asaterisk on the same box
04:50.05a1fato my sip peer
04:50.10zavalanice times though
04:50.19zavala;-)
04:50.41a1fathats only my asterisk->sip provider
04:50.46a1fabut my home->asterisk
04:50.47a1fa19ms
04:50.48a1fa:P
04:50.50a1faits still slow
04:50.50zavalabugger.. this sccp thing is going to drive me up the damn wall... I've been looking for an answer for the past three hours
04:51.01zavalawhat's total time?
04:51.07zavalaend to end?
04:51.53a1fadunno
04:51.55a1fabout 50s
04:51.57a1fa50ms
04:52.09a1fafreaking broadvoice
04:52.13a1fatheir shit is always slow tho
04:52.18zavalahahaa...
04:52.24zavalaever tried VoicePulse?
04:52.57a1fai've heard about them
04:53.02a1fabut they dont offer the plan i need
04:53.07a1fafree western europe :P
04:53.12a1fafor $19
04:53.21zavalaahh... nope... broadvoice it is..
04:53.57a1fa<PROTECTED>
04:54.04a1fais the fastest gateway they have
04:54.09a1faand it is 35ms
04:54.13*** join/#asterisk bkw__ (n=brian@adsl-70-142-59-48.dsl.tul2ok.sbcglobal.net)
04:54.19Math`its not that bad
04:55.49*** join/#asterisk [hC] (n=hardcore@S0106000e9b96114f.vf.shawcable.net)
04:56.13a1fayeah
04:56.22a1fai need to send a SIP phone to bosnjia
04:56.25a1faBosnia
04:56.32a1fai wonder if they will have problems connecting to my SIP
04:56.41a1fasince they will be getting 100ms round trip
04:56.51zavalaouch....
04:57.19Math`I've a VSAT SIP phone working #1
04:57.20*** join/#asterisk liew (n=liew@60.49.6.190)
04:57.26Math`as long as you got no packet loss, you're ok
04:57.31Math`the round trip is about 500ms
04:57.46a1fai hope not
04:57.52Luke-JrIs there a decent priced open source replacement for a PAP2-NA?
04:57.53a1faudp is not very reliable :P
04:58.02liewmay I know who is from malaysia that able help me implement asterisk for my client?
04:58.08a1faLuke-Jr: i bought a PAP2-NA for $50!!
04:58.17Luke-Jra1fa: where?
04:58.22Math`I bought a mediatrix 2102 for 35$cad (ebay:P)
04:58.23a1fawww.voipsupply.com
04:58.29Math`(worth 150$)
04:58.32Luke-Jra1fa: $60 there...
04:58.40a1fawell $10 more will not kill you
04:58.51a1fago to OfficeDepot and pickup PAP2
04:58.52a1faand hack it
04:58.54Luke-Jra1fa: well, I was hoping to find something with open firmware
04:59.01a1fahm!
04:59.05a1fathere is no such thing :P
04:59.08Luke-Jr:\
04:59.10Luke-Jrwhy not?
04:59.17*** join/#asterisk goh (n=goh@60.49.6.190)
04:59.20a1faLuke-Jr: asterisk is open :P
05:00.00*** join/#asterisk tengulre11 (n=tengulre@61.185.224.66)
05:00.08Luke-Jra1fa: but it's not a low-power device, nor cheap for ATA abilities
05:00.48a1fawell IaxY
05:00.49a1fathen
05:00.57a1faget iaxy
05:01.01a1faits open source
05:01.03wunderkinpa168 phones or something like that
05:01.07h3xit aint open source
05:01.11a1fait aint?
05:01.12h3xno
05:01.14a1fabastards
05:01.16h3xthe fuckin firmware is binary
05:01.47zavalaon the other side of the coin, everybody's got to put food on the table some how
05:01.49a1fasuX
05:02.05a1faliew: how much are you paying for remote assistance?
05:02.20Math`they get money for the hardware, whats the use of not opening the platform
05:02.25a1fapaypal that $$$ upfront.. and you get remote administration
05:02.55liewa1fa, where is ur stay right now? I want local people for assistance
05:03.08liewa1fa, ar u own a company
05:03.13zavalabecause there is nothing really special in the iaxy other then how they impliment the code
05:03.57a1faliew: yes..
05:04.10a1faliew: You get remote assitance.. and you can watch what I am doing, and learn
05:04.11liewar you from Malaysia?
05:04.14a1fano
05:04.22a1fa:P
05:04.56liewthat mean I still need to configure all the thing until can remote access
05:06.25a1famoney upfront :P
05:10.36*** join/#asterisk mkl1525 (n=daniel@212.80.239.117)
05:13.57*** join/#asterisk _-_ (n=nabudoco@206.135.48.98)
05:15.47zavalafor just some simple sip to sip phone calls inside of one network.. it shouldn't take you more then 30 minuts to an hour after you read the docs
05:17.45*** join/#asterisk jcrock7 (n=jared@ool-44c1bb21.dyn.optonline.net)
05:20.16a1fahm
05:20.21a1faanybody used that dialpad editor
05:20.26a1fajava editor
05:22.34a1faLuke-Jr: http://www.tomsnetworking.com/Sections-article153.php
05:22.37a1facheck this shit out man
05:22.41a1fajust what you were looking for
05:22.44a1faembedded pbx
05:23.26*** join/#asterisk jcrock7 (n=jared@ool-44c1bb21.dyn.optonline.net)
05:31.34*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
05:34.07*** join/#asterisk wellng (n=welles@222.90.141.49)
05:34.46Math`thats nice
05:35.25*** join/#asterisk cyber (n=kani@220.247.245.2)
05:38.24*** join/#asterisk slan (n=lba@user-12lml5g.cable.mindspring.com)
05:42.39a1fayah
05:42.43a1fatoo weak tho
05:42.56a1facant do anything with that bad boy but make 1 call at the time
05:42.56a1fa:P
05:43.08jbroomeyou need a cluster of them! :)
05:43.10a1fagood if you have 400 of these
05:43.17a1fathats what i was about to say
05:43.25a1faAsterCluster
05:43.26Math`you can go up to 6 calls in ulaw
05:43.31Math`if you read the article thru :P
05:43.34a1fai did
05:43.50Luke-JrAre PAP2s currently unlockable?
05:44.01a1faLuke-Jr: http://www.tomsnetworking.com/Sections-article153.php
05:44.02a1fayes
05:44.09a1fathey are unlockable
05:44.34Luke-Jrwhat's the current method? :)
05:44.41Math`its on the wiki
05:44.48Luke-Jrwhich wiki?
05:44.51*** join/#asterisk Medvekoma (i=bear@funyiro.webpress.hu)
05:44.59a1favoip-info.org
05:45.02Math`http://www.voip-info.org/wiki/view/Linksys+PAP2+Unlocking+Methods
05:45.16*** join/#asterisk |omni| (n=rob@net98.limelyte.net)
05:45.56*** join/#asterisk GiRL[23] (i=SeRDaR@server.ivinskis.kursenai.lm.lt)
05:46.36*** join/#asterisk Fire (n=Lady_Han@server.ivinskis.kursenai.lm.lt)
05:46.38a1fahttp://www.voip-info.org/wiki/view/Linksys+PAP2+Unlocking+Methods
05:48.02a1fanight
05:48.31*** part/#asterisk franck (n=franck@tikiwiki/franck)
05:49.19*** join/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com)
05:49.42*** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net)
05:49.53Netgeeksdigium mail lists down?
05:50.58Netgeekswow, quiet here too, I knew the aliens were gonna come and abduct all the asterisk people
05:51.22*** join/#asterisk yeawsing (i=yeawsing@218.50.182.187)
05:52.26yeawsingHi I have a question about License Fee for Asterisk.  Anyone able to help me out
05:52.42Trazzif i am going to use pure sip and no hardware how do i get teh extensions configured ?
05:54.49Math`in sip.conf
05:54.58Math`yeawsing: whats the question
05:55.10*** join/#asterisk MatsK (n=mk@3.80-203-81.nextgentel.com)
05:56.08yeawsingIf I use the Asterisk Server for commercial use, do i need to buy the license $995 from digium.
05:56.10TrazzMath, what do i need in there? I have the x-lite up and running and logged in but cant dial anyting because i didnt configure any extensions i dont think
05:57.22Math`Trazz: well configure them
05:57.24Math`yeawsing: no
05:57.29Trazzi get this on the cli
05:57.29TrazzJan 16 00:09:19 NOTICE[5415]: pbx.c:1731 pbx_extension_helper: Cannot find extension context 'internal'
05:57.57Math`Trazz: you need to define extensions contexts, then point your SIP devices to that context
05:58.00QwellTrazz: Then the context internal doesn't exist
05:58.30*** part/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com)
05:58.35Math`yeawsing: you can use asterisk for any purpose without having to buy Asterisk Business Edition from digium
05:59.01*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)
05:59.55*** join/#asterisk Netgeeks (n=root@68-185-24-8.static.mdfd.or.charter.com)
06:00.27yeawsingok thanks Math
06:01.32*** join/#asterisk pengyong (n=lala@222.188.141.234)
06:01.47*** join/#asterisk Netgeeks (n=chris@68-185-24-8.static.mdfd.or.charter.com)
06:02.59yeawsingIf I need more 120 simultaneous call, do i need to purchase the License.
06:03.18Netgeeksno
06:03.27Math`yeawsing: *ANY* purpose :P
06:03.45Netgeeksyou need to purchase the license if you want to have digium provide you software support
06:03.52Math`you do need to purchase licenses for g729 channels (at 10$/channel)
06:04.11justinuyikes
06:04.12yeawsingI just want to make sure that I do the proper way.
06:04.15Math`digium is gonna give support anyways... just going to charge you per-hour
06:04.25justinuwhat does digium charge/hr?
06:04.51Math`I have no idea
06:05.45Math`In addition to the free installation support for Digium hardware, Digium provides full support for the entire Asterisk software suite, including hourly rates with no commitments. For more information on commercial Asterisk support or any other Digium professional service, please contact sales@digium.com or call us toll free at 877-LINUX-ME (877-546-8963).
06:06.57yeawsingWhen I search the Digium License, the FAQ and brochures answer is not specified or very open.
06:08.13yeawsingSo let say I want to purchase the License, is that mean $995 is just per/server or I can install many server using SINGLE license
06:08.22*** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca)
06:09.25yeawsingI thought Digium Sales team is here.
06:10.00Math`you should contact sales directly
06:10.22yeawsingya. I guess so.  Thanks Math, appreciate that.
06:10.29Netgeeksit's currently sunday evening around midnight at digium.  Even if some of their sales team hangs out here, I doubt they are here now
06:10.40yeawsingha..
06:10.42justinuwussies
06:11.07Netgeekslol
06:11.49Math`lol
06:14.21yeawsingIf I want to be a service provider using Asterisk to make PC to PSTN call, what do I need and who should I connect to?
06:14.43Qwellyou need knowledge of telephony
06:14.50Qwellin other words...you'd know the answer to that question
06:15.42justinulol
06:16.09yeawsingWhen I read on the net, they mention about IXC.
06:16.46justinucall someone like at&t and order a PRi
06:17.49yeawsingSo r u guys connecting to at&t,sprint or MCI
06:18.04justinunone of the above
06:18.47yeawsingSo how can we make cheap long distance call?
06:19.02justinuyou need money
06:19.08yeawsing:)
06:19.19justinucredito
06:19.33justinuprobably $50,000
06:20.02Luke-Jryeawsing: Voipjet or Voxee seem good
06:20.05yeawsingis that for buying Hardware or buying time.
06:20.11justinubuying time
06:20.13Luke-Jryeawsing: Voipjet is 1.3 cents/min or Voxee is 1.1 cents/min
06:20.14justinuhardware is cheap
06:20.39justinuyou need a rock solid IP link
06:20.40Luke-Jryeawsing: but they're both VoIP, so you'll need bandwidth too
06:20.47yeawsingI've check with VoIPjet, but I still not sure about their call quality
06:20.58Math`voipjet is great
06:21.13justinui'm talking about rates 1/4 of that
06:21.18justinumaybe 1/6th
06:21.19*** join/#asterisk Medvekoma (i=bear@funyiro.webpress.hu)
06:21.32Luke-Jrjustinu: where?
06:21.37yeawsingIt seem to be VoIPjet is very popular.
06:21.48justinuwholesale, level3 or global crossing
06:22.05Math`justinu: IP or PRI?
06:22.21justinuip
06:22.41justinupri is a big PITA
06:23.23Math`and you say level3 can do 0.0025$/min to us/ca?
06:24.25yeawsingWhen u guys talking about Level3 is that mean Layer
06:24.30Luke-Jrno
06:25.00yeawsingOr just the H/W Level3,4 and 5
06:25.05Math`Level3 is a company :P
06:25.14Math`www.level3.nety
06:25.16Math`www.level3.net
06:25.27justinumath: yeah, around that
06:25.31yeawsingoh. thanks for clear that up.
06:26.20h3xlevel3 costs a hell of a lot more than that
06:26.23Math`justinu: wow, contacting level3's sales dep tomorrow :P
06:26.27h3xunless you commit to 100 grand a month
06:26.36justinuso what's the problem?
06:26.46yeawsingwow
06:27.24h3xand it sucks coz it aint two way
06:27.24justinueh?
06:27.26Math`eh?
06:27.31h3xinbound only
06:27.32justinuorigination and termination are seperate
06:27.35Math`go on the website, it says both origination and termination
06:27.39*** join/#asterisk dasuberdavid (n=david@pcp01534754pcs.huntsv01.al.comcast.net)
06:27.41h3xtermination requires you to lock it to a device basically
06:27.55h3xand its # minutes per device/account
06:27.56justinuyou use sip proxies
06:28.06yeawsingSo if we want to connect to PSTN we can use VoIPjet.
06:28.07justinui dunno where you get that
06:28.24h3xl3 has two different wholesale products
06:28.33h3xthe one with inbound, outbound, and e911 is turnkey
06:28.40h3xlike the one that packet 8 uses
06:28.58h3xbut at that point all you are doing is pushing paper
06:29.12justinuoh, we're not doing that
06:29.13h3xtheir LI product is by the minute, but they dont have decent outbound termination to accompany itr
06:29.27h3xyou may as well use long distance termination
06:29.46*** join/#asterisk rkioko (n=rkioko@196.200.26.42)
06:30.03yeawsingIs VoIPjet can consider as long distance termination
06:30.16h3xvoipjet illegally exports your calls to canada :P
06:30.37h3xgrey market anyway
06:30.46yeawsingoh..
06:30.56h3xthats why its so cheap
06:31.13Math`illegally export calls....
06:31.20Luke-Jrillegally?
06:31.20Math`how come
06:31.25yeawsingbut we can use it to make call using our Asterisk server
06:31.29h3xwell its going to another country and coming back
06:31.32Math`yeawsing: yeah you can
06:31.47Luke-Jrh3x: nothing illegal there...
06:31.57Math`h3x: nothing illegal there but... where is it going? :P
06:32.11h3xwell
06:32.20h3xit might be since voipjet dosent have a FCC 214 license
06:32.21h3xbut its voip so
06:32.33mishehuand the mpaa says that you're illegally copying movies when you record them off of the tv.
06:32.34Math`voipjet is canadian
06:32.37*** join/#asterisk dasuberdavid (n=dasuberd@gateway.digium.com)
06:32.42Luke-Jrpfft, FCC
06:32.43Math`mishehu: lol
06:32.46mishehuillegal is in the eye of the beholder.
06:32.56mishehuor the gavel of the judge
06:33.12h3xthey have equipment in the us
06:33.12h3xheh
06:33.14Math`and I can say nobody has the right to log the text I say without my express written consentement
06:33.28Math`and everyone here that is logging this channel is performing an illegal act passible of jail
06:33.37yeawsingha
06:33.38mishehuMath`: you can say that, but enforce it?
06:33.39mishehupssh
06:33.43mishehuI'm logging you anyway.
06:33.45Math`well... maybe :P
06:33.46Luke-JrMath`: bs
06:34.29mishehunow, you do the Math`
06:34.32mishehu*rimshot*
06:35.06Math`lol
06:35.47jaikethis channel is logged..i saw a page somewhere with all the messages here
06:35.54yeawsingMath what provider are u using if u don't mind I asking?
06:36.08jaikeeven read my old messages
06:36.18Math`yeawsing: for what?
06:36.19justinuthere's some interesting drug talk in those logs
06:36.20Math`termination? dids?
06:36.21Trazzhow do i give my x-lite two extensions so i can call myself?
06:36.31Math`Trazz: you can call your own extension
06:36.39Math`its just gonna ring on another line
06:36.48Math`tho its pointless
06:36.51Trazzok
06:37.33*** join/#asterisk ThaZZa_Work (n=me@203.80.44.200)
06:41.21*** join/#asterisk Myk3 (i=Myk3@cpe-67-9-95-36.hot.res.rr.com)
06:41.24Myk3hello all
06:41.38TrazzMath, i was trying to call myself to see how voicemail works
06:41.44Trazzit never picks up
06:41.46Myk3i have 2 lines how can i configure each number to ring a different phone?
06:42.19Math`Trazz: well you need to configure your dialplan properly
06:42.48Myk3when both numbers are dialed only one of the phones ring
06:43.08Myk3can i set it up to ring different numbers for differenet phones?
06:43.14Math`DIal(TECH/ext1&TECH/ext2)
06:43.15Myk3"useing softs to test"
06:43.53Myk3any thoughts?
06:45.03Myk3anyone?
06:46.50Myk3ok there where people talking now there aint did i scar them away?
06:46.54*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
06:48.08TrazzMath, is voicemail and ivr built in and turned on by default ?
06:48.26Math`you need to tell asterisk everything...
06:48.37Math`it wont go to voicemail if you don't pickup automaticly
06:48.40Math`you need to make it go there
06:48.55Math`Dial(SIP/phone1|30)
06:49.02Math`VoiceMail(u1000) (for example)
06:49.12Myk3how can i make asterisk got this phone for this number?
06:49.15Trazzok
06:51.05Trazzso that would be exten => 244,2,Dial(SIP/sam,30) and exten => 244,2,VoiceMail(u1000) ?
06:51.26Myk3anyone?
06:51.32Myk3help please
06:53.31yeawsingThanks guys.  Talk to u guys later.
06:53.34*** join/#asterisk liew (n=liew@60.49.6.190)
06:53.45*** part/#asterisk yeawsing (i=yeawsing@218.50.182.187)
06:54.35*** part/#asterisk liew (n=liew@60.49.6.190)
06:54.50*** join/#asterisk goh (n=goh@60.49.6.190)
06:55.05TrazzMath, does that look correct?
06:55.06Trazzso that would be exten => 244,2,Dial(SIP/sam,30) and exten => 244,2,VoiceMail(u1000) ?
07:00.15QwellTrazz: no, you can't have two of the same priority
07:00.44Trazzok gotta make it 3 then
07:02.46*** join/#asterisk welles (n=welles@61.150.60.123)
07:05.32*** join/#asterisk meriad_ (i=Turtle@24.83.211.78)
07:05.47*** part/#asterisk meriad_ (i=Turtle@24.83.211.78)
07:06.48ast_new-Bsomebody please help me with dialplan that limit call by duration, group ext that can dial in long distance or international ?
07:09.01*** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au)
07:10.06*** join/#asterisk jahani (n=k@adsl196-206-241-217-196.adsl196-16.iam.net.ma)
07:10.21jahanihi
07:10.56jahanifrom where i can get the codec g723.1 for asterisk ?
07:12.02*** join/#asterisk bzbw (n=wlwzhang@ip-33-107-134-202.rev.dyxnet.com)
07:12.32gohfor extension call to another extension within same asterisk pbx will audio path go though asterisk server? are the audio path go peer to peer. and only the sip state will go through asterisk server?
07:12.46bzbwhas anyone connect * to a traditional PBX?
07:13.06bzbwthis PBX require to dial 9 before getting to the outside line
07:13.36bzbwbut even if I send a 9, the tranditional pbx seems just looping back to my *.
07:13.42bzbwit really puzzles me.
07:15.02ast_new-Btry to set ignorepat => 9;
07:15.07ast_new-Bin exentions.conf
07:15.48*** join/#asterisk leto3 (n=l@car75-1-81-57-13-34.fbx.proxad.net)
07:18.04*** join/#asterisk EriSan (n=erisan@151.8.109.90)
07:20.03jahaniwhere i can find the codec g723.1 for asterisk ?
07:21.13Qwelljahani: legally?
07:21.42Qwelltip: nowhere
07:22.42jahaniyes
07:22.46jahanilegally
07:23.40TrazzQwell, any idea why i get these errors?
07:23.40TrazzJan 16 01:35:05 NOTICE[6347]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
07:23.40TrazzJan 16 01:35:05 WARNING[6347]: pbx.c:1690 pbx_extension_helper: No application 'VoiceMail2' for extension (macro-vmessage, s, 1)
07:24.17ThaZZa_Workhey all.
07:24.19GarryHexit
07:24.22*** part/#asterisk GarryH (n=guangyao@S0106009027bbc526.ed.shawcable.net)
07:24.51ThaZZa_WorkHow do i fix this? Accepting UNAUTHENTICATED call from X.X.X.X:
07:25.35QwellThaZZa_Work: give the user a password
07:25.58QwellTrazz: VoiceMail2 obviously doesn't exist, and something is wrong with your Dial line, or...simply, the phone isn't reachable
07:26.32Qwelljahani: please don't msg me
07:26.35Qwelland the answer is, you don't
07:27.22ThaZZa_WorkQwell: It is an * box connecting via iax to another * box. Should i have 2 register commands.. One at both ends?
07:27.47jahaniso how can i use g723.1 with asterisk ?
07:27.55Qwell* boxes shouldn't really register with each other...make them static
07:28.02Qwelljahani: passthrough, and that's it
07:28.23jahaniwhat u mean by passthrough ?
07:28.51Qwellasterisk doesn't decode/encode the audio, it just passes it through
07:30.27jahanii know but where i get the codec i buy buy from digium g729 but i need also g723.1
07:30.33ThaZZa_WorkQwell: Can you refresh memory on where to look to make static?
07:30.43Qwelljahani: You don't
07:30.56jahanii don't ??
07:31.01QwellThaZZa_Work: iax.conf, just give them passwords, and put in the IPs instead of host=dynamic
07:31.09Qwelljahani: for the third time, right
07:31.28jahanii don't understand what u mean by i don't !!!
07:31.45QwellYou don't use g723 with asterisk, unless it's done in passthrough
07:32.15jahanihow to passthrough ?
07:32.18ThaZZa_WorkQwell: Cool thanks mate.. :D
07:32.33Qwellcall another phone that does g723
07:34.10jahaniits not work
07:34.26*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-118.claranet.co.uk)
07:34.38jahaniit say commune client and gateway have no commun codec
07:35.02Qwellthen g723 isn't an allowed codec
07:35.11ThaZZa_Workjahani: sounds to me like both phones dont' have a compatable set of codec's to use.
07:35.25QwellThaZZa_Work: You'd think that would be obvious
07:35.30niZonpoor Qwell, bombarded with questions
07:35.56ThaZZa_WorkQwell: I saw it, and i am classifed as dumb. lol
07:36.05*** join/#asterisk svenna_ (n=svenna@p548D09D9.dip0.t-ipconnect.de)
07:36.17*** join/#asterisk zAmifage (n=g4l3ku5@c-67-187-20-28.hsd1.tx.comcast.net)
07:36.25ThaZZa_WorkniZon: Poor Qwell earning his keep of being allow to get really drunk tonight.. ;-)
07:37.09PoincareIf I want to do a Dial(SIP/user&ZAP/g1/1234), is it possible to set outgoing callerid only for the ZAP part?
07:37.17jahaniok
07:37.25QwellPoincare: I don't think so
07:37.34Poincarebummer :-(
07:37.39Qwellbut I don't know for certain
07:37.43*** join/#asterisk tehdely (n=delysiid@home.teambarry.org)
07:38.18QwellPoincare: You might be able to dial a Local/ exten, which sets cid
07:38.22tehdelylilo: ---> zAmifage spamming in every chan
07:38.52PoincareQwell: can i do a Dial(SIP/user&somelocalextension) then?
07:39.09Qwellso like, replace Zap/g1/1234 with Local/1234, and have like exten => 1234,1,Set(CALLERID(number)=12345) then 1234,2,Dial(Zap/g1/1234)
07:39.12QwellThat may work, but...ymmv
07:39.33Poincarei will try...
07:39.39*** join/#asterisk Fire (i=user226@server.ivinskis.kursenai.lm.lt)
07:39.49Poincarethe LOCAL has to be in the same context?
07:40.00QwellYou can do Local/1234@context
07:40.10Poincareok, I'll let you know :-)
07:41.17*** join/#asterisk Oden (n=rhousand@rrcs-24-199-246-10.midsouth.biz.rr.com)
07:45.13*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
07:45.53*** join/#asterisk _vic (n=riccardo@gw-fi.esaote.com)
07:46.10zAmifageWhoa look at this.. http://www.progenic.com/vote/?id=Galekus
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07:46.21zAmifagehttp://www.progenic.com/vote/?id=Galekus
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07:46.38Qwelltime for a DDoS :P
07:46.39zAmifageCheck it out man! http://www.progenic.com/vote/?id=Galekus
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07:46.48MrChimpyop me!
07:46.49zAmifageCheck it out man! http://www.progenic.com/vote/?id=Galekus
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07:46.57_vicshut it down
07:47.05_vicyeah !
07:47.06*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)
07:47.07MrChimpyyeah. good
07:47.41_vic(only thinked.  i have black magic  0=)
07:48.23*** join/#asterisk _vic (n=riccardo@gw-fi.esaote.com)
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07:49.24*** part/#asterisk tehdely (n=delysiid@home.teambarry.org)
07:53.33[av]bani=0
07:56.10PoincareQwell: it works! Thanks :-) Let me know when you're in the neighborhood, I'll buy you some drinks...
08:02.32*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
08:02.37*** join/#asterisk wellng (n=welles@61.150.60.123)
08:02.55ThaZZa_Workanyone use festival?
08:04.59NetgeeksSo, is the digium mail lists down?  It's been awfully quiet in my inbox
08:05.19QwellNetgeeks: would seem that way
08:06.02NetgeeksGlad to see my nightmare of a world conspiracy against me is once again delayed!
08:09.13*** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it)
08:09.23QwellNetgeeks: the conspirists took the weekend off for the holiday
08:09.54MrChimpycurrently worrying about connecting TDM400 to PBX here. blowing stuff up would not be popular.
08:10.14*** part/#asterisk jaike (n=a@203.131.137.76)
08:10.28MrChimpyfloor ports are RJ45, need to go to RJ11 on the card
08:11.04MrChimpyi shall let our telephone dude worry about it
08:12.31*** join/#asterisk eivindtr (n=wingnut-@cC3012269.inet.catch.no)
08:13.35justinuRj11's plug right into RJ45s, you know
08:13.47NetgeeksMrChimpy don't worry, the TDM400 card is safe, it's too cheap a component to be the one to blow up.  Just ding around find the most expensive or most unavailable component, and it will be the one to go
08:14.00Netgeeksdig round
08:14.07Netgeeksaround
08:14.15Netgeeksdamn beer and fingers
08:14.53Qwelljustinu: new cards are rj11 I hear
08:15.08NetgeeksJust because the rj11 will plug into a rj45 doesn't mean they are wired right
08:16.51dippo_man i wish there was some way i could communicate with a registered sip phone and get it to dial a number and engage the speakerphone
08:17.04dippo_i have a reigstered sip device in close proximity to an alarm on a door that was just tripped
08:17.09dippo_i'd love to hear if there's anything going on :P
08:18.30Netgeeksif you had a cisco7960 for example and set up one of it's lines as a speakerphone auto answer, you could just dial that line
08:18.38Netgeekslot of if's there
08:18.46dippo_yeah
08:18.53dippo_this is a grandstream budgetone
08:19.04dippo_not exactly equipped with a long list of extras
08:19.26Netgeekshowever, I always make the assumption that I'm not the only person in the world who configures one line on all my cisco 7960s as auto-answer speakerphone
08:19.36Netgeeksyep, not much there in that budgetel
08:20.27dippo_hm
08:20.31dippo_interestingly there is auto answer
08:20.35NetgeeksBut don't worry, no one is there, if they opened the door, the draft from the movement of the door alone would have blown the budgettone off it's desk/table and if it fell more than 2 cm, it would have broken adn un-registered
08:20.37dippo_<PROTECTED>
08:20.39dippo_the call and turn the speaker on
08:20.44dippo_heh
08:21.09Netgeeksor I should say, failed to re-register given you have the reg timeout at some sane value
08:21.18dippo_i am sure it was a false alarm. the cops said they secured the area (10 mins after the alarm was tripped, on the second floor of a warehouse downtown behind 3 other locked doors. yeah, right.)
08:21.34Netgeeksunfortunately you probably don't want the speaker auto-answer to be it's single line default mode of operation
08:22.03dippo_indeed
08:22.08dippo_but i can change it temporarily from here
08:22.17dippo_here goes nothin
08:22.19Netgeeksnow I have an open web browser staring me in the face and I done forgot what I was going to have it do
08:22.28*** join/#asterisk welles (n=welles@61.150.60.123)
08:22.31Netgeekscalling the phone?  maybe the perp will answer?
08:22.39Snake-Eyeshappens to the best of us
08:22.41dippo_bingo
08:23.32dippo_all quiet on the western front
08:23.48Netgeeksthats good news for the eastern front then
08:23.53MrChimpynetgeeks: yep. not so worried about tdm400, more worried about our PBX :)
08:24.33justinurj11 and rj45 wiring are compatible
08:24.40Qwelldippo_: That may very well have been the first report of asterisk being used to detect burglars :P
08:24.46MrChimpyat least the wiring is standard for the TE411
08:24.50NetgeeksI would suggest the handy use of a butt set to verify the right configuration on both sides before hooking them up
08:25.04MrChimpyju: but physically incompatible
08:25.17Netgeeksyou should be able to become pretty comfortable you've got everything right with a little checking
08:25.24dippo_heh
08:25.30dippo_pretty handy, i have to say
08:26.06MrChimpyyeah. difficult though when I don't know if the TDM will give the right signals, given I don't have it configured through not connecting it to anything :)
08:26.15MrChimpyI got the lights to come on. that was good.
08:26.17Netgeeksthey say you can generally solve the physically incompatability with copious amounts of alcohol
08:26.31justinuanyone know the difference between a Tie Trunk and a DID trunk?
08:26.35justinuin traditional PBXes
08:26.52NetgeeksI could only guess
08:26.57Netgeeksthus I won't
08:27.23Netgeeksbut my empty browser has a google input... lemme see what it says
08:27.50Netgeekstie trunk: A telephone line that directly connects two private branch exchanges (PBXs).
08:28.07MrChimpywhilst people seem awake, when I do the TDM411 stuff I'll need to be able to tell what number the caller dialled, so I can tell which service to direct them to (not using extensions). What does asterisk term this functionality?
08:28.26*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
08:28.33Qwellcalled number identification, I think
08:28.42MrChimpysweet. ta.
08:28.49Qwell~cnid
08:29.04justinuDNIS, usually
08:29.08Qwell~dnis
08:29.10jbotdnis is probably A telephone service that identifies the number that the caller dialed for the receiver of the call. DNIS is a common feature of 800 and 900 services, and can identify the number originally dialed when multiple 800 or 900 numbers terminate on the same destination trunks. DNIS works by passing the dialed number to the destination device, which can ...
08:29.25Qwellwhich can?!
08:29.32justinucliff hanger
08:29.52MrChimpyok, i'll search for both. how exciting.
08:29.55*** join/#asterisk wellng (n=welles@61.150.60.123)
08:29.56Netgeeksbad jbot
08:29.59Qwellwhich can act upon this data to control its routing, queuing, IVR, or other call behavior. DNIS is typically used to separate call treatment for different inbound campaigns or help desk numbers, whether in one enterprise or at a service bureau.
08:30.03*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
08:30.14MrChimpyDNIS is exactly what I mean.
08:30.19justinuDNIS is when you have a block of 100 DIDs on a PRI
08:30.32justinuit's how you tell which DID someone dialed
08:30.35Qwellnot always possible with analog lines
08:30.49Qwellbut, with analog lines it's kinda easy
08:30.53Netgeeks*shrug* Analog lines are the devil spawn
08:30.54MrChimpyyeah, this'll be on E1s though
08:30.55Qwellyou know which port it's coming from
08:31.06*** join/#asterisk ReX (n=ReX@AMarseille-252-1-13-217.w83-197.abo.wanadoo.fr)
08:31.10ReXhi all
08:31.14justinuthey have/had analog DID lines at one time
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08:31.29NetgeeksHello ReX from france
08:31.29justinubut i think those are all gone
08:31.54ReXis anybody can help me about chan_btp compilation plz??
08:32.10Netgeeksnot I, sorry
08:32.16MrChimpyanyone actually configged the TE411? is it especially tricky?
08:32.29Netgeeksshouldn't be
08:32.36MrChimpyi hope not :)
08:32.51Netgeeksyou've configured other digium T1/E1 cards?
08:32.57MrChimpynope.
08:33.02Netgeeksah, okay
08:33.18MrChimpyi'm a poor linux engineer who has been thrown at this stuff
08:33.27Netgeeksfollow the example confs, pretty simple, you set up your T1/E1 parameters in /etc/zaptel.conf
08:34.01Netgeeksfor t1 that would be the esf,b8zs stuff, for E1 it's some C's and S's but I don't remember off the top of my head
08:34.07MrChimpycool. attempting with dev card and normal lines first
08:34.26Netgeeksyou also tell it how to configure the channels in /etc/zaptel.conf as well
08:34.44ReXi have connected my cellphone on asterisk by BT, i just want asterisk dial and forward call when i call the cellphone number, is it possible?
08:34.50Netgeeksonce you are done, (make sure you load the right module, someone here can pipe up the exact module name)
08:35.02justinuthe digium cards aren't bad to setup
08:35.18Netgeeksand then run ztcfg -v, you shouldn't get any errors, if you do, something is amiss
08:35.28MrChimpyi'll doc what I do. i'm a good chimp.
08:35.41Netgeeksthen I always run zttool just to make sure I've got no red or blue or yuellow alarms (with the T1/E1 connected)
08:36.11Netgeeksonce you get that far, you can move on to zapata.conf which is not as simple, but if you can read you should be fine
08:36.36MrChimpyi can read. it's thinking that makes my head hurt.
08:36.48Netgeeksdrink beer, it helps
08:37.04Netgeeksit helps stop the hurt, that is... for a while
08:37.10Netgeeksdoesn't help the thinking tho
08:37.12MrChimpyit's 8.30AM here. i try not to start so early.
08:37.42Netgeekshrm, NZ, AU?
08:37.52Netgeekswait no, wrong side
08:38.16NetgeeksGMTish
08:38.27MrChimpyUK
08:38.55MrChimpyLondon to be precise
08:40.03NetgeeksAh, one of my not so favorite places.  The two times I visited it was over 90 degrees F (sorry dunno what that is in celcius) and I had a tiny little hotel room with no AC and a window that opened to the exaust from some seafood restraunt
08:40.53MrChimpysounds familiar
08:40.58Netgeekshaha
08:41.38MrChimpyair conditioning is like strange voodoo to us. the tube during summer is so pleasant.
08:42.20NetgeeksI did however end up sitting at dinner right next to George Lucas and his daughter
08:42.43MrChimpystill, it's feckin freezing or raining so it's bearable for the one week we actually get summer
08:42.59iDunnohmm
08:43.47NetgeeksThen I got to ride this thing you called a train from london to Middlesborough...  I called it a can of smoke...  *sigh*
08:44.00Netgeeksbut middlesbrough (spelling) was quite nice
08:44.53iDunnoYou have: tempF(90)
08:44.53iDunnoYou want: tempC
08:44.53iDunno<PROTECTED>
08:45.04Netgeeksthanks iDunno
08:45.06iDunno:)
08:45.25MrChimpyaye. out of the big smoke places are quite nice. i'm just here for the cash ;)
08:45.27iDunnonow it's time to go to work :)
08:45.34Netgeeksit was the fifth decimal place 2 that broke the camels back... if it was just 32.22220 I would have been fine
08:46.11MrChimpyyou have a very specific comfort zone.
08:46.32Netgeekshow true that is....
08:47.06MrChimpyhm. 15 mins until everyone else gets in and starts bothering me
08:47.31NetgeeksI moved to this little speck on the map because it has very nice weather year around, well, good luck
08:47.40NetgeeksI'm going to hit the sack myself
08:47.44MrChimpywhere are you?
08:48.11NetgeeksBrookings, Oregon, about 30 meters from a cliff that drops into the pacific
08:48.31MrChimpyah nice.
08:48.39MrChimpywell, later....
08:48.48MrChimpythanks for the help
08:48.54Netgeekshope it works out okay
08:49.15MrChimpyi'll let you know in about a week when i've torn my remaining hair out
08:49.41Netgeeksit grows back
08:49.45Netgeeksmost of it
08:51.03benjkvoipbuster announced end of life
08:51.19benjkoh well
08:53.49ReXnobody knows "Ext. Phone >> Cellphone BT (incoming call) >> Asterisk >> VoIP (outgoing call)" here? (without the use of cellsocket of course)
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08:59.12benjkcellsocket won't work in the UK, its US only
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08:59.22benjkyou need a GSM gateway
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08:59.41benjkSiemens makes those
08:59.45benjkand some company in CZ
09:00.33ReXthx benjk, i try to configure asterisk with a bluetooth cell phone, but i don't arrive to compile chan_btp, i don't know what is the problem
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09:02.53ReXchan_bluetooth is installed and works fine (http://www.crazygreek.co.uk/content/chan_bluetooth)
09:03.25ReXbut for the incoming call to cellphone, i suppose i need chan_btp
09:03.51ReXvery sorry for my bad english :((
09:08.50ReXbenjk, do you mean this one http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html is a good way?
09:11.30SkumlingI've got af simple question... how can I forward a call on asterisk?
09:11.48[av]banibring a shrubbery
09:11.48SkumlingI'm using ISDN phones connected through a HFC card in NT-mode
09:12.13thazzaReX: Hmmm.. Yummmy.. bluetooth to asterisk.. :D
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09:15.21llirkhi
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09:20.17bsbhi all! question to all, cisco 7914 is support SIP?
09:20.31justinui've stumbled into another timezone
09:20.40ReXSkumling, exten => AAAAAAAAAA,1,Dial(SIP/BBBBBBBBBB@yourcontext), where AAAAAAAAA is your incoming call do you want to fw and BBBBBBBBBB, your forwarded number, i suppose
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09:22.38SkumlingReX: uhm okay, well now I just realize, that what I ment was "how do I transfer a call" :)
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09:24.39benjkReX: yes, voiceblue, that's the one
09:25.37ReXif i don't arrive to install chan_btp, i must to buy a gateway gsm, but it's more expensive :(
09:25.46benjkas I understand it, it goes GSM network ----GSM---> Voiceblue-box ---SIP---> Asterisk
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09:26.07benjkits more expensive, but it's solid
09:26.24nicksinGood morning all
09:26.32ReXi suppose the simcard must to be introduce into this gateway gsm
09:26.35benjkchan_btp may not even do what you want
09:26.48ReXsure?
09:26.50benjkbecause it is only for detecting *presence*
09:27.07benjkhence btp, p for presence
09:27.19benjkso Asterisk knows that you are in the vicinity
09:27.48benjkthere is also a bluetooth thingie to make a bluetooth headset or phone a terminal of Asterisk
09:27.51ReXI thought that chan_btp was the same as SIP for the bluetooth connexion
09:28.28benjkwell I am not sure, but last time I looked at it this was the status back then
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09:29.05benjkanyway, I think a proper gateway is the only solid thing to do
09:29.05ReXwich is the configuration file for incoming call by bluetooth, do you know?
09:29.15benjkno
09:29.19ReXokay
09:29.26ReXthx a lot benjk
09:29.37benjkand yes on the SIM card, the gateway needs to have it inserted
09:29.46benjkthere are also gateways with multiple SIMs
09:29.54ReXcool
09:29.56benjkso you could have one SIM for each network
09:30.07ReXvery cool :o))
09:30.52benjkthis is meant for least cost routing but these days with LNP on mobile networks I am not sure how you (or that gateway) would know which number belongs to which network
09:32.12*** part/#asterisk secure75 (n=mic@p549A18C0.dip0.t-ipconnect.de)
09:32.33benjkanyway, I am getting a new Bluetooth capable mobile soon so if you check back in a while I will be able to tell you more about what the Bluetooth add-ons for Asterisk can do
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09:34.37nicksinIm looking for some assistance in configuring 2 duxbury ISDN bri modems in a box. 1 for incoming and the other to the old PABX.
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09:36.31ReXbenjk (private msg) ;)
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09:45.10littleballhello, any good tutorial about real time?
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09:47.57MrChimpyholy crap! I got a dial tone and busy tone out of my dev kit. wonders will never cease.
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10:00.31jahaniwhat mean this error : Spawn extension (users-sip, h, 1) exited non-zero on 'SIP/1001-710a' ?
10:01.52trixterthe hangup extention for users-sip exited with a value other than 0, it normally means that there was an error in what it called
10:02.01trixterbut often can be ignored even if there was an error
10:02.16*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
10:02.24trixterdial iirc returns -1 if there is a hangup, app_conference I know returns 0 if the user hangs up
10:02.33trixterer doesnt return 0
10:02.37trixter0 is only if they press #
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10:04.28jahaniwhen i call its ring 1 time and the call is terminate
10:05.06puzzledmorning all
10:06.32Modcutsmorning, if i have two sip lines, one of which has a local geographic number on it, does it depend on the provider if i want the incoming line to work with both sip lines?
10:08.29bn-7bcit seem that asterisk only captures the first key in the keymap so when I hare  * for atxfer and *2 for atomon non of them worked, is this a known issue in v1.2.1?
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10:15.55festr_hello
10:16.03festr_isnt there problems with digium mailing lists?
10:16.16festr_i've last message from date 15.1.
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10:21.49MrChimpyhmm!
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10:22.21MrChimpyi have handset connected and getting dial tone etc from ast dev card
10:22.36MrChimpyast can see me pick up and put down the phone
10:22.56*** join/#asterisk ]]]papatya[[[[[ (i=E-sHeee@server.ivinskis.kursenai.lm.lt)
10:23.04MrChimpyhandset is on channel 1
10:23.19MrChimpyin zapata.conf I have
10:23.27MrChimpy[channels]
10:23.27MrChimpycontext=test
10:23.27MrChimpyusecallerid=yes
10:23.27MrChimpycidsignalling=v23
10:23.27MrChimpyhidecallerid=no
10:23.28MrChimpyimmediate=no
10:23.30MrChimpy<PROTECTED>
10:23.32MrChimpycontext=internal
10:23.34MrChimpysignalling=fxo_ks
10:23.36MrChimpyechocancel=yes
10:23.38MrChimpygroup=1
10:23.40MrChimpychannel=1
10:24.10MrChimpyI have an internal config in dialplan
10:24.35MrChimpywhcih answers, does a background(enter-ext-of-person), waits and hangs up
10:25.08MrChimpyit doesn't though. I start ast and all I ever get on picking up the handset is a dialtone, then busy if i dial anything
10:25.31MrChimpyany pointers?
10:26.45*** join/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net)
10:26.56ckruetzeHi
10:27.38ckruetzeI've a question regarding sendURL(). is sending and url something only * can do or is that sip standard?
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10:29.14trixteris it a sip standard even?
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10:33.12thazzaMrChimpy: What happens in the CLI? Have you asterisk -vvvvvvr
10:35.34MrChimpycor. that'd be rather verbose then
10:35.52MrChimpyi pick up, it says -- Starting simple switch on 'Zap/1-1'
10:36.37thazzaMrChimpy: When you have issues.. The more verbose the better.
10:36.46MrChimpyaye :)
10:37.00MrChimpyall I see is :
10:37.07MrChimpy<PROTECTED>
10:37.07MrChimpy<PROTECTED>
10:37.24MrChimpythat corresponds with me picking up and hanging up
10:38.49MrChimpyshow channels
10:38.49MrChimpyChannel              Location             State   Application(Data)
10:38.49MrChimpyZap/1-1              s@internal:1         Rsrvd   (None)
10:38.49MrChimpy1 active channel
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10:39.54MrChimpyso that looks fine. maybe it's just audio playing that's broken. the file is there... does it normally work out of the box? it's just one of the standard ones.
10:40.30MrChimpy[internal]
10:40.30MrChimpyexten => s,1,Answer()
10:40.30MrChimpyexten => s,2,Background(enter-ext-of-person)
10:40.30MrChimpyexten => s,3,Wait(2)
10:40.30MrChimpyexten => s,4,Hangup()
10:40.37*** join/#asterisk jluk (n=jon@80-235-135-92.cable.ubr07.nail.blueyonder.co.uk)
10:40.59MrChimpyit's in /var/lib/asterisk/sounds/enter-ext-of-person.gsm
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10:42.12MrChimpyi did set my defaultzone to uk, so maybe it's a country thing?
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10:49.14MrChimpyaha!
10:49.33MrChimpyi added echo test to extension 600 - that works
10:49.46MrChimpyit's s that doesn't work
10:49.55cricalixwoodhi. Does anyone know if there is a problem with the Asterisk-users mailing list? I have not received anything since noon (GMT) yesterday.
10:50.24MrChimpycric: apparently there is
10:50.50cricalixwoodok, thanks. Just confirming that it was not my end that was at fault.
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10:53.57agxhello, i cannot use rx_fax i get this message " ast_set_read_format: Unable to find a path from slin to unknown" ... have i to enable some extra codec or format in modules.conf? i'm using an isdn card via chan_modem to handle it
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11:00.51agx<agx> hello, i cannot use rx_fax i get this message " ast_set_read_format: Unable to find a path from slin to unknown" ... have i to enable some extra codec or format in modules.conf? i'm using an isdn card via chan_modem to handle it
11:01.25*** join/#asterisk LoRez (i=lorez@freenode/staff/lorez)
11:02.26MrChimpydoes the standard demo setup play a sample when you pick up the handset?
11:02.28effape_so we have any zap experts in here?
11:02.50effape_so/do
11:03.02pengyongany receive maillist from asterisk?
11:03.09trixterbbs news is covering the labia presidency
11:03.11trixterer liberia
11:03.14pengyongi don't got mails since last day
11:04.09trixterno one is getting mail from the lists
11:05.33*** join/#asterisk rowter (n=woot@201.145.5.26)
11:06.35tzafrir_laptopeffape_, they're out for lunch. Ask your question anyway
11:06.59MrChimpycurrently I have the demo set up working, but I have to press 2 to get away from dial tone. (this is to handset connected to dev card)
11:07.10MrChimpyis that normal
11:07.12MrChimpy?
11:08.08RoyKhm
11:08.16RoyKis there such a thing as a billsec timeout?
11:08.26RoyKabsolutetimeout isn't really suited for callingcards
11:08.33effape_aha ok. Basically the problem i'm having is with a tdm400p. When i get an incomming call with cidstart=polarity the call doesn't get routed to sip but if it's cidstart=usehist it does. However with usehist when i pickup the sip phone it gets a polarity switch and disconnects the zap channel
11:08.43effape_it's all very odd
11:11.18*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
11:11.19effape_pastebin.com/507986 seems to be where it all happens
11:11.45effape_any ideas?
11:12.33pifhi, anyone tried using alaw files for musiconhold?
11:12.50pifwith good results?
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11:14.14effape_tzafrir_laptop who should i be asking about it?
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11:19.30trixterRoyK: execute a macro on connect that sets the timeout from the connection perspective -- absolutetimeout(current used - current available) which should equal bill seconds left
11:19.42trixterer flip those two numbers ... that would result in a negative number likely
11:20.01trixterwell actually shouldnt that be a plus instead?  I think so
11:20.04rowter!seen
11:21.17ReXis anybody can help me about cellphone BT connexion plz?
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11:22.16ReXi've installed asterisk with chan_bluetooth, the cellphone is correctly connected and i can make outgoing call
11:22.24ReXmy problem is for incoming call
11:23.51ReXwich configuration file i must to use to configure incoming bluetooth calls?
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11:24.39ReXI thought that chan_btp was the same as SIP for the bluetooth connexion, but i'me not sure and i don't arrive to compile chan_btp
11:24.42RoyKtrixter: absolutetimeout includes call setup and ringing, which is somehow sub-optimal
11:25.54trixterRoyK: ok, then use the macro, although I am not sure how to get the currently used time, other than maybe some variable set when the call is answered
11:26.01trixterwhich should be accurate +/-1 sec
11:26.26RoyKperhaps i should write a TIMEOUT(billsec)
11:26.47trixterperhaps I was looking at this from the 5 minute solution perspective :P
11:27.08RoyKi'm using absolutetimeout how
11:27.27RoyKbut it's not good enough for a commercial solution
11:27.29RoyKimho
11:27.49trixterwell if that is your argument there are a lot of other things that need to be tossed out as well :P
11:28.17trixtera macro that sets the timeout based on currently used time + available time and that macro is called when dial connects
11:29.01trixterseems simple enough until something else cna be done, if this isnt something that has to be fixed in under 5 minutes then a proper solution would be better although take longer, especially for the testing phase
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11:36.36MrChimpy<PROTECTED>
11:37.23trixtercongrats
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11:42.15effape_i'm guessing noone yet then ;)
11:42.56RoyKhm
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11:44.17effape_i mean really if i just knew who i was looking out for :)
11:45.06backblueanyone with dundi?
11:45.28muteffape_?
11:46.06muto nvm
11:47.22effape_basically why is this happening? http://pastebin.com/507986
11:47.53effape_when i answer the sip it drops the zap chanel
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12:07.28effape_sigh
12:08.58fugitivoeffape_: i'm sure the problem is codec related
12:09.13fugitivowhat codec is your sip device using?
12:09.36thazzaeffape_: Looks like you have reverse poltirty hangup turned on. Try turning it off.
12:09.56effape_yeah but if i turn it off it never gets dropped.
12:10.03fugitivoeffape_: codec
12:10.21effape_What i don't understand is why it gets a reverse polarity on sip pickup.,
12:10.26effape_just checking codec now
12:10.43effape_i have allow=ulaw and alaw
12:10.46llirkif I want to use the newest version of ooh323 with 1.2.1, is it best i pull down the latest CVS? or am i able to just upgrade the channel driver itself?
12:10.56fugitivoeffape_: disallow=all allow=ulaw for your sip device?
12:11.04thazzaeffape_: Perhaps a hardward issue. this line makes me wonder. -  Exception on 23, channel 3
12:11.06effape_yeah
12:11.17fugitivoJan 16 03:30:06 DEBUG[2325]: Ooh, format changed from unknown to ulaw
12:11.24effape_yeah but that follows the ulaw
12:11.25effape_yeah
12:11.38fugitivoit's a problem with codecs
12:12.04effape_what should i try?
12:12.06fugitivowhat sip device are you using?
12:12.26effape_for testing its xlite but in production it will be snom360
12:12.40fugitivosetup xlite to use only ulaw
12:14.30effape_i'm sorry but how do i do that? :/ Can't find option
12:14.46effape_is that g711u?
12:16.10fulgasyes
12:16.35effape_aha cool. Cheers. I can't try that just yet but i'll give it a go
12:16.49effape_would that be why it's getting polarity reversal though?
12:17.19sivanamorning
12:18.00effape_mornin'
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12:28.37[av]banianyone done a kind of 'virtual extensions'? where employees log in and out of a phone
12:29.05[av]banieg employee sits down at a phone, hits *something to log in, then his pin and access code
12:29.12[av]baniwhen he leaves, hits *something to log out
12:29.29sivana[av]bani: you can do that with queues
12:29.35[av]baniugh
12:29.43thazza[av]bani: Sounds like queues and agent logins
12:30.04[av]baniand if i dont want to use queues :)  that would mean i'd have to setup a queue for every individual employee?
12:30.06thazza[av]bani: Yet i suppose you could do it.
12:30.59mutooooooweeee
12:31.02[av]baniit would be a sort of roaming feature, like you can do with wifi
12:31.05muti made some strong coffee this morn
12:31.47*** join/#asterisk znoG_ (n=gs@33-138-114-200.fibertel.com.ar)
12:32.30fugitivo[av]bani: use the phone login
12:32.39[av]banieh
12:33.29mutjust make some whacked out dialplan for it
12:33.38[av]banifugitivo: ???
12:34.48mutdbput 1234 5432
12:34.49*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
12:34.53mutdbput 5432 1234
12:35.01mutdial 5432. dbgeet 5432
12:35.03mutdial 1234
12:35.05mut:P
12:35.19[av]banio_O
12:35.37mutheh well it's an option
12:36.56[av]banifugitivo: what do you mean use the phone login?
12:37.17muthe means login to the phone, wherever it's logged in, assuming it's sip or some such and not analog
12:37.40effape_yeah i know the snom 360's i have have a login button to enter username and password
12:37.48fugitivoif it's analog he can use the ata/gateway login
12:37.52effape_rather than config in the phone to a fixed login
12:38.10*** join/#asterisk amir (n=amir@gentoo/developer/amir)
12:38.17mutfugitivo: the what?
12:38.58[av]banifugitivo: eh?
12:39.21mutfugitivo: analog being like a channel bank
12:39.32fugitivomut: oh
12:39.38mutnot a phone hooked to an ata
12:39.43[av]banilogin to the phone, you mean webinterface and change the config?
12:39.54mut[av]bani: man you're good
12:40.14[av]baniso give the employees the admin pw to the phone...
12:40.38muteh they're all options
12:40.42mutwe've given ya a few
12:40.46fugitivoso? it's just a phone
12:40.48fugitivonot a server
12:40.50effape_no
12:40.52mutquite an array of choices if i do say so
12:41.04[av]baniwell the queue idea sucks the least
12:41.16effape_on the snom phones you have a screen. You can set them to require login
12:41.18[av]banithe others are varying degrees of horrible
12:41.30mutwell thems ya options
12:41.34fugitivoyou can code a web interface to post the data to the webinterface of the phone
12:41.44fugitivoso you don't need to give admin passwd to your users
12:41.52mut^ there ya go
12:41.54[av]baniassuming they have access to a puter where the phone is, great
12:42.01*** join/#asterisk diego_br (n=diego@200.208.241.178)
12:42.05mutprogram a dial plan to do it
12:42.09mutand execute an agi script
12:42.17fugitivothere you go
12:42.18effape_am i just taking rubbish?
12:42.27fugitivothe agi script is going to post the data to the webinterface
12:42.36fugitivoand login the user to the phone
12:42.53muthot damn we make a good team fugitivo
12:42.55mut:P
12:42.56fugitivosee? more brains, more ideas
12:42.59backblueanyone with dundi?
12:43.00fugitivohell yeah
12:43.07[av]baniyeah, just wondering which is less work, setting up piles of queues or writing agis
12:43.21fugitivo[av]bani: agi is going to be easy
12:43.23mutboth are fairly simple..
12:43.33fugitivoi think i'm going to code it myself, i liked the idea
12:43.34[av]banii thought of agi but eh
12:43.52mutk well i must get to organizing my week of work
12:43.54mutbbfew
12:44.02fugitivome too, i have a meeting
12:44.03fugitivocya
12:44.13[av]banichanging config on phones all the time isnt really appealing
12:44.24[av]baniqueues seems more 'natural' for asterisk
12:44.25effape_why do you need to?
12:44.29fugitivonot config, just login information
12:44.33effape_yes
12:44.35fugitivothat's why it's there
12:44.37fugitivoto login users
12:49.42*** join/#asterisk linville (n=linville@azure.tuxdriver.com)
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12:54.56warthawgi have a really stupid question:  how much money can a small business < 50 employees save by using Asterisk instead of a commercial PBX?
12:55.14*** join/#asterisk rstandy (n=rastandy@d83-176-116-85.cust.tele2.it)
12:55.44fugitivowarthawg: that answer can't be answering because there're a lot of variables on each company
12:56.18warthawgfugitivo, thanks.  is it fair to say thousands of dollars would be typical?
12:56.30*** join/#asterisk pobre (n=seymore@203.215.73.192)
12:56.33fugitivowarthawg: again, it depends on your company
12:56.48warthawgok, gracias amigo, i hope they don't catch you
12:57.18fugitivowarthawg: the idea is not saving costs, but adding funcionalities at less cost
12:57.25*** join/#asterisk joelsolanki (i=joelsola@202.160.161.93)
12:57.29warthawgaha, ok
12:57.31xhelioxIs there any particular reason why an IAXy wouldn't natively bridge (IAXy -> * -> IAX Carrier) --- they're using the same codec on both sides... Asterisk says "attempting to xfer" and then fails, that's the only error that's received.
12:58.23mutanyone ever expiramented with using ucspi-tcp with asterisk?
12:58.41mutblock annoying users who login 100 times a minute and fail
12:59.27*** join/#asterisk many (i=many@krikkit.ukeer.de)
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13:00.34manyheyya.  is there anything to obeye if one uses asterisk+misdn+(bristuff+)rxfax? it seems rxfax connects correctly when being called by sip, but there is silence when being called by misdn
13:00.48*** join/#asterisk littleball (n=littleba@cm78.epsilon175.maxonline.com.sg)
13:00.56Asterisk_newbies
13:01.04littleballhelo. does asterisk 1.2.1 support realtime on postgresql?
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13:02.22Asterisk_newbiehello anyone have a manual on howto install asterisk on debian and postgresql as database
13:05.15fugitivogood, people using postgresql and not the other crappy toy sql server :)
13:08.23littleballfugitivo, but UnixODBC postgresql is bad, right
13:08.24littleball?
13:08.33*** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com)
13:08.46fugitivoi don't use realtime, for me, realtime is bad
13:09.44muti wish i had time to dev my own module
13:10.21mutaltho there is a appmysql isn't there
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13:13.54RoyK~seen zoa
13:14.04jbotzoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 3d 16h 14m 13s ago, saying: 'apt-get install libssl-dev'.
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13:15.52Drukenmorning peoples
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13:19.16hackeronhey, is there anything built in to asterisk that displays the average latency, the % packet loss, etc?
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13:25.05pigpenin the cli: sip show peers
13:25.13pigpenshows you the latency...no packet loss.
13:25.49trixterhackeron: no there isnt that will show packet loss and jitter and such, latency based on each qualify is shown if you set qualify=yes (or a non 0 value) in your sip.conf
13:26.13*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)
13:26.26trixterthere are add-ons to ethereal and such that will show jitter, as for packet loss that is harder to see from a network perspective, the sniffer can tell if there is a missing sequence number but not what was actually received
13:26.28*** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net)
13:26.40hackerontrixter: our grandstream phones crap out if I set qualify=yes :(
13:27.40trixterwell the latency shown by qualify=yes is not network latency only, it includes the time it takes to generate sip packets and such since it measures that by sending an invite
13:28.03RoyKer
13:28.05RoyKwtf?
13:28.05RoyK<PROTECTED>
13:28.36pigpentrixter, thanks for the clarification....I learned something...
13:29.30trixterum ok
13:29.39trixtersounded like you already knew that based on your earlier comment
13:29.48*** join/#asterisk Soul (n=Soul@87-196-39-131.net.novis.pt)
13:31.09*** join/#asterisk CuTe^G (n=cuteg@202.63.195.79)
13:31.10pigpenI thought is was latency only..not the additional time to create the sip packet, etc..
13:31.44CuTe^Ghello guys i need help ... i m making Asterik base IP-PABX for domestic
13:31.45pigpennow the built in packet loss feature in the cli would be nice.
13:31.52CuTe^Gi need FXO cards
13:32.04pigpenhow many ports?
13:32.06CuTe^Gwhich 4 ports FXO PCI card which i best and cheap
13:32.13CuTe^G?
13:32.19CuTe^G4 / 8 / 16 FXO
13:32.20CuTe^Gany
13:32.26tzangerI only know of one four port FXO PCI card
13:32.36pigpenI haven't used any others then digium, they have a 4 port and the new 24 port modular
13:32.50pigpen4 port is modular too
13:33.07CuTe^Ghow in one PCI Card 24 ports ?
13:33.22CuTe^Gi m new user of Asterisk
13:33.25pigpenthe 24 port is in increments of 4 similar functions (fxo/fxs)
13:33.33pigpenI am ordering one today.
13:33.58pigpennice thing about the 24 port guy, is that you can get it with echo cancellation built on.
13:34.20fugitivopigpen: did you get one?
13:34.21CuTe^Gi need simple
13:34.36fugitivoCuTe^G: check the tdm2400
13:34.42pigpenordering one today.  but it sounds great....I hear it is better on faxing too.
13:34.43CuTe^G4 port FXO
13:34.43CuTe^Gi put in 4 PSTN Lines
13:34.55CuTe^Gand thru SIP Softphone my 4 agents will received the calls
13:34.58fugitivopigpen: really? it's not a big tdm400? ;)
13:35.02trixterI had considered adding a res_snmp thing which would require counters for things like packet loss, throughput, etc..  just never got around to it
13:35.13fugitivoCuTe^G: only 4? get a tdm400p with 4 fxo modules
13:35.16pigpenfugitivo, not from what I hear...it is a complete redesign.
13:35.27trixterI think that if those counters existed (no matter where) traps could be sent easily enough and certainly anything could pull the info
13:35.34CuTe^GTDM2400 how much ??
13:35.46CuTe^Gthen i need
13:35.49trixter$1800-2100 full depending on fxs or fxo
13:35.55fugitivoCuTe^G: maybe $300, where are you located?
13:35.56CuTe^GTDM400P FXO
13:35.59pigpenright.
13:36.03trixterthe tdm400p is much cheaper
13:36.09CuTe^GI am from PAKISTAN
13:36.15CuTe^Ghow much TDM400P
13:36.22fugitivoCuTe^G: no idea then
13:36.32trixterI think those run full about $300-400 incl shipping
13:36.44fugitivoa TDM04B (tdm400 with 4 fxo) is $450 in Argentina
13:36.45*** join/#asterisk Assid (n=assid@203.115.64.5)
13:36.46Assidheya
13:37.13trixterwell there may be duties on that which I dont consider shipping charges
13:37.22CuTe^Gand how much 1 FXO port ?
13:37.29fugitivo$180
13:37.31trixterif you ship one to the UK for example its 3% duty + 17% vat on top, ireland is 21% vat 3% duty
13:37.44trixterso that can add a considerable amount, I have no idea what, if any, pakistan would charge
13:37.49pigpendam!  I am glad I don't have to screw with that....
13:37.58fugitivotrixter: it's always better to find a local reseller
13:38.05fugitivotrixter: they'll have lower prices
13:38.13trixterthat may not always be an option
13:38.25Assidhrmm.. anyone here using a cisco 7960 ?
13:38.38trixterand I was selling electronics to europe mainly becuase I could undercut local resellers by $100 on the same exact item and still profit like $100
13:38.39fugitivoif i have to buy a card in the US, I have to add shipping + 50% custom taxes + 21$ iva (local taxes)
13:38.43trixtersometimes local resellers arent cheaper
13:38.49trixterespecially if there is little competition
13:39.08trixterwhere are you that there is 50% duties?
13:39.09CuTe^Gfor domestic call center TDM400 is best ?
13:39.14fugitivotrixter: Argentina
13:39.17trixterahh
13:39.28ReXanyone hre using a bluetooth cellphone for inbound and forward calls? ;)
13:39.33trixteryeah its 20-25% for europe so that isnt as bad but still...
13:39.45trixterI was selling mp3 players for a $100 profit and couldnt get em fast enough a couple years ago
13:39.49fugitivotrixter: custom taxes are a pain in the ass
13:40.18CuTe^Ganyone help me plz msg me in pvt
13:40.24trixterthe EU lets you do a little arbitrage though, once you get it into the EU the item can generally move around without additional duties
13:40.35trixterso you locate the cheapest country send em there then redistribute
13:40.40trixterbut that doesnt work well for 1 item
13:40.57CuTe^GAsterisk X100P FXO $25.95
13:40.59CuTe^GAsterisk X100P FXO $25.95 ???
13:41.21[TK]D-FenderCuTe^G : Avoid the X100.... How many lines are you looking at bringing into *?
13:41.22fugitivocustom taxes should exist for items made in your own country, if the item is not made in your country, why must i pay taxes for it?
13:41.31fugitivoCuTe^G: the tdm400 will be the best option
13:41.42CuTe^Gi need 8 lines
13:42.00fugitivonow you need 8?
13:42.01CuTe^Gthere is any 8 lines port PCI card for FXO ?
13:42.22fugitivono
13:43.19fugitivoCuTe^G: you have 2 options, 2 TDM04B for around $450 each, or a TDM2402 for around $1000 ($1200 with echocan)
13:43.23cfhis possible monitor the state of isdn line on a sip phone?
13:43.25DrukenCuTe^G: just the 24 :)
13:43.58CuTe^Gwhy 1 port FXO for 26$ and 4 port FXO TDM400P for 450$ ??
13:44.12fugitivoCuTe^G: the x100p is a bad card
13:44.19fugitivoCuTe^G: it's a cheap modem
13:44.33CuTe^GDigium TDM400P
13:44.35CuTe^G$141.40
13:44.35CuTe^GDigium TDM400P
13:44.35CuTe^G$141.40
13:44.46Drukenfugitivo: i tell ya... i use the damn x100p's more than the tdm, because my tdm is a peice of shat
13:45.06fugitivoCuTe^G: that's the card without modules, add 4 fxo modules
13:46.20fugitivoDruken: my x100p cards suck, maybe i got the bad ones
13:46.20fugitivoDruken: echo, no hangup, etc
13:46.20Drukenyou got the shitty x100p, i got the shitty tdm :)
13:46.20fugitivohehe
13:46.29SkumlingWhen using a Siemens Gigaset on at HFC-S in NT-mode, how can I then transfer a call? I think it's difficult to find docs on this, it seems that everyone but me knows how to it? When I hit # on the handset, I just hear the DTMF tones
13:46.32ManxPowerCuTe^G, That's $141.40 for the carrier card.  You still need the FXO modules
13:46.34[TK]D-FenderCuTe^G : Other options are FXO->SIP gateways (Mediatrix,AudioCodes,etc...) or a Sangoma A200 PCI solution (2 cards a little cheaper than TDM400 solution)
13:46.35Drukenfugitivo: my x100p's give me no echo, but my tdm echo's back my voice, and the tdm barely works...
13:46.47fugitivoSkumling: press flash on the phone
13:47.12fugitivoDruken: that's weird
13:47.18fugitivoDruken: you should return that card
13:47.31Skumlingfugitivo: the phone has no flash button... or... eh... hrm. Some of the handsets actually has...
13:47.35Drukeni've had it for a year, don't think they would take it back... hehe
13:47.39fugitivoSkumling: yes it has, i have one
13:48.15Skumlingfugitivo: there's many different handsets...
13:48.26ManxPowerSkumling, FLASH is sometimes called RECALL
13:48.37fugitivoSkumling: what model is your siemens?
13:48.58CuTe^Ganyone guide me complete solution i m very new
13:49.03Drukenflash or link in canada here :)
13:49.20*** join/#asterisk brockj49464 (n=brockj49@22.105.dhcp.hope.edu)
13:49.27Skumlingfugitivo: I've got 3000C, 3000S and 4000 Micro
13:49.31DrukenCuTe^G: www.voip-info.org bone up :)
13:49.35*** part/#asterisk brockj49464 (n=brockj49@22.105.dhcp.hope.edu)
13:49.52fugitivoCuTe^G: we already told you, 2 tdm400p with 4 fxo modules each, or an fxo->sip gateway (more expensive)
13:50.05fugitivoSkumling: i have the 4000
13:50.29fugitivoSkumling: and the flash button, is a small circle at the inferior left side
13:50.34fugitivoSkumling: with 2 arrows
13:51.14ManxPowerTelecom is expensive.  If you can't handle that then don't mess with it.
13:51.29fugitivoSkumling: did you find it? :)
13:52.23DrukenManxPower: are you in a mood this morning?? :)
13:52.27fugitivohehe
13:53.15ManxPowerDruken, I just get tired of newbies expecting to be able to put a bunch of ports in a box for less money than most of us spend on COFFEE in a week.
13:53.28fugitivolol
13:53.43DrukenManxPower: you must have an iv drip for your coffee....
13:54.02[TK]D-FenderManxPower : You clearly need a better coffee distributer!
13:54.08Drukenbut yeah i know what u mean :)
13:54.14fugitivoor he's going to die soon
13:54.15ManxPowerDruken, I was referring to people that go to a coffeehouse.
13:54.38ManxPower$4/latte x 5 days per week
13:54.40Drukentim hortons gets expensive :)
13:54.46MrChimpybored now. got my dev card working now it's dull.
13:54.49fugitivocoffee is going to kill you
13:55.01fugitivodrink tea or mate
13:55.02MrChimpywaiting for a server to stick the E1 card in
13:55.05ManxPowerThe nearest coffeehouse if like 30 miles from me.  I don't go to them very often.
13:55.07Kattymorning.
13:55.30IkarusManxPower: it is bullshit that that isn't the case yet, the actual technology isn't expensive, the cost of getting it tested is, bunch of rip-off test labs
13:55.34DrukenManxPower: get yourself an assistance, and send him to get you coffee :)
13:56.09tzangerManxPower: I just use the free company coffee. :-)
13:56.17ManxPowerDruken, Where can I get an assistant to work for free?
13:56.22Skumlingfugitivo: is it a 4000 C or M?
13:56.36tzangerand I make a point to say how much easier it is to get a coffee from Tim Horton's than Starbucks when I'm picking up my sweetheart's coffee
13:56.39fugitivoSkumling: errr
13:56.43tzanger(I call it her pretentious coffee, heh)
13:56.50fugitivoSkumling: it just says siemens 4000 at the back side
13:56.51[TK]D-FenderKatty: Mew.
13:56.56ManxPowerI brew my own.
13:57.28Skumlingfugitivo: okay... hrm... I'm gonna google for the specs on my handset... thanks
13:57.40fugitivoSkumling: you don't have that button?
13:57.45Katty[TK]D-Fender: :>
13:58.02Katty[TK]D-Fender: apparently everyone else is too busy to say hi.
13:58.12tzangerGood morning my dear
13:58.20Skumlingfugitivo: on my 3000C's I have a "R"-button - should be flash
13:58.23Kattytzanger: too late!
13:58.24DrukenManxPower: dunno, what countries have you looked in? i hear africa will work for mere food... :)
13:58.33Skumlingfugitivo: but on my 4000 M I don't seem to have that button...
13:58.38tzangerKatty: oh well, I'll have to make it up to you another time
13:58.45Kattytzanger: k
13:59.11fugitivoSkumling: on my 4000, i have 2 tiny circle buttons at sides of the mic, do you have those circle buttons?
13:59.35DrukenKatty: was on the phone, and in another program, so good mOOrning :)
14:00.05DrukenSkumling: wouldn't "r" be release?
14:00.10KattyDruken: (=
14:00.47trixterhi kattykat
14:00.51Ahrimanesanyone here done mass config of snom phones?
14:00.59*** join/#asterisk _rehash (n=rehash@ppp-48-118.atnr.ro)
14:02.36Kattytrixter: hihi.
14:04.17ManxPowerGads!  People are dying left and right.  First my significant other's father, now the mother of my tech contact at my largest customer.
14:06.11*** join/#asterisk pengyong (n=lala@222.185.197.28)
14:06.17RoyKanyone here running a rev1 te410p on 1.2?
14:06.56ManxPowerRoyK, I am.
14:07.01*** join/#asterisk NewSole (n=dave@d38-53-48.commercial1.cgocable.net)
14:07.03trixterKatty: hows it goin?
14:07.11trixterdid you may those mushroom thingys?
14:07.17tzangerRoyK: I am running oldschool TE405 with svn trunk
14:07.26Kattytrixter: i'm sorta in a weird awakeful state between naps.
14:07.29*** join/#asterisk paljas (n=paljas@sarastro.cs.uu.nl)
14:07.38trixterKatty: how cat like :P
14:07.39Skumlingfugitivo: seems like it's what they call "the net carrier key" on 4000 Micro :)
14:07.56Kattytrixter:  i have a new favorite quote!
14:08.01trixterwhats that?
14:08.02SkumlingDruken: here in Denmark, the R-keys generally equals to flash
14:08.03Kattytrixter: Now Men......Men are like a fine wine. They begin as grapes, and it's up to women to stomp the gutts out of them until they turn into something acceptable to have dinner with.
14:08.14trixterhaha
14:08.33trixterand if they dont find a woman to stomp them they shrivel up and turn into rasins?
14:08.36MrChimpydon't stomp on my grapes!
14:09.15*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
14:09.25trixterscience channel has a thing on now about the comet dust
14:10.04trixter<-- dreading when the mailing lists come back from the 1359013951 emails asking if the list is down
14:10.08*** join/#asterisk phos-phoros (n=James@unaffiliated/phos-phoros)
14:10.24Kattyhaha.
14:10.58trixterwow by placing quail brains in a chicken while its in the egg they have been able to get chickens to cluck like qualis and even mimik their movements and stuff
14:11.15*** join/#asterisk jahani (n=k@adsl196-206-241-217-196.adsl196-16.iam.net.ma)
14:11.42[TK]D-Fendertrixter : Big deal.... China made pigs that glow in the dark!
14:11.46MrChimpytrix: sounds worthwhile
14:11.56MrChimpychickens have terrible personalities
14:12.09trixterheh
14:12.36trixterwellk they also mapped like 30% of the wooly mamoth genome so far which means that jurisic park can have those too
14:13.16Kattyyou're about to - you're about to - you're about to enter an echo test!
14:13.23trixterthey just better get their operators correct and not miscount genders!
14:13.28Kattymy co-workers are having too much fun this morning.
14:13.33*** join/#asterisk basta (n=basta@213-156-52-98.fastres.net)
14:13.57ManxPowerI found a new way to torture users!
14:14.12tzangerManxPower: ?
14:14.50ManxPowertzanger, a script to disable their mailbox if 1) it has the default password or 2) they did not record all three (busy, unavail, name) greetings.
14:14.55bastais the mailing list server down ? I'm not receiving mails at asterisk-users since two days ...
14:15.13trixterbasta: nah it works fine
14:15.47tzangerManxPower: why would you do #2?
14:16.16ManxPowertzanger, We get 2 or 3 trouble tickets per week where the user complains about the "the user at extension xxx is busy" messages.
14:16.31ManxPoweror complains that they are not listed in Directory
14:16.35*** join/#asterisk welles (n=welles@222.90.92.113)
14:16.44tzangerhow is that a trouble ticket?  they didn't do what they needed to do
14:17.00ManxPowertzanger, Correct.
14:17.31ManxPowerThen we have to explain to them that only morons don't read the instructions we provided to them when the mailbox was setup.
14:17.32tzangerso you lock them out as a preemptive measure.  I like it.  :-)
14:17.49bn-7bcwhi is it tha res:heatures omly evere works with single key kombinations (ex: etxfer works hen i map it to * but not when mapped to *1)?
14:17.55tzangerwow the lists have been down for a couple days now
14:19.19bn-7bcdoes anyone else have the problem (nv 1.2.1)?
14:19.56sivanatzanger: you just wake up? :)
14:20.17*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
14:20.19tzangersivana: of course
14:20.22sivanaI'm coming down tomorrow... taking a sick day?
14:20.24trixterthis is silly..  this guy does a robbery, and kills his sons girlfriend who turned him in, then when in jail for life for that murder he orders the murder of witnesses for the murder he was serving time for.  Now that he has spent 23 years on death row fighting his case the argument is that 'he is old now so he shouldnt be put to death'..  while I am not commenting on whether or not its right or wrong for the death penalty I think that is a silly reas
14:20.24trixteron to ask for clemency.  which means late tonight/early tomorrow he will be put to death beucase the courts dont seem to accept that spending years fighting your case for which you are found guilty of when it involves multiple murders is grounds
14:20.31tzangerI can't, I have to be in Niagara Falls
14:20.54tzangerhahahaha
14:20.58tzangerthat is one fucked up situation
14:21.25trixterpersonally I think the death penalty should only be used when its cheaper, which 99% of the time its not so ...  and when sentenced to death you get an automatic appeal, most defendants are getting court appointed lawyers and it takes at least 10 years often longer to get all that done, so generally it costs 2-3 times what life would cost
14:22.31Skumlinghrm... how do I make an "s"-entry in my dialplan, that gives me a dialtone, and then allows me to enter the digits and call out?
14:22.34trixterbut the guy is legally blind, deaf, and now cant walk (all of those things he wasnt when he ordered the murder of witnesses, killed one testifying against him and the underlying robbery  that started it all..
14:22.50trixterSkumling: what medium are you connecting with?
14:23.00trixtervoip protocols typically send the whole number dialed not digit by digit
14:23.08Skumlingtrixter: ISDN phone via Zaptel
14:23.09trixterfxs ports will send digit by digit though
14:23.12SkumlingHFC
14:23.16ManxPowerSkumling, almost nobody needs to do that.
14:23.30trixterwhat signalling is the isdn doing?  doesnt that normally send the entire number as data?
14:23.40trixterbecuase of  the out of band signalling that is used
14:23.48trixteror does that one send it digit by digit?
14:24.17enemy^xWhat should "show hints" actually display incase of a client beeing on the phone? In my case it displays "241                 : SIP/241               State:Idle            Watchers  0" .....
14:24.26SkumlingManxPower: maybe... but humm, the issue is, that when I transfer a call, it runs the "s", when hitting flash immidately?
14:24.53ManxPowerSkumling, it should not do that unless you have something like immediate=yes
14:24.58*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
14:25.21SkumlingManxPower: ah...
14:25.22shmaltzlooks like the lists are down
14:25.36[TK]D-Fenderenemy^x :  8247                : SIP/8247              State:InUse           Watchers  0
14:26.05SkumlingManxPower: my zapata.conf has a immediate=no string...
14:26.32shmaltzSkumling, what type of zap card do you have?
14:26.36SkumlingHFC-S
14:26.40Skumlingusing Bristuff
14:26.57trixtershmaltz: nah they work fine
14:27.07shmaltzimmediate is used when the signaling (like DID) gets done in touchtones
14:27.17shmaltztrixter, what the lists?
14:27.22SkumlingI tried AMP, and when I had it running, I got a dialtone... but the dialplan in AMP is very complex (to me), and it really doesn't fit my needs, so I've decided to do custom config
14:27.23trixtersure why not
14:27.46[TK]D-FenderSkumling : CONGRATULATIONS
14:27.53shmaltztrixter, take a look:
14:27.55shmaltzhttp://lists.digium.com/pipermail/asterisk-users/2006-January/date.html
14:28.02shmaltz<PROTECTED>
14:28.04shmaltzArchived on: Sun Jan 15 05:17:07 CDT 2006
14:28.12trixterpeople just took off for the holiday here
14:28.19Skumling[TK]D-Fender: huh?
14:29.08[TK]D-FenderSkumling : For getting away from AMP
14:29.29[TK]D-FenderSkumling : What do you really need out of *?
14:29.33*** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com)
14:29.36Skumling[TK]D-Fender: hehe... thanks ;)
14:30.45fugitivoSkumling: congratulations too!
14:30.56Skumling[TK]D-Fender: I need a not-so-complicated system, but I need to run 4 separate (very small) companys on it, with different welcome etc, but I also need to be able to transfer calls between local phones, hold music, voicemailboxes etc.
14:31.15Skumlingvoicemail, hold music etc. should not be an issue
14:31.21SkumlingI also need a telemarketing torture script :-D
14:31.37[TK]D-FenderSkumling : All very easy....
14:32.22[TK]D-FenderSkumling : A days work tops.
14:32.25Skumling[TK]D-Fender: maybe ;)... I'm a asterisk newbie, and I've spend quite some hours reading docs, and I want to have a basically understanding of how things work
14:32.27*** join/#asterisk barinoff (i=izida@82.162.60.62)
14:32.42barinoffcan i ask dummy question?
14:32.55[TK]D-Fenderbarinoff : Hasn't stopped enough people yet... shoot!
14:32.58[TK]D-Fender;)
14:33.11Skumling[TK]D-Fender: my only problem right now, is that when I hit the flash-key to transfer a call, the "s" is executed immediately
14:33.26shmaltztrixter, nah, I don't think that in england they have a MLK holiday
14:33.32shmaltzunless it's a bank holiday :P
14:33.36Ahrimanesanyone tried 4 or 8 port ata's ?
14:33.46[TK]D-FenderSkumling : What interface?
14:33.49Skumling[TK]D-Fender: I would like some kind of prompt letting me enter a number etc.
14:34.03Skumling[TK]D-Fender: ISDN Gigaset handset on a HFC-C in NT-mode
14:34.05barinoff:) I first time install this soft and now have a big question - have it http or gui iterface for configure?
14:34.10shmaltzAhrimanes, yes
14:34.16shmaltzmediatrix
14:34.17Skumling[TK]D-Fender: using Bristuff
14:34.25[TK]D-FenderSkumling : Don't know ISDN.... wish I could help there.
14:34.27Ahrimanesscardinal: ok, know the pricing? did they work well?
14:34.44barinoffas i understand - it's workable with webmin?
14:35.08[TK]D-Fenderbarinoff : There are web-based GUI's for setting up * but they all create crappy dialplans and take away the control from you.
14:35.46[TK]D-Fenderbarinoff : No WebMin.  there is AMP, ScopServ, FirstLane, and a few otehrs, none of which I suggest unless you are looking at a large install.
14:36.03Skumling[TK]D-Fender: damnit... but can't I make a "s"-entry in my dialplan?
14:36.14barinoff[TK]D-Fender where i can read about it?
14:36.33_upsitehey guys
14:36.55shmaltzD-Fender, barintoff, it's thirdlane, not first lane
14:37.06shmaltzalso, thirdlane is a webmin module
14:37.40barinoffshmaltz - some link?
14:37.41*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
14:37.54[TK]D-FenderSkumling :Yes in the context that your phones use.  I'd suggest you not mix other kinds of phone into that context however
14:37.56shmaltzthirdlane.com
14:38.11[TK]D-Fendershmaltz : Shit by any other name :)
14:38.13_upsiteif i register my *  like " register =>  user:pass@some.dialup.host.dyndns.org"   an this host changes it's ip address ..* is not re-resolving the new ip address.  any fixes or workarounds for that?
14:38.14Skumling[TK]D-Fender: that's no problem
14:39.12[TK]D-FenderSkumling : You can get dial-tone, but the problem is you want to use this for TRANSFER, not to place another direct call...
14:39.25backblueanyone with DUNDi working?
14:39.45[TK]D-FenderSkumling : if all you wanted to do was use "flash" to place another call you'd use DISA in your "s" pointing to the same context.
14:40.10backblue_upsite: that its a problem, and there is no correction avaliable yet.
14:40.28[TK]D-FenderSkumling : so in [myphones] you'd have exten => s,1,DISA(myphones,nopassword)
14:40.34*** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net)
14:40.36*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
14:41.03*** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
14:41.40[TK]D-Fender(or whatever the nopassword parameter is)
14:41.56Skumlingphone, brb
14:42.11*** join/#asterisk SplasPood (i=jwb@206.252.198.100)
14:42.58*** join/#asterisk secure75 (n=mic@dslb-084-057-040-132.pools.arcor-ip.net)
14:43.41enemy^x[TK]D-Fender: http://pastebin.ca/36970 (could you please look at this, I`m not getting why my show hints doesnt work out :( )
14:44.02Skumling[TK]D-Fender: okay...
14:44.04shmaltzanybody from Digium is up already?
14:44.08shmaltzkevin?
14:45.01*** join/#asterisk rhousand (n=rhousand@rrcs-24-199-246-10.midsouth.biz.rr.com)
14:45.06[TK]D-Fenderenemy^x : Which one isn't working?
14:45.13[TK]D-Fender(specific #)
14:45.33pifwhat is the g711-alaw sample rate?
14:45.51enemy^x[TK]D-Fender, none of them... I`m working with my own, 236 for the moment
14:45.52[TK]D-Fenderpif : 8000
14:46.19[TK]D-Fenderenemy^x : pastebin your sip.conf
14:46.23trixtershmaltz: the list took a holiday, let it have its vacation
14:46.28piffunny, when I listen to the files from MusicOnHold they sound too fast
14:46.50shmaltztrixter, only if it shares the beer with me, and maybe dances with me
14:46.54*** part/#asterisk secure75 (n=mic@dslb-084-057-040-132.pools.arcor-ip.net)
14:47.28Skumling[TK]D-Fender: hrm, when placing the DISA-stuff into my dialplan, I get a very funny sounding and jittering dialtone
14:47.32trixterhey its free you cant expect it to actually work too!! (the open source mantra)
14:47.39*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
14:47.41shmaltzlol
14:47.48Skumling[TK]D-Fender: and when dialing a local number nothing happens :-/... hrm.
14:48.07[TK]D-FenderSkumling : Pastebin your extensions.conf
14:48.13enemy^x[TK]D-Fender: http://pastebin.ca/36971
14:48.14*** join/#asterisk MatsK (n=mk@3.80-203-81.nextgentel.com)
14:49.16shmaltzthis guy is an idiot:
14:49.17shmaltzhttp://video.google.com/videoplay?docid=-1532509206579317476&q=George+Galloway
14:49.30[TK]D-Fenderenemy^x : So if 236 is on the phone it won't show you the hint for it?
14:49.46enemy^x[TK]D-Fender: correct
14:49.49*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:49.50*** mode/#asterisk [+o anthm] by ChanServ
14:50.26Skumling[TK]D-Fender: http://pastebin.ca/36972
14:50.40[TK]D-Fenderenemy^x : paste the line that "show hints gives for it
14:51.30Skumling[TK]D-Fender: the dialplan is only for testing so far :)
14:51.52[TK]D-FenderSkumling : sunno....
14:52.00[TK]D-Fenderdunno..
14:52.10Skumling[TK]D-Fender: what I expect to be able to, is to hit the flash key, then forward a call to eg. local 12 or 15
14:52.38enemy^x[TK]D-Fender: http://pastebin.ca/36973
14:53.18*** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie)
14:54.01[TK]D-FenderSkumling : Could be becuase you have 2 entries for SIP/236 <-  You shouldn't double them up like that...
14:54.20*** join/#asterisk Davey|Work (n=davey@unaffiliated/davey)
14:54.27JohnnieAnyone handy with ztdummy installations?
14:54.31*** join/#asterisk mhnoyes (n=mhnoyes@user-38lc07j.dialup.mindspring.com)
14:54.32Davey|WorkIs it possible to have a remote SIP client with a dynamic IP?
14:54.41JohnnieI'm having a problem which I can't seem to nail down.
14:54.42[TK]D-FenderSkumling : Well DISA won't help you transfer like I said.  Try using the "#" method in your dial commands for it instead.
14:54.50[TK]D-FenderDavey|Work : Sure
14:55.15shmaltzfor some reason I can't stop laughing at this guy:
14:55.17shmaltzhttp://www.fool.com/News/mft/2006/mft06011321.htm
14:55.18shmaltzI know it sounds rude, but I can't help it
14:55.22Skumling[TK]D-Fender: hum okay. any hints for where to search for docs?
14:55.23Davey|Work[TK]D-Fender, the guy who configured this server, set defaultip to the guys dyndns address, does that sound sane?
14:55.50shmaltzFunny Quote of the Day - Lucille Ball - "The secret of staying young is to live honestly, eat slowly, and lie about your age."
14:56.28shmaltzDavey|Work, it should actualy work, there should be a patch for asterisk that should support dyndns
14:56.35[TK]D-FenderDavey|Work : The remote sip extension should have "host=dynamic".
14:56.51[TK]D-FenderDavey|Work : That will let him register from "wherever
14:57.20[TK]D-FenderDavey|Work : don't use defaultIP.
14:57.50Davey|Work[TK]D-Fender, I figured that was the case, as it never worked ;)
14:58.20Davey|Work[TK]D-Fender, so the phone knows where the asterisk server is and when its IP changes, it automatically tells Asterisk?
14:58.29JohnnieAnyone? :)
14:58.40Davey|Workor, you know, he can power-cycle the phone or something?
14:58.55shmaltzDictionary.com Word of the Day - capricious: whimsical; changeable.
14:58.56shmaltzhttp://dictionary.reference.com/wordoftheday/archive/2006/01/16.html
14:59.15*** join/#asterisk tengulre (n=tengulre@219.144.204.16)
14:59.38shmaltzwe gota put this one for asterisk as well:
14:59.39shmaltzQuote of the Day - Michael Jordan - "I can accept failure, but I can't accept not trying."
14:59.44[TK]D-FenderDavey|Work : It'll register wherever it is and thats that.
15:00.02Davey|Work[TK]D-Fender, OK, so I remove the defaultip, and set host=dynamic instead?
15:00.27[TK]D-FenderDavey|Work :: Typically if it looses contact it'll try to re-register at least a few times which will eventually succeed and update * as to its location
15:00.33[TK]D-FenderDavey|Work : Correct
15:00.54Davey|Work[TK]D-Fender, it already has host=dynamic, is the defaultip=hostname breaking it?
15:02.42barinoffi really cant uderstand how can i add account that sip software phone connect to asterisk? it write me that wrong username/password and in console sip show users is empty
15:02.51*** join/#asterisk BeHappy_ (n=willy@host54-203.pool877.interbusiness.it)
15:03.20*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
15:04.32[TK]D-FenderDavey|Work : no, just reload in CLI
15:04.42Davey|Worksee, I have no idea how to do that :)
15:05.11[TK]D-FenderDavey|Work : "asterisk -r" and then "reload"
15:05.16Skumlingasterisk -r should pickup the process
15:05.17Davey|Workthanks :)
15:05.26Skumling[TK]D-Fender: sorry ;)
15:05.52Davey|Workis there a way to do like apachectl configtest, to do a sanity check before I do that? :)
15:07.14[TK]D-FenderDavey|Work : Sanity is sold seperately, not part of the default install :)
15:07.32Davey|Workwait, we can buy sanity now? Man, nobody told me :/
15:07.33[TK]D-FenderDavey|Work : See "Terms & Conditions"
15:08.34backblueomg no one uses dundi, how the hell do you make a asterisk cluster without ser?
15:08.45*** join/#asterisk gvag11 (n=g@ppp71-adsl-133.ath.forthnet.gr)
15:08.50gvag11hi guys
15:09.31gvag11Looking for recommended TYAN mobos... Or any other good mobo ... any idea ????
15:09.50ManxPowerbackblue, You mean like as documented in the README files.
15:09.53ManxPowerand on the wiki
15:09.56ManxPowerand on the mailinglists
15:13.08Skumling[TK]D-Fender: seems like the DISA-stuff can do the trick...
15:14.18xhelioxCan't dial out from Teliax, anyone else experiencing this too?
15:14.18MrChimpyi got music on hold working! my life's mission is complete! now people can dial in and hear "your mother's got a penis" at will.
15:14.43[TK]D-FenderSkumling : But that doesn't have anything to do with "transfer" which is what you were actually hoping for.
15:15.14dippo_yes xheliox
15:15.15[TK]D-FenderSkumling : You should use the "#" transfer feature in your dial statements instead for those phones
15:15.16dippo_we are down entirely
15:15.24dippo_getting retries on outgoing stuff via the IAX trunk
15:15.30dippo_i filed a ticket, no response yet
15:15.34dippo_i have not been too pleased with teliax so far
15:15.47xhelioxThis is fairly unusual, in my experience.
15:16.03dippo_our service has been okay, but I get very little in the way of responsiveness from them in person
15:16.06*** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
15:16.11dippo_as far as answering billing/support questions
15:16.20xhelioxyeah, they have staffing issues
15:16.33Skumling[TK]D-Fender: okay, but how do I get to enter the tones without "s" being executed as soon as I hit flash?
15:17.30xhelioxTheir support number doesn't go through from my mobile.
15:17.43[TK]D-FenderSkumling : You don't USE FLASH.
15:17.52*** join/#asterisk Splas (i=jwb@206.252.198.100)
15:18.03[TK]D-Fenderuse DTMF "3" and add the "t" to your dial lines
15:18.27Skumling[TK]D-Fender: humm okay :-/...
15:18.30ManxPowerSkumling, There are several ways to do Transfers in Asterisk.
15:18.55ManxPowerThere is the evil stupid ugly hack of usint "t" or "T" on the Dial line to handle devices that are too brain dead to support their own transgers.
15:19.15SkumlingManxPower: okay... tell me all about it, I haven't been able to find much docs on transfers
15:19.15ManxPowerThere is the correct method of transfer for the device (FLASH for analog), Transfer button for SIP.
15:19.24[TK]D-FenderManxPower: he's running ISDN phones whose "flash" button starts a seperate call.  So I think that qualifies as "dumb"
15:19.29*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
15:19.36ManxPowerSkumling, Since very few people use BRI phones with Asterisk, you won't get a lit of useful help here.
15:19.55*** join/#asterisk pifiu (n=myassisb@208.205.181.170)
15:20.00ManxPower[TK]D-Fender, How do you know it's not just a config issue with the phone?
15:20.11*** join/#asterisk skamp|tee (i=skambar@p54841A45.dip0.t-ipconnect.de)
15:21.13SkumlingManxPower: okay... humm, but for wireless use, I don't think there's a nice solution without using DECT
15:21.28ManxPowerSkumling, your phone acts weird with you are trying to do a transfer.  It might be best if you talk with people that actually have an ISDN BRI phone.  Many of them hang out on #asterisk-drinkers.
15:22.20*** join/#asterisk dorphalsig (n=dorphals@200.106.223.5)
15:22.38SkumlingManxPower: I think the problem is, that the ISDN base station (where I hook up with BRI to the HFC-card) itself is a small PABX...
15:22.39dorphalsigHey! Anybody knows if I can use ASterisk real Time with * 1.0.10?
15:22.41[TK]D-FenderManxPower : I don't.
15:23.08ManxPowerdorphalsig, NO!  Realtime is only for 1.2
15:23.17SkumlingWhat should the flash-button do, when it acts properly?
15:23.49*** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net)
15:24.23ManxPowerSkumling, That depends on the device.  For most zap devices it sends a FLASH to Asterisk, for SIP devices it handles everything internally.
15:24.47ManxPowerSkumling, I assume you enabled threeway and transfer in zapata.conf?
15:24.52*** join/#asterisk secure75 (n=mic@dslb-084-057-039-021.pools.arcor-ip.net)
15:24.57SkumlingI have transfer=yes
15:25.16file[laptop]you need to enable threesomes er I mean threeway calling
15:25.31tdonahuegood morning all
15:26.01*** join/#asterisk _cleric_ (n=dacleric@p5482BFC6.dip0.t-ipconnect.de)
15:26.05Skumlingand if I define "s" to dial another local phone, then when I hit flash the other local phone rings, I pickup, have connection between the two local phones, and when I hangup, the call is routed between the external party and the "new" local phone
15:27.44*** join/#asterisk Bambr (n=Bambr@213-35-236-199-dsl.end.estpak.ee)
15:28.13Skumlinghumm wtf. now on a newer base, the flash key doesn't work at all...
15:28.26Skumling(I've got two different ISDN base stations for testing)
15:29.53ManxPowerSkumling, Well if you insist on not solving the underlying problem then use DISA.
15:29.55jbalcombI have a 'ast_expr2.fl: ast_yyerror():' in my log. Is there are way to know which file the syntax error is in?
15:30.30sivanais T.38 suppose to work with traditional analog fax machines, or completely different hardware?
15:30.33backbluecan i can test extensions? in asterisk console? like dial, just to know its working.
15:30.57SkumlingManxPower: I would love to solve the underlying problem, but I don't really know where to start :-/
15:31.10*** join/#asterisk ibob63 (n=hp@bb-87-82-25-51.ukonline.co.uk)
15:31.18ManxPowerSkumling, so use DISA
15:31.55*** join/#asterisk azzie (n=az@azzie.net)
15:32.22ManxPowerSkumling, a file said you also have to enable threeway
15:33.05ibob63Does asterisk have a billing module?  I am planning to install it in my studio which is shared by four friends. Could we then work out the bill for each phone?
15:33.24iCEBrkrdooo be doobe dooooooo
15:33.28*** join/#asterisk microcode (n=microcod@70.88.212.117)
15:33.47*** join/#asterisk tmccrary (n=tmccrary@68.78.185.254)
15:34.00*** part/#asterisk microcode (n=microcod@70.88.212.117)
15:34.03iCEBrkrjbalcomb: I'm thinking that error you're getting is in a GotoIf() statement
15:34.12tmccraryIs there something syntactically wrong with this line: exten => 2501,1,Dial(SIP/2501&SIP/2503,12,Ttr)
15:34.27*** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie)
15:34.36iCEBrkrtmccrary: Looks good..
15:34.47tmccrarybecause it only calls 2501, never 2053
15:34.51*** join/#asterisk stef2_ (n=stef@65.39.228.5)
15:34.52*** join/#asterisk microcode (n=microcod@70.88.212.117)
15:35.04tmccraryexten => _25XX,1,Dial(SIP/${EXTEN},12,Ttr)
15:35.07tmccraryI also have that
15:35.17tmccraryWould that cause a problem? I need both.
15:35.22*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
15:35.32Kattyuhm.
15:35.35Kattyhow do i save in vim?
15:35.39Katty:sq?
15:35.40ravenpiiCEBrkr: I *think* it's supposed to be a "|" between the two numbers, not a "&".  (I haven't tested it, and could be wrong.)
15:35.41tmccrary:wq
15:35.42Kattythanks.
15:35.43tzangerKatty: :wq
15:35.44fugitivo:x
15:35.50iCEBrkrravenpi: Nope.
15:35.51tzanger:w is write, :q is quite
15:35.53tzangerer quit
15:35.59tzangerif you just want to write use :w
15:36.00fugitivo:x = shortcut to :wq
15:36.01Kattyfugitivo: :<
15:36.03stef2_hello, i'm looking for version 1.3.4 of PWLIb and 1.9.4 of oh323 in order to compile the asterisk oh323 driver, do you have a mirror ?
15:36.06tmccraryI hate vim with a passion :)
15:36.11iCEBrkrtmccrary: That may cause a problem.
15:36.16microcodeAnyone familiar with Asterisk on Mac OS X?
15:36.23tmccraryiCEBrkr: what would?
15:36.27*** join/#asterisk klictel (n=klictel@207.107.208.137)
15:36.28tmccraryoh the pattern?
15:36.37drumkillamicrocode: what's your problem
15:36.41iCEBrkrtmccrary: _25XX + 2501
15:36.55tmccraryI have the 2501 listed first in the config.
15:37.12iCEBrkrtmccrary: I'm thinking it may 'match' 2501 first, but I'm not 100% sure.
15:37.15pifanyone using a cisco 7920 wi-phone?
15:37.27microcodedumkilla: thanks.  just curious on how media processing (tone detection and announcement streaming) is done on the Mac.  My understanind this that there is no zaptel lib, correct?
15:37.32tmccraryI wonder if there's anyway to force an exception like that
15:37.45tmccraryI have 2501 listed first in the config
15:37.50microcodedrumkilla: i am using a softphone to connect to Asterisk via SIP right now
15:37.53drumkillamicrocode: yeah, that has nothing to do with zaptel, actualy.
15:37.59microcodeoh
15:38.11iCEBrkrtmccrary: I'm thinking it'll match first since it's absolute.
15:38.13stef2_Anyone familiar with asterisk-oh323 ?
15:38.15iCEBrkrnot a regexp match
15:38.18tmccraryI wonder if I go 2501,1 then _25XX,2 ?
15:38.20microcodeso, media processing is in the core of Asterisk then?
15:38.23tmccraryno that wont work
15:38.37tdonahueif i have an analog fax machine attached to a channel bank, should I have faxdetect enabled or disabled?
15:38.44drumkillamicrocode: well, most of the time.  app_meetme, for example, uses zaptel to do conferencing
15:38.57iCEBrkrtmccrary: What the hell are you trying to do?
15:39.17tmccraryI have a set of extensions, 25XX, that I need to be able to call like normal.
15:39.30*** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk)
15:39.30tmccraryHowever, when someone calls 2501, I need it to ring two extensions.
15:39.38microcodedrumkilla:  thanks.  so, in the conf. call w/ zaptel, the zaptel lib provides access to a hardware implementation of the conf. media processing?
15:39.49*** join/#asterisk discofish (n=Rob@rrcs-24-97-85-209.nys.biz.rr.com)
15:39.51backbluedrumkilla: hi, can you give me a hand with dundi? i cant get help from anywhere!
15:39.52CurusSetCallerPres(prohib) works great on Zap
15:39.56iCEBrkrtmccrary: Why can't you make it 2601 or 1501 or something different?
15:40.13CurusWhen I try it on a different PBX connected via SIP, I have no luck with it
15:40.14tmccraryI'm thinking I'll have too, man asterisk really needs work still. So many hacks and work arounds.
15:40.31[TK]D-Fendertmccrary : You don't want to use 25XX.  That allows you to dial non-valid extensions.  You also DEFINATELY don't want to mix 25XX and 2501 together.
15:40.32tmccraryIt really shouldn't be 1.0 yet I don't think.
15:40.33drumkillamicrocode: well, it uses zaptel's timing to assist in the conferencing implementation.  there is no hardware conferencing available with zaptel at this point, but there could be in the future ...
15:40.36iCEBrkrtmccrary: Hacks and work arounds? Just don't do shit all jacked-up and you'll be fine.
15:40.38file[laptop]it's rather easy to do conferencing with zaptel... you create a pseudo channel for the conference, write in signed linear and read in signed linear (you get it mixed)
15:40.51*** part/#asterisk barinoff (i=izida@82.162.60.62)
15:41.05tmccraryWell, there should be a mechanism for calling two extensions at once, that's very common.
15:41.08stef2_Where can I download version 1.3.4 of PWLib
15:41.13CurusIsn't the Zap-connected asterisk supposed to see that the CLID presentation has been prohibited in the SIP call?
15:41.23microcodedrumkilla:  thanks for the help!
15:41.33drumkillamicrocode: no problem  :)
15:41.40iCEBrkrtmccrary: You people forget, just because it's a highly configurable PBX software that you think it can do anything you want.  Think about all the other PBX's out there that can't do what Asterisk does..
15:41.49tmccraryThank you for the help though iCE, I don't want to come off as rude.
15:42.03iCEBrkrNo, not rude.. Just nieve.
15:42.05drumkillabackblue: try this link, it's very helpful ... http://leifmadsen.com/papers/dundi-intro.pdf
15:42.06tmccraryI know, some of my code is in asterisk right now (albeit not a lot)
15:42.07iCEBrkr:P
15:42.45iCEBrkrThere are a lot of people who come in here with all these great expectations cuz Asterisk is so configurable and you can make it do jumping jacks.. but yet they really don't understand WTF they're trying to do.
15:42.52*** part/#asterisk stef2_ (n=stef@65.39.228.5)
15:42.55tmccraryHonestly, I think I'm going to write something to allow calling multiple extensions, I really would like it.
15:43.11iCEBrkr...and on top of that, it seems like people want to be all secretive about their what they're trying to do so we can't help them do it the right way
15:43.38iCEBrkrtmccrary: It dials multiple extensions the way you have it.. It's just you're whacked on your extension naming.
15:43.41backbluedrumkilla: dundi looks for one existing extension, or it can looks for one REGISTERED peer? i have a problem because i have 2 asterisks and mobile extensions, that have to register on both servers.
15:43.44jbalcombiCEBrkr: i see. its an unexpected TOK_EQ then says Input: = 0800. I did a grep on the configs. 71 lines with 0800.
15:44.00backbluedrumkilla: i'm trying to take a look on dundi if it solves my problem.
15:44.15jbalcombiCEBrkr: they all says "exten => s,6,GotoIf($[${dnd} = 0800]?10:7)"
15:44.35iCEBrkrtmccrary: and seriously-- depending on how many extensions you have, don't be lazy and use _25XX for matching.. Give each extension a 1 liner to a macro
15:44.43tmccraryYeah, but there should be a mechanism for dialing multiple extensions at once. Right now, I have to create all kinds of crazy extra extensions and generally hack around the config to get things to work. Just because its the way asterisk does it right now, doesn't mean it's the best way.
15:44.48drumkillabackblue: it looks for the existance of an extensions
15:44.51iCEBrkrjbalcomb: ${dnd} appears to be blank..
15:44.57[TK]D-Fendertmccrary : You CAN dial both at the same time, and your first pasted sample was FINE.  it was just CONFLICTING with your pattern match!
15:45.12iCEBrkrtmccrary: What [TK]D-Fender said.
15:45.13[TK]D-Fendertmccrary : Do NOT mix 2501 and 25XX!
15:45.27backbluedrumkilla: theres is nothing to lookup for registered peers on the time we try to make a call?
15:45.44iCEBrkrjbalcomb: exten => s,6,GotoIf(X$[${dnd} = X0800]?10:7)
15:45.49iCEBrkrjbalcomb: Try that instead
15:46.00drumkillabackblue: well, there is ... now bare with me here :)
15:46.14jbalcombiCEBrkr: this is the line before that "exten => s,4,DBget(dnd=dnd/SIP/63)"
15:46.28drumkillabackblue: are you using realtime?
15:46.30iCEBrkrjbalcomb: It's still possible that DND won't be set.
15:46.42dpryoThe ?: notation in asterisk, is the same as the one in c? ( <true/false> ? <if true> : <if false> )
15:46.55iCEBrkrjbalcomb: Just stuff an X before the variables you're comparing
15:47.01iCEBrkrjbalcomb: like I pasted
15:47.15jbalcombiCEBrkr: will do. whats the big x do?
15:47.37iCEBrkrjbalcomb: Nothing.. It's just a place holder to fill in any NULL or nothing values
15:47.42*** join/#asterisk LoRez (i=lorez@freenode/staff/lorez)
15:47.58[TK]D-Fenderjbalcomb : You are using < 1.2 commands and trying to use 3 dimensions in DB1 !  "dnd/SIP/63" is NOT valid.
15:48.06iCEBrkrjbalcomb: It's just a warning you're getting anyhow, so it's not really hurting anything
15:48.16[TK]D-Fenderjbalcomb : DB1 is FAMILY/KEY.
15:48.36tmccraryWouldn't it make more sense for asterisk to process non matching rules first and THEN match any regex'd type rules?
15:48.42tmccraryI'm going to write a patch.
15:48.54tmccraryor at least have an option to do so.
15:49.03iCEBrkrtmccrary: See, that's your problem right there, you're hell-bent on doing it the fucked up way.
15:49.20tmccrarywtf are you blathering about
15:49.33wunderkinanyone else getting mailing list posts?
15:49.48anderivwunderkin: nope
15:49.52tmccraryI would prefer to have similar extensions for everyone in the same area, that's just common sense.
15:49.56iCEBrkrI'm blathering about you breaking Asterisk :P
15:50.16*** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com)
15:50.17tmccraryHow exactly would following common sense break asterisk?
15:50.20backbluedrumkilla: i'm not, but i can use it, and i pretend to, my problem its nat issues. i dont know how is realtime with nat.
15:50.21iCEBrkrWhy would you make a series of extensions for people and then have a GROUP in the same series?
15:50.34tmccraryExceptions are handy.
15:50.58drumkillabackblue: nat would have no effect on using realtime.  it's just storing the config in a database as opposed to text files.
15:51.07jbalcomb[TK]D-Fender: ok, i don't get the /programming/ part of Asterisk yet but this is a setup I am taking over from people who no longer exist. :/ Thanks.
15:51.11iCEBrkrJoe, Jim, Bob, and Mike have 2501 to 2504.  Why would you make 2505 ring a group of phones when 250X are designated as individual's extensions.
15:51.35backbluedrumkilla: hoo, ok i can use database storage, no problem with that, but how that solves my problem?
15:51.36iCEBrkr[TK]D-Fender: Yea, Jim was shoved into this Asterisk project without any notes :)
15:51.39tmccraryLets say, sally who is 2505, needs to be reachable in two locations.
15:51.56jbalcombiCEBrkr: thanks man. im trying to clean things up as I go about learning this setup.
15:51.57tmccraryBut Sally SHOULDNT need two different extensions.
15:52.01iCEBrkrtmccrary: That I understand.
15:52.02drumkillabackblue: well, it should work with "rtupdate" turned on in sip.conf
15:52.06drumkillabackblue: another option is this ...
15:52.14[TK]D-Fendertmccrary : Pastebin your whole dial-plan and I'll see what I can do.
15:52.18[TK]D-Fender~pb
15:52.21jbotsomebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
15:52.28tmccraryokay one sec, thank you.
15:52.31drumkillabackblue: you can use DUNDi in combination with using the regexten feature in sip.conf.  That will, the extension will only exist when the peer is registered
15:52.39iCEBrkrtmccrary: So is there a problem with creating 2501, 2502, 2503 extensions individually??
15:52.59*** join/#asterisk blitz[class] (n=lmadsen@199.227.185.35)
15:53.08tmccrarynope
15:53.21tmccraryreally, I just want to be able to create an exception within a matching rule.
15:53.27iCEBrkrI mean if you have 100 extensions (2500-2599) doing that could be a pain in the ass..
15:53.47iCEBrkrBut if you only have 10 (2500-2509) then, what's the issue?
15:54.17tmccraryI don't see why that would cause a problem?.
15:54.24iCEBrkrjbalcomb: what version of asterisk they got over there?
15:54.34CurusHmm actually SetCallerPres(prohib) deletes all caller id information and replaces it with unknown
15:54.40iCEBrkrtmccrary: The problem is having to type in all that shit :)
15:54.47CurusThat's entirely useless, we have to send it for emergency services
15:55.00tdonahueif i have an analog fax machine attached to a channel bank, should I have faxdetect enabled or disabled?
15:55.12iCEBrkrtmccrary: and your 'exception' / 'patch' would just confuse people more than they already are..
15:55.17Curus(For SIP, that is, for Zap it works perfectly)
15:55.39tmccrarybut its just for special cases, so lets say most phone/calls just use the pattern match rule. However, when you want to do special things within that range of extensions, you can create a rule that is checked first.
15:55.59tmccraryso like 2-3 phones have special handling for calls (for multiple recipients, etc)
15:56.10iCEBrkrtmccrary: and your 'exception' / 'patch' would just confuse people more than they already are..
15:56.13iCEBrkr^^^^
15:56.20*** join/#asterisk MommomeryCliff (n=willy@host230-24.pool873.interbusiness.it)
15:56.23[TK]D-Fendertmccrary : Pastebin it....
15:56.25tmccraryI'm sure you could say that about anything that makes something better.
15:56.42backbluedrumkilla: can we talk better in private?
15:56.48tmccraryDon't fear change my friend. :)
15:56.55ManxPowerIf they are in the same context, then the closest matching exten => line will be the one that's used.
15:57.17tmccraryBut that's not the case Manx.
15:57.18iCEBrkrtmccrary: I fear confusion..
15:57.22ManxPowerextensions included with include => are always considered "least specific"
15:57.36dippo_xheliox: you heard anything from teliax yet?
15:57.38tmccraryI have one rule that is 2501 and it gets ignored for a _25XX rule.
15:57.48[TK]D-Fendertmccrary : How many extensions do you have?
15:57.50ManxPowertmccrary, in the same context?
15:57.53tmccrary2501 is much more explicit than _25XX
15:57.54tmccraryyes
15:58.00iCEBrkrtmccrary: and if you hang out in here long enough, you'll find out that people come up with the craziest shit and they're severly confused on how things work in extensions.conf
15:58.02jbalcombiCEBrkr: Asterisk 1.2.1
15:58.07ManxPowertmccrary, is 2501 before the pattern in extensions.conf?
15:58.11iCEBrkrjbalcomb: Ok yea, you're gonna have to update the DBGet() stuff.
15:58.15iCEBrkrjbalcomb: It's completely changed.
15:58.16tmccraryManx: Yes.
15:58.25iCEBrkrjbalcomb: Checkout the Wiki for the new syntax
15:58.30ManxPowertmccrary, I don't know what the problem is, since *I* use this feature all the time.
15:58.33[TK]D-FenderiCEBrkr : I've said as much including the error in formatting
15:58.51iCEBrkrhrrm?
15:58.54ManxPowerFor example I have an exten => _XXXX,1,Macro(disconnected) at the bottom of my extensions.conf
15:58.56*** join/#asterisk Run (n=BIGGIRL^@85.108.145.224)
15:59.18jbalcombiCEBrkr: sounds like fun. immature software mixed with lazy ass admins is good for creating new nueral patterns.
15:59.28ManxPowertmccrary, what happens when you comment out the  _25XX lines?
15:59.38tmccrarylet me paste the config
15:59.47iCEBrkrjbalcomb: naaa, I dunno about that.. 1.0.x code is significantly differen than 1.2.x
15:59.49ManxPowertmccrary, what happens when you comment out the  _25XX lines?
16:00.00*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
16:00.13iCEBrkrjbalcomb: ...and it's all for the better.. You'll figure it out, Just dig through the Wiki for the DB() functions.
16:00.14tmccraryhttp://pastebin.com/508268
16:00.22iCEBrkrI gotta head to lunch to beat the parade.
16:00.22*** join/#asterisk flujan (n=flujan@internet.nube.com.br)
16:00.32jbalcombiCEBrkr: voip-info.org wiki?
16:00.38[TK]D-Fendertmccrary ..... is that an AMP created context?
16:00.39iCEBrkrjbalcomb: yeah
16:00.49jbalcombiCEBrkr: thanks. good on ya mate.
16:00.50iCEBrkrjbalcomb: http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands
16:00.56iCEBrkrSpecifically
16:01.02tmccraryNo
16:01.11backbluedrumkilla: that really solves my problems, nice, but i still have one problem that its mailbox, where its mailbox located in the end?
16:01.19iCEBrkrok, I'm really leaving now :)
16:01.21[TK]D-Fendertmccrary : Hom many extesions do you have, and no-one has voicemail?
16:01.27*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
16:01.29jbalcombiCEBrkr ciao.
16:01.51tmccraryNo, I disabled voicemail for the time being.
16:02.09*** part/#asterisk ibob63 (n=hp@bb-87-82-25-51.ukonline.co.uk)
16:02.14[TK]D-Fenderand the # of extensions?
16:02.21flujanhi guys... We are setting up a test environment to work with asterisk and hard voip phones... So asterisk will work as a gateway between the phones. Actually we have 150 internal analog phone points. We expect to change this all to voip.
16:02.30tmccrary3
16:02.34flujanI know its also possible to work with adapters...
16:02.39ManxPowertmccrary, Do you REALLY want callers to be able to transfer themselves?
16:02.41flujanwhich is the best solution?
16:02.42[TK]D-Fendertmccrary : Not worth and pattern match.
16:02.59tmccraryI don't want to hack it up and being adding every extension manually.
16:03.01flujanand which is the cheaper solution?
16:03.16ManxPowertmccrary, nevermind, aswer my other question first.
16:03.31tmccraryI'd rather do this the RIGHT way, which apparently is impossible without the ability to add a rule that overrides the match (i.e. 2501,1)
16:03.47ManxPowertmccrary, people do it all the time.
16:03.54[TK]D-Fendertmccrary : You have 3... if you jumped to 10 its 7 more lines... whats the big deal?  its a cut & paste job.  Pastebin you entire extensions.conf and sip.con and I'll work a small miracle for you.
16:04.17tmccraryAgain, I'd rather have it be done the right way. I'm sure I could come up with all kinds of crazy work arounds.
16:04.31*** join/#asterisk m160858 (n=ubuntu@200.89.12.46)
16:04.51tmccraryI don't want to add every extension manually, whats the point of having a matching/regex feature in asterisk if I have to add them manually.
16:04.54dippo_xheliox: we appear to be back up on teliax fwiw
16:05.02m160858hi everyone
16:05.03tmccrarySounds like something is broken to me....
16:05.04ManxPowertmccrary, I see you are not listening.
16:05.13m160858i have problems with my asterisk
16:05.19tmccrarySorry manx, I am talking to Fender.
16:05.20[TK]D-Fendertmccrary : There is no way to prove the extensions validity!  So forget pattern match!
16:05.21m160858y try to install asterisk at home
16:05.22tmccraryWhat was your other question?
16:05.29m160858but i can't turn on
16:05.41ManxPowerFor the THIRD time, what happens when you comment out the  _25XX line and issue a "reload" on the asterisk CLI?  I'm not asking again.
16:05.58[TK]D-Fenderm160858 : For A@H plase go to #amportal
16:05.58tmccraryChances are, that will work.
16:06.01SDGLFirst off, I appologize for the lengthy question...  here it goes.  I've been investigating a random deadlock within Asterisk for about a month, and I seem to have it narrowed it down to the call parking feature.  We park about 50 calls per day, and the feature seems to be working pretty good, except that once in a while, while parking a call, the Asterisk box completly deadlocks, and a killall -9 asterisk is the only thing that will bring it back to life.  It
16:06.06ManxPowertmccrary, Try it.
16:06.08tmccraryLet me try it for you and reiterate.
16:06.09*** join/#asterisk daCount (n=cosg@19-103.241.81.adsl.skynet.be)
16:06.14m160858apears like it doesn't exists
16:06.34m160858oh
16:06.36m160858thanks
16:07.16[TK]D-Fendertmccrary : Add an "i" exten in your phones context and check if the EXTEN matches 25XX and playback "thanks for trying to dial something that LOOKS like an extension but isn't!"
16:07.20flujanhi guys... We are setting up a test environment to work with asterisk and hard voip phones... So asterisk will work as a gateway between the phones. Actually we have 150 internal analog phone points. We expect to change this all to voip.
16:07.42flujanI want some recomendation to work with hard phone... a specific product
16:07.42ManxPowerflujan, Great!
16:07.50ManxPowerI recommend Polycom
16:08.09[TK]D-Fenderflujan : Best solution change them all to PoE voip hardphones on a dedicated LAN.
16:08.11*** join/#asterisk SDGL (n=sdgl@64.5.206.131)
16:08.24SDGLSorry... lost my connection for a minute
16:08.27daCountany idea about a soft phone wich works with openbsd without too much hassle?
16:08.29[TK]D-Fenderflujan : I will second ManxPower's suggestion for Polycom.
16:08.42ReXwhat are the libraries necessary to compile chan_btp plz?
16:09.14ManxPowerReX, almost nobody uses that channel since it's still alpha quality.  The README did not give any clues?
16:09.28flujanok
16:09.41ReXno ManxPower
16:10.01ManxPowerReX, how about the mailing list archives?
16:10.10mutpaycheck taxes are teh suck
16:10.11flujanManxPower: which is the best sollution? Work with adapter in the analog phones or change all to the voip hard phones.
16:10.24ManxPowerflujan, that is ip to you.
16:10.44ReXI did not find anything
16:10.53ManxPowerflujan, you should build a prototype system first
16:11.25ManxPowerResults 1 - 10 of about 302 from lists.digium.com for  bluetooth. (0.32 seconds)
16:11.40trixterI like to use a rapid prototyping system when making prototypes, that way I dont have to wait for someone to mail me a case instead I can have a custom case made right on the sopt :P
16:12.21[TK]D-Fenderflujan : Like I said, change them all for VoIP hard-phones (suggest Polycom) using PoE on a dedicated LAN.
16:12.42ManxPower[TK]D-Fender, seems like overkill to me.
16:12.51*** part/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk)
16:14.30*** join/#asterisk GillesR_IMG-IT (n=GillesR_@64.5.206.131)
16:14.56[TK]D-FenderManxPower : He asked what the BEST solution was.  Not the CHEAPEST.
16:15.03flujanManxPower: yes, I will bought one... :D and [TK]D-Fender what to you mean by PoE?
16:15.48jbalcombPoE is awesomeness. Power over Ethernet, no AC adapter
16:15.49flujan[TK]D-Fender: Actually we work with fix ip address in the computers
16:15.50[TK]D-Fenderflujan : Power Over Ethernet.  This is a standard that allows you to power your phones off of a special switch instead of having to plug them into a power bar at each desk.
16:15.57*** join/#asterisk razu (n=razu@80-235-90-19-dsl.prn.estpak.ee)
16:16.36[TK]D-Fendertmccrary : Offer still open for me to re-vamp your setup....
16:16.44jbalcomb[TK]D-Fender I heard the polycom phones dont do DTMF via SIP INFO, true?
16:16.54flujan[TK]D-Fender: with hard voip phones we will need to create proportional IP address. is it right?
16:17.26[TK]D-Fenderjbalcomb : Not sure, but I know they do RFC2833 which is preferred.
16:17.50[TK]D-Fenderflujan : They can use DHCP or static IP.  I suggest using DHCP for everything personally.
16:18.45Luke-JrAny idea on recovering a PAP2-NA? ethernet/blue & power/red lights are lit solid
16:19.08jbalcomb[TK]D-Fender gotcha. we seem to be using 'via SIP INFO' but I don't know if there is a real reason.
16:19.14flujanAnd about adapters to analog phones... I also need suggestions since I must create to proposal to my boss. :P
16:20.00[TK]D-Fenderjbalcomb : programmer error <- :d
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16:20.37[TK]D-Fenderflujan : Another which might be easier for you would be to buy Norstar Digital sets and use them on Citel SIP gateways.
16:20.51[TK]D-Fenderflujan : Check out http://www.atacomm.com
16:20.53flujanwell, In order to build a prototype env two hard ip phones and a machine is enought, isn't it?
16:21.08[TK]D-Fenderflujan : For just a test of what * can do, sure.
16:21.11jbalcomb[TK]D-Fender haha.. i dont doubt it. i like the most that know one here has any idea why we are doing anything the way we are doing it
16:21.20jbalcombs/know/no/
16:21.31backbluewhen i make reload somemodule.so it does not shows "parsing ... blah" why?
16:21.53jbalcombjbot please dont do that anymore. its annoying.
16:22.49[TK]D-Fenderjbalcomb : Just don't add the trailing "/" then
16:23.05[TK]D-Fenders/Just/Ignored !
16:23.07JohnnieAnyone familiar with ztdummy and strange compiling issues?
16:23.10[TK]D-Fendersee?
16:23.19[TK]D-Fenderjbalcomb : Just don't add the trailing "/" then
16:23.23[TK]D-Fenders/Just/Ignored/
16:23.25flujan[TK]D-Fender: thanks... I will buy this and set up a prototype env.
16:23.28[TK]D-Fenderthere you have it
16:23.37flujanManxPower: thanks for the help.
16:23.42[TK]D-Fenderflujan : What are you looking to buy?
16:24.08jbalcomb[TK]D-Fender gotcha
16:24.15flujan[TK]D-Fender: two voip phones and two adapters to test with our analog system.
16:24.19dippo_is iax2 trunking encrypted at all? if one could tcpdump an iax2 stream could you decode audio packets from it?
16:24.26jbalcombs/gotcha/jbot is a tool of Satan/
16:24.35flujan[TK]D-Fender: first, we will set up a env to work with the internal phones
16:24.59*** join/#asterisk dily (n=dily@ip-85-108.sn2.eutelia.it)
16:25.03[TK]D-Fenderflujan : I typically don't suggest analog, but it is cheaper.  For your test I suggest 1 x SPA-2002, and 2 Higher-end Polycom phones (501 or 601)
16:25.08dilyhi@all
16:25.15flujan[TK]D-Fender: but we have 3 E1 and later we will adapt it to do calls outside the internal phone system...
16:25.45[TK]D-Fenderflujan : You are using channel-banks on your E1's internally?
16:25.51*** join/#asterisk fulgas (n=fulgas@209.8.233.229)
16:26.48jbalcomb[TK]D-Fender if im reading this righ, DBGet(dnd=dnd/SIP/63) is depricated for Set(dnd=${dnd(SIP/63)})?
16:28.50Davey|Workwhich ports do you typically need to open for SIP?
16:29.42flujan[TK]D-Fender: hum... I don't know... There is a company wich take care of our telepnony... We will make it internal now.
16:29.56*** join/#asterisk dily_ (n=dily@ip-85-108.sn2.eutelia.it)
16:29.57brookshiredavey: lots of them :)
16:29.57[TK]D-Fenderjbalcomb : Very wrong.
16:29.58flujan[TK]D-Fender: is there a way to me discover that?
16:30.10tdonahueif i have an analog fax machine attached to a channel bank, should I have faxdetect enabled or disabled?
16:30.33*** join/#asterisk denon (i=denon@synapse.subneural.net)
16:30.34*** mode/#asterisk [+o denon] by ChanServ
16:30.44flujan[TK]D-Fender: we can receive external calls with private real numbers in the company.
16:30.46dily_i recevice an error when i hangup an incomingcall,  -- Incoming call: Got SIP response 503 "Server error" back from xxx.xxx.xxx.xxx.   Any ideas?
16:30.50[TK]D-Fenderjbalcomb dnd/SIP/62 does not look right unless the last "/" gets included in the KEY.  Which looks confusing at best
16:31.03ravenpiCouple Polycom questions: 1) Any idea how to disable that occasional MWI "warble"?  [Not the stutter -- I got rid of that.]  2) How about re-programming the keys?  Found the place in the manual, but it's not very descriptive -- a sample of the <keys> section would help (want to have it dial the paging extension).
16:32.25flujan[TK]D-Fender: for instance, if someone wants to speak with me internally they just type 502
16:32.45flujan[TK]D-Fender: to receive external calls... they use a real telephone number
16:33.27backblueDavey|Work: you can say that in your conf files.
16:33.49backblueyou just specify a range and add the range to your firewall rules
16:34.30[TK]D-Fenderravenpi : Which "keys" are you looking to reprogram?
16:35.01ravenpi[TK]D-Fender: Preferably, the "Services" key (#31 on the 501, according to the docs).
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16:35.18*** part/#asterisk SDGL_ (n=SDGL@64.5.206.131)
16:35.19Davey|Workbackblue, Ok, thanks
16:35.24*** join/#asterisk SDGL_ (n=SDGL@64.5.206.131)
16:35.26Davey|Workbackblue, which config? :)
16:35.30[TK]D-Fenderflujan : Ok I'm not sure what kind of equipement you have now.  Your wording is confusing where you are mixing up equipment used for your internal phone and technology used for your outside phone link
16:35.37ravenpi[TK]D-Fender: Though I'll use the bottom-most soft key if I need to...
16:36.17*** part/#asterisk GillesR_IMG-IT (n=GillesR_@64.5.206.131)
16:36.21*** part/#asterisk SDGL_ (n=SDGL@64.5.206.131)
16:36.51[TK]D-Fenderravenpi : Not sure how you would go about changing that key....  And what do you mean the "bottommost soft-key"?
16:36.51Mimmusdoes anyone use Digium's Asterisk Enterprise Edition in the world?
16:37.19*** join/#asterisk GillesR_IMG-IT (n=GillesR_@64.5.206.131)
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16:37.42backblueDavey|Work: rtp.conf, we need on ip_conntrack_rtp :P
16:37.48backblues/on/one
16:38.18*** part/#asterisk tmccrary (n=tmccrary@68.78.185.254)
16:39.01ravenpi[TK]D-Fender: *hmmm*  Oh, well.  The three keys on the left (two of which I have assigned to extensions) -- I'll make the third one dial away.
16:40.03[TK]D-Fenderravenpi : If you don't use the "messages" button to pick up voicemail you could use that....
16:40.27ravenpi[TK]D-Fender: we do, alas.  But thanks...
16:40.30*** join/#asterisk ping1 (n=DLBaker@67-133-167-72.dia.cust.qwest.net)
16:41.16[TK]D-Fenderravenpi : oh well.. Acutally what DOES "services" do on an IP50x?  Its not supposed to have the microbrowser....
16:41.43*** part/#asterisk SDGL (n=SDGL@64.5.206.131)
16:42.05ravenpi[TK]D-Fender: darned if I know.  Pushing it does zilch -- which is why, combined with its ambiguous name, I thought it was an ideal candidate for re-programming.
16:42.18*** join/#asterisk Skkip (n=Skipper@216.160.91.91)
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16:42.28[TK]D-Fenderravenpi : Could be.  They may also be planning MB support... who knows.
16:42.47dily_anyone know how to resolve this error -- Incoming call: Got SIP response 503 "Server error" back from xxx.xxx.xxx.xxx. ? plz
16:43.05flujan[TK]D-Fender: we use http://www.zox.com.br/tz20.htm
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16:43.14ravenpi[TK]D-Fender: Thanks for the help.  So: any idea on the MWI "warble"?  I know *some* user's gonna want it disabled...
16:43.21flujan[TK]D-Fender: I don't know how to call this in english... :D
16:43.47[TK]D-Fenderravenpi : I looked it up, but don't see anything for the warble...
16:44.08festr_hello, i've problem with ringing tone between transfering two fxs. Scenario-> IAX -> asterisk FXS1 (ringing) press # and transfer to FXS2 (ringing, but this ring is not heared in IAX channel), what can be wrong? is it bug?
16:44.34ravenpi[TK]D-Fender: Yeah, I've been Googling for a while...  Oh, well: when I only have two things to whine about after a rollout, that's a pretty good day.
16:44.37[TK]D-Fenderflujan : You already own a lot of those?
16:44.46[TK]D-Fenderravenpi : hold on...
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16:46.37backblueDavey|Work: well you have a patch for iptables if is your case.
16:46.38[TK]D-Fenderravenpi : I think you can kill it in sip.cfg under the <ALERTING> section by setting it to "silence"
16:47.02[TK]D-Fenderravenpi : Also under "miscellaneous"
16:47.32ravenpi[TK]D-Fender: AWESOME.  I owe you a <beverage of choice>.
16:47.42[TK]D-Fenderravenpi : no biggie
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17:01.26flujan[TK]D-Fender: sorry, I was researching about our telephony system. :)
17:01.51flujan[TK]D-Fender: we have a dialogic d/600 jct - 2 e1 card
17:02.01flujan[TK]D-Fender: and a d/300 jct
17:02.20flujan[TK]D-Fender: both from dialogic
17:02.45*** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV)
17:04.02[TK]D-Fenderflujan : Well You can probably forget about that card in your new setup.
17:04.54*** part/#asterisk rstandy (n=rastandy@d83-176-116-85.cust.tele2.it)
17:04.59*** join/#asterisk santiago (n=santiago@208.195.215.222)
17:05.14flujan[TK]D-Fender: which card do you recommend in the new setup? First to a prototype env.
17:05.37*** join/#asterisk EriSan (n=erisan@81-174-42-154.f5.ngi.it)
17:05.42nextimeanyone using ooh323c?
17:05.44flujan[TK]D-Fender: and later on to replace them. :D
17:06.08[TK]D-Fenderflujan : Ok what do you use for phones right now, and what do you use for lines?  Can you run new Cat5 for VoIP hard phones?
17:06.28*** join/#asterisk jpablo (n=jpablo@200.94.130.194)
17:06.51flujanwhat to you mean by Cat5
17:06.52flujan?
17:06.59docelm0SIGH ETHERNET!
17:07.06docelm010-100-1000 BASE T
17:08.00[TK]D-Fenderflujan : And how many total phones / lines?
17:08.13jarrodanyone familiar with using cisco as5400/7x00 as media gateways?
17:08.28jarrodhow do I take an incoming call on a PRI and turn around and forward out another channel?
17:08.40flujan[TK]D-Fender: 90 phone lines... :)
17:09.04flujan[TK]D-Fender: we use headsets like that I showed you.
17:09.19[TK]D-Fenderflujan : Ok, that'd be the 3 E! you were mentioning earlier?
17:09.45[TK]D-Fenderflujan : Are the headsets attached to another phone or are they actually full phones themselves?
17:10.15jpablohey people, I'm getting this error when trying to ztcfg -v a dual e1 card:
17:10.17jpabloSPAN 1: CAS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
17:10.18jpablo31 channels configured.
17:10.18jpabloZT_CHANCONFIG failed on channel 1: Invalid argument (22)
17:10.37jarrodcheck /etc/zaptel.conf
17:11.03jpabloi got:
17:11.06jpablospan=1,1,0,cas,hdb3
17:11.07jpablocas=1-15:1101
17:11.07jpablodchan=16
17:11.07jpablocas=17-31:1101
17:11.09[TK]D-Fenderjpablo : Maybe youshould paste the line it says is bad. <-
17:12.16jpablo[TK]D-Fender, excuseme ?
17:12.17*** join/#asterisk Someone123 (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net)
17:12.31flujan[TK]D-Fender:  the can work as both. Depends on the software configuration... http://www.zox.com.br/download/tz-20-manual.doc
17:12.44*** join/#asterisk UlbabraB_ (n=salama@host241-43.pool8172.interbusiness.it)
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17:13.01flujan[TK]D-Fender: in this link You have full specification about our produtcts...
17:13.09PoWeRKiLLhi
17:14.12*** part/#asterisk darkskiez (n=darkskie@194.247.78.146)
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17:15.04flujan[TK]D-Fender: I found a english version http://www.zox.com.br/ingles/download/tz-20-manual.doc
17:15.25PoWeRKiLLsomeone have a the Asterisk Sound List in french ?
17:15.52*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)
17:15.56[TK]D-Fenderjpablo : I don't see "cas" as a valid parameter anywhere....
17:16.07jbalcomb[TK]D-Fender this DB1 is tad confusing. the entry i have is "/SIP/Registry/6310.0.101.181:5062:3600:63:sip:63@10.0.101.181:5062
17:16.47[TK]D-Fenderjbalcomb : Thats created automatically by * to store SIP registration info.
17:16.49*** join/#asterisk MommomeryCliff (n=willy@host230-24.pool873.interbusiness.it)
17:16.55jbalcomb[TK]D-Fender so i dont get how that correlates to a check on dnd/SIP/63 matching 0800
17:17.12[TK]D-Fenderjbalcomb : it DOESN'T have anything to do with your DND.
17:17.19jpablo[TK]D-Fender, you really don't know about configuring e1s, do you ... ? how else do i configure a cas group ?
17:17.48[TK]D-Fenderjpablo : Check the WIKI and the sample file.  I don't know them by heart for that tech... only N/A PRI
17:17.52*** join/#asterisk roulduke_ (i=bhcj49ft@p508D28AE.dip0.t-ipconnect.de)
17:18.10[TK]D-Fenderjbalcomb : Pastebin your extensions.conf and I'll take a lok
17:18.39jbalcomb[TK]D-Fender hrmm.. ok, so dnd/SIP/63 doesnt have anything to do with /SIP/Registry/63
17:18.56[TK]D-Fenderflujan : Of how many of those do you have already?
17:19.05[TK]D-Fenderjbalcomb : correct.
17:19.32IMG-GRHi guys, I keep getting "/usr/bin/ld: cannot find -ltonezone; collect2: ld returned 1 exit status; make[1]: *** [chan_zap.so] Error 1; make[1]: Leaving directory `/usr/local/src/libpri-1.2/channels'; make: *** [subdirs] Error 1" error while compiling Zaptel.  Any idea, since I've done all the required step to compile it?
17:20.04jbalcomb[TK]D-Fender is there anyway to think of family, keytree, and key, as database, table, and values?
17:20.34flujan[TK]D-Fender: 150 :D
17:20.51[TK]D-Fenderjbalcomb : not like that.  DB1 is only family/key.  not 3 dimensionsal. I mentioned this earlier.  Pastebin your entire extensions.conf and I'll see if I can fix it up for you.
17:21.21jbalcomb[TK]D-Fender i dont know pastebin. :/ one sec.
17:22.00[TK]D-Fenderflujan : ok, then my suggestion is to get AudioCodes 24 FXS SIP gateways and keep your current headsets.  Much cheaper since you already have the phones (which is what those usints are and I'm sure they weren't too cheap either...)
17:22.00ManxPower~pb
17:22.03jbotfrom memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
17:22.04[TK]D-Fender~b
17:22.05jbotpicobot: c
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17:22.30Kato41hi
17:22.36*** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net)
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17:23.34flujanok thanks
17:23.41jbalcomb[TK]D-Fender our extensions.conf has five includes for 'ease of some crap' ill paste bin the include where the trouble is.
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17:24.21Kato41got a problem in running two hfc-based cards in a asterisk@home setup
17:24.30[TK]D-Fenderjbalcomb : ok, pastebin first.  If it makes me cringe I might ask you to e-mail the whole thing
17:24.43*** part/#asterisk cfh (n=luca@82.193.23.6)
17:26.21[TK]D-Fenderflujan : a full solution including 6 AudioCodes gateways and a Sangoma A104d card for your E1's would be about $17,800 USD
17:26.32[TK]D-Fenderflujan : And not require massive rewiring.
17:26.49jbalcomb[TK]D-Fender well, i gaurantee you're gonna cringes atleast from lack of macros
17:27.37[TK]D-Fenderjbalcomb : We'll see how generous I feel on seeing it :)
17:28.02ManxPowerYet another thing in New Orleans that isn't working.  A friend's mother died and the cemetary where their family tomb is located is not open.
17:28.41Kato41which OS/asterisk combination would be the best to run asterisk with two hfc isdn cards in euroisdn?
17:29.02ManxPowerKato41, Yes.
17:29.55[TK]D-FenderManxPower : 42?
17:31.16*** join/#asterisk ToTo (n=ToTo@host16-146.pool872.interbusiness.it)
17:31.35jbalcomb[TK]D-Fender http://pastebin.com/508395
17:31.43*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfk1f.dialup.mindspring.com)
17:31.57*** join/#asterisk juice (n=juice@209.33.105.105)
17:32.38Mimmuscan any good will man/woman give a look at this 'pri debug' output http://pastebin.com/508398 ?
17:32.39[TK]D-Fenderjbalcomb : 1 phone/queue?
17:33.01manydoes anybody know how to read Hz value in linux systems?
17:33.56*** join/#asterisk wpayne (n=wpayne@vgateway.libertyrms.info)
17:34.36wpayneHello everyone
17:34.54Mimmusa call starting from a legacy Alcatel PBX connected to * is dropped with NOANSWER
17:35.03jbalcomb[TK]D-Fender i dont think so. as i understand it we have a host of 800's that go to a group of reps who answer based on callerid
17:35.35*** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar)
17:35.35[TK]D-Fenderjbalcomb : Just trying to figure out who the DND applies to.... where does it get set?  What does it really mean?
17:37.59*** join/#asterisk greendisease (n=jack@fedora/greendisease)
17:38.31jbalcomb[TK]D-Fender another include, applications.conf, has this: http://pastebin.com/508407
17:39.49ping1many: cat /proc/cpuinfo ?
17:39.57[TK]D-Fenderjbalcomb: what does 0800 indicate?
17:40.10Kato41no fee
17:40.18manyping1: no. Not CPU Freq, Hz Value.  zgrep HZ /proc/config.gz solved it for me. thanks anyway =)
17:40.22[TK]D-Fenderjbalcomb: and why the "SIP" in that DB entry?
17:40.48many(but config.gz was to trivial to be intuitive *g*)
17:42.24[TK]D-Fenderjbalcomb : I'm suspecting that it only needs a minor rename and a 5 minute fix
17:42.51*** join/#asterisk RoadRunnR (n=MrRoadRu@213.187.82.17)
17:44.10Kato41^^ why dont i have got a /dev/zap ?
17:44.50RoadRunnRhi all
17:44.56wpayneis the zaptel module loaded Kato41?
17:44.57RoadRunnRis the mISDN guru arround?
17:45.31Kato41yes, wpayne
17:46.28manyteh misdn guru?
17:46.34RoadRunnRwell, cricher or crich as he is called in beronet's mantis
17:46.50RoadRunnRsorry, crichter
17:47.18*** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfk1f.dialup.mindspring.com)
17:48.05*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
17:48.51crich1999here i am !
17:49.13Kato41i am looking for a howto for asterisk 1.2 and two hfc-isdn cards
17:49.33crich1999well 3 possibilities: 1. mISDN 2. bristuff 3. visdn
17:49.48RoadRunnRcrich1999: great, we have be bouncing arround messages on the bug #174
17:49.56Kato41euroisdn komp plz
17:50.02*** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com)
17:50.13crich1999thats you RoadRunnR
17:50.23RoadRunnRmight be faster this way ;-)
17:50.42crich1999did you restart yet
17:50.50RoadRunnRcrich1999: i can give you a login on the box, if you want
17:50.55RoadRunnRbox came up just now
17:51.34RoadRunnRoutgoing works again
17:51.46crich1999Kato41: just read http://www.voip-info.org/wiki/view/chan_misdn
17:51.54crich1999hehe thought so
17:51.59Kato41thx crich1999
17:52.13RoadRunnRincoming also ...
17:52.28crich1999Kato41: get the mqueue branch RoadRunnR has got it working now i think
17:52.36SkumlingKato41: I was in your situation until a week ago... tried some different things before ending up with downloading bristuff, wich has a installation routine that downloads zaptel, lipbri and asterisk and compiles the whole bunch itself
17:52.37wpayneanyone using cisco 79xx phones with asterisk?
17:52.55blitz[class]lots of people
17:53.13RoadRunnRcrich1999: this is still the same version that crash logs are from, i guess i can trash it again if i keep the connection open logn enough
17:53.43RoadRunnRwpayne: get snom 320 or 360, they are great
17:53.44*** join/#asterisk Abbas (i=Abbas@203.81.194.208)
17:54.05wpayneI had my 7940 setup up with chan_sccp then my system died and I reinstalled everything. now I'm trying with chan_skinny with varying results.
17:54.19Kato41Skumling: bristuff doesnt do it ... the way i used it...
17:54.31jbalcomb[TK]D-Fender the 0800 indicates 8 AM which is 'that guys' way of noting that the phone is set to after hours
17:54.41crich1999RoadRunnR: what do you mean with still the same .. you mean when your calls are longing too long they crash ?
17:54.42*** join/#asterisk burton (i=mimx@w201.ljudmila.org)
17:55.02crich1999Kato41: whats the problem with bristuff ?
17:55.06[TK]D-Fenderjbalcomb : Ok, you're running 1.2.x?
17:55.35jbalcomb[TK]D-Fender i do not know why 'that guy' chose the confusing DB entry using dnd and SIP as family/key or however that is structured
17:55.46*** join/#asterisk saftsack (n=oliver@p54A7E695.dip.t-dialin.net)
17:55.46jbalcomb[TK]D-Fender yes, Asterisk 1.2.1
17:56.33wpayneI think I've messed up the tftboot stuff, unfortunately I have forgotten the way I had set it up before with chan_sccp. it should be the same with chan_skinny right?
17:57.22wpayneI wish it was just a sip phone.
17:57.26saftsacksomeone here who has experiences with iaxmodem?
17:57.37RoadRunnRcrich1999: what i mean, that this is the new version (svn current) that before show the problems, and those problems only occured after a call was terminated abnormaly, the first calls after a power-cycle use to work before as well, so my guess is that the same problem still is in there somewhere
17:58.19saftsackhttps://sourceforge.net/projects/iaxmodem does the page work for you?
17:58.22*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk)
17:58.57crich1999RoadRunnR: i see. I'm testing the mqueue stuff only on my dev box now, there i very often restart either asterisk or the whole machine .. so i didn't get this behaviour yet
17:59.22*** join/#asterisk frenzy (n=frenzy@196.45.144.40)
18:00.06RoadRunnRcrich1999: i see, i have the trace on now, i'll update the bug as soon as i see the problem again, anything else i should log?
18:00.56frenzyhey all...
18:00.56frenzydo I need to purchase g729 license for able to connect g729 (UAs) ?
18:00.56frenzyor is it only used when asterisk acts as UA?
18:00.57frenzychannel.c:2685 ast_channel_make_compatible: No path to translate
18:01.35frenzy?
18:02.14jbalcomb[TK]D-Fender In the Asterisk CLI executing ‘database put family key value’ produces the database entry ‘/family/key: value’
18:02.14ManxPowerfrenzy, You need a G729 license for most things.
18:02.20[TK]D-Fenderjbalcomb : heres a better sample for you to use - exten => _*62,n,Set(DB(dnd/62)=0800)
18:02.24crich1999RoadRunnR: just the trace and syslog, you can mail me the whole trace when the problem happens, then i can check what happened before. How long does it work till the problem happens ?
18:02.50frenzymy UA is g729 however i'm terminating using ulaw
18:03.04ManxPowerIf Asterisk does not have to transcode (both devices using G729), doesn't have to listen to DTMF (t/T/w/W) or play any sound (voicemail, playback, background) then you prolly don't need a license.
18:03.09frenzyI'm getting no path to translation
18:03.15RoadRunnRcrich1999: varies, seems to depend to the call duration
18:03.16ManxPowerfrenzy, that would be called transcoding.
18:03.21ManxPoweryou need a license to transcode.
18:03.52frenzyohh
18:04.13*** join/#asterisk juice (n=juice@209.33.105.105)
18:04.18frenzyand how are ports counted?
18:04.20crich1999RoadRunnR: when i've finished finding the hfcmulti bug (which causes the unloading crash) i'm going to install the mqueue stuff on our pbx. Thats a good Testenv :)
18:04.25frenzyIf I'm using g729 to connect and at the same time using it to terminate
18:04.30frenzyis that two ports
18:04.30frenzyor one port?
18:04.34ManxPowerfrenzy, you purchase license for X concurrent channels
18:04.39frenzyas its one call
18:04.44ManxPowerusually 1 call = 1 channel
18:04.53frenzyoki
18:04.55frenzythanks
18:05.04[TK]D-Fenderjbalcomb : And check for it like - exten => 1,1,GotoIF($[${DB(dnd/62)}=0800]?4)
18:05.09ManxPowerBut why not just spend the money on a couple of licenses
18:06.01ManxPowerIf you want a three-way call then you would need 2 licenses for it.
18:06.27ManxPowerAnd any MeetMe conference uses 1 license per G729 user
18:08.02jbalcomb[TK]D-Fender so my line exten => s,5,GotoIf($[${dnd} = 0800]?9:6) becomes exten => s,5,GotoIF($[${DB(dnd/62)}=0800]?9:6)
18:09.15[TK]D-Fenderjbalcomb : no need for the "6".  thats implied.  exten => s,5,GotoIF($[${DB(dnd/62)}=0800]?9)
18:09.25[TK]D-Fenderjbalcomb : An you don't need the dbget that preceds it
18:10.41ManxPowerYes, but at some verbose or debug levels it complains if you don't specify both priorities.
18:10.48[TK]D-Fenderjbalcomb : use Set(DB(dnd/62)=nope) to disable
18:11.01[TK]D-FenderManxPower : Really?  how anal....
18:11.25[TK]D-FenderManxPower : Since when does any other language care that you don't have an "else" ?
18:11.28ManxPower[TK]D-Fender, I always do it just to be explicit.  I like explicit.
18:11.54[TK]D-FenderManxPower : I like explicit too, but thats just wrong....  and a PITA if you have to renumber everything...
18:12.21ManxPower[TK]D-Fender, that MIGHT have changed when they added the "n" priority.  I don't know.
18:12.43*** join/#asterisk Chiardon (n=yo@200.71.58.39)
18:12.57Chiardonjustinu, --> Hey
18:13.07*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
18:16.33jbalcomb[TK]D-Fender ah, gotcha on the fall thru. excellent on taking out the DBGet.
18:17.17jbalcomb[TK]D-Fender apparently we are using IPswitchboard to let people turn the dnds on and off so we are stuck with the family/key:value structure and the creating and deleting of the DB entry
18:18.00*** part/#asterisk techie (i=gus@antibala.com)
18:22.06*** join/#asterisk Darwin35 (n=Darwin@sta-208-139-193-162.rockynet.com)
18:22.26*** join/#asterisk bangawanga (n=ahecker@ppp-82-135-70-13.mnet-online.de)
18:22.33*** join/#asterisk Chiardon (n=pirch@200.71.58.39)
18:22.38bangawangahello guys
18:22.42[TK]D-Fenderjbalcomb : Well thats not a bad thing, but you can jsut turn them on/odd from a phone... not sure why you'd use IPSwitchboard to change them... monitor perhaps...
18:23.14*** part/#asterisk m160858 (n=ubuntu@200.89.12.46)
18:23.37Kato41crich1999?
18:23.44[TK]D-Fenderjbalcomb : as for the overall design of your setup I think it can be massively reduced with a few more variables.
18:23.49*** join/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com)
18:23.54crich1999Kato41!
18:24.25Kato41see the priv chat?
18:24.34crich1999no
18:26.04Kato41could you try to invite me instead?
18:27.19crich1999Kato41: just opened a private dialog ot you
18:27.31Kato41u dont see what i write?
18:27.35*** join/#asterisk darkskiez (n=darkskie@bb-195-172-51-236.ukonline.co.uk)
18:27.36Luke-JrLinksys refuses to honor their warranty
18:27.37Luke-Jrwtf
18:27.53*** join/#asterisk jimmy_deanPB_ (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net)
18:28.07crich1999Kato41: no
18:28.18*** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net)
18:28.52jbalcomb[TK]D-Fender yeah, i wasnt aware that we were using IPswitchboard until 30 minutes ago. i assume with macros and variables our config could be more that cut in half but it'll be some time before i 'get it' enough to do that.
18:29.41TrazzI can get extension 500 to work but i can get voicemail or anything else to work ...
18:29.48crich1999Kato41: just offering you a dcc chat, you don't get it i assume ?
18:29.49jbalcomb[TK]D-Fender i really appreciate the info/assist. i assume that understanding the DB and use of it is a big step toward making good use of Asterisk/VoIP
18:30.23docelm0Anyone in here using Mera MVTS?
18:30.25*** join/#asterisk razu (n=razu@213-35-170-76-dsl.trt.estpak.ee)
18:30.42darwin_35<PROTECTED>
18:30.43darwin_35Jan 16 11:30:42 WARNING[71144]: loader.c:554 load_modules: Loading module app_dbodbc.so failed!
18:30.43[TK]D-Fenderjbalcomb : a lot of the time you don't need db for anything, but it has a few nifty places for things exactly like what you're using them for
18:31.29[TK]D-Fenderjbalcomb : I think I could severely shick the size of your setup and make it easier to process...  if you're willing to sub-contract for a small fee :)
18:31.32darwin_35input pls
18:31.36Kato41@crich1999 got a priv chat and a dcc open with you
18:31.58crich1999crich1999: can't be i have none
18:32.26*** join/#asterisk sysdebug (n=sysdebug@200.163.193.247)
18:32.34darwin_35I need this fixed for mysql
18:32.48filedarwin_35: 1. Rude 2. What version of Asterisk
18:32.56darwin_351.2.1
18:33.07darwin_35sorry file forgot to say good morning
18:33.10jbalcomb[TK]D-Fender perhaps. we have a consultant right now that is a jacknut. my boss asked me to decide if we should bring him back. if not i'm to decide if we should get someone else.
18:33.12darwin_35my bad
18:33.17jbalcomb[TK]D-Fender are you local to 216?
18:33.21*** join/#asterisk backblue_ (n=moo@87-196-15-214.net.novis.pt)
18:33.32[TK]D-Fenderjbalcomb : Nope, but obtainable over SIP /  1-800
18:33.35jbalcomb[TK]D-Fender that being Cleveland, OH ;)
18:33.56filedarwin_35: you're using a version of app_dbodbc that uses something that is a patch on the bug tracker, and is not in Asterisk
18:34.32[TK]D-Fenderjbalcomb : Montreal, QC here...
18:35.05[TK]D-FenderSSH / SIP don't care too much about distance, only latency :)
18:35.13darwin_35is there a working ver of app_dbodbc
18:35.14*** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
18:35.14[TK]D-Fender... and bandwidth :D
18:35.22FuriousGeorgehey all
18:35.32darwin_35for asterisk
18:35.33filedarwin_35: there might be... go Google, browse the bug tracker
18:35.58FuriousGeorgei notice my iax minutes provider doesnt like certain 800 numbers, is this common to many voip accounts or is it just me
18:36.26filedarwin_35: next time Google for ast_direct_realtime, because it led me right to the bug number and bug report for it
18:36.27jbalcomb[TK]D-Fender ah, I don't know if we are allowed to work with Canadians. Sorry. =)
18:37.42[TK]D-Fender:O
18:38.53Kato41after making mqueue misdn i got to cp the channels-dir to /usr/src/asterisk/ ?
18:39.23*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:39.31*** join/#asterisk MommomeryCliff (n=willy@host230-24.pool873.interbusiness.it)
18:40.51crich1999Kato41: no you just need to go into asterisk/channels/misdn, then type make and go back to /usr/src/asterisk and type make install
18:40.54*** join/#asterisk g4m (n=g4m@dsl253-014-003.sba1.dsl.speakeasy.net)
18:41.03g4mhowdy
18:41.22FuriousGeorgeanyone else here using nufone?
18:41.35FuriousGeorge4 outbound
18:41.49sivanaFuriousGeorge: yes
18:41.50g4mAnyone have any suggestions on using vonage and asterisk together?
18:41.54*** join/#asterisk dalabera (n=dalabera@pmr.pmrtechnologies.com)
18:42.12sivanag4m: I don't think you can
18:42.14fugitivog4m: yes, choose another provider
18:42.17*** join/#asterisk Assid (n=assid@59.183.31.83)
18:43.19g4mheh
18:43.48g4mfugitivo: like?
18:44.16*** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net)
18:44.22fugitivoteliax, voicepulse, broadvoice, errr, a lot
18:44.48Lotsfug what you think of nuvio
18:45.06fugitivodidn't try it
18:45.26Lotsjust wondering it has the highest ratings on this site.
18:45.27*** join/#asterisk jimmy_dean__ (n=jhodapp@indianalifesciences.com)
18:45.34Kato41i think miranda doesnt support whispering ^^
18:45.39*** join/#asterisk M-I-A- (n=mai@wsp05975102wss.cr.net.cable.rogers.com)
18:45.45*** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net)
18:45.49crich1999Kato41: probably ;)
18:45.56crich1999Kato41: try xchat
18:46.00tzafrir_laptopthe list is back...
18:46.06Trazzis there some basic configs i can look at that has a few extensions and voice mail working?
18:46.38Lotshttp://www.dslreports.com/gbu
18:46.58tzafrir_laptopmozilla/firefox with the irc extension
18:47.09*** join/#asterisk [Sanem] (n=PisiKoLo@stjhnf0112w-142162198201.pppoe-dynamic.nl.aliant.net)
18:47.20fugitivoTrazz: asterisk examples
18:47.21tzafrir_laptop(for windows folks , that is
18:47.24tzafrir_laptop)
18:47.34fugitivobitchX baby!
18:47.36*** join/#asterisk FastJack (i=fastjack@p5091E335.dip.t-dialin.net)
18:47.39Trazzyes i need some examples that work.. i am not getting anything but extesion 500 to work
18:47.54fugitivoTrazz: make samples after you compile asterisk
18:48.15fugitivo?
18:48.47crich1999I am afraid of bitchX users
18:48.54fugitivowhy?
18:49.01*** join/#asterisk Katonka (n=Katonka@p54BEE46D.dip.t-dialin.net)
18:49.03fugitivoi'm afraid of windows users
18:49.10jbroometheir horrible part messages scare me
18:49.12crich1999you should be!
18:49.32Katonkatest
18:49.58Katonkahow to whisper in xchat?
18:50.18crich1999Katonka: are you Kato41
18:50.20crich1999?
18:50.26Katonkayes
18:50.40_jpkis asterisk-user and -devel down at the moment?
18:50.45crich1999you have multiple personalities ?
18:50.47Katonkai think i whisper allready? :)
18:50.57crich1999no i don't think so
18:51.14_jpkproxy:~ # dig mx lists.digium.com
18:51.22_jpklists.digium.com. 52m10s IN MX 10 69.16.138.164.digium.com.
18:51.29_jpkand that dows not resolve...
18:51.53wunderkinit was fixed
18:52.02*** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net)
18:52.05_jpkwhen?
18:52.28wunderkinabout an hour ago, i just started getting a couple messages
18:52.45*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)
18:53.11_jpkic. Just flushed the dns cache.
18:53.33trixterI am thinking that there will be a bunch of queued messages that get delivered soon, and I fully expect 21359139513 emails over the next hour
18:55.15Trazzfugitivo, i have the examples in and reloaded it. is there a cheat sheet on configuring x-lite softphone with asterisk with voicemail, etc ?
18:56.21fugitivoTrazz: sip.conf
18:58.51*** join/#asterisk mikeyb_work (n=michael@66-193-82-211.gen.twtelecom.net)
18:59.46[TK]D-FenderTrazz : There is a specific example in the sample sip.conf
19:01.07mikeyb_workI have a sip channel that did not properly disconnect from asterisk when the call ended.  I do not want to have to restart asterisk... is there any way to reset a sip channel? the output from "show channel SIP/172-22-12-57-8850" tells me "Blocking in: ast_waitfor_nandfds"
19:01.12*** join/#asterisk rhousand (n=rhousand@rrcs-24-199-246-10.midsouth.biz.rr.com)
19:01.34Katonkacrich1999, test
19:01.55crich1999Katonka, test back
19:02.08Katonkacrich1999: is this direct now?
19:03.03*** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net)
19:03.19jbalcombiCEBrkr I can't seem to find anything that discusses the use of that X, do you know where I should look?
19:04.20crich1999Katonka: no i think not
19:06.53Luke-JrWarning to anyone planning to buy Linksys stuff
19:06.57Luke-Jrthey don't honour their warranties
19:07.03jbalcomb[TK]D-Fender do you know about the X iCEBrkr suggested using to avoid the warning when the value is null? I can't find references to its use.
19:07.46*** part/#asterisk jimmy_dean__ (n=jhodapp@indianalifesciences.com)
19:08.20Katonkacrich1999, svn hast to be executed where? to insert it in /usr/src/asterisk...
19:09.25[TK]D-Fenderjbalcomb : didn't follow you there...
19:11.10crich1999Katonka: you will need to get the whole asterisk via svn. svn is the replacement of cvs. just do that: svn co http//svn.digium.com/svn/asterisk/team/crichter/0.3.0 asterisk-0.3.0 and follow the instructions on the wiki page
19:11.29tzangerhahaha
19:11.34infinity1where can we get the latest polycom firmware? i don't see it on polycom's website
19:11.46Trazzcan you hook skype up to asterisk?
19:11.47[TK]D-Fenderjbalcomb : If you meant his sample where he shoved an "X" in front of the DB(), thats completely unnecessary for your sample
19:11.48Katonkacrich1999, i see. thx
19:11.55[TK]D-Fenderinfinity1 : Check with your reseller
19:12.04crich1999Katonka: no problemo
19:12.32infinity1[TK]D-Fender: oh geez. there use to be links online.
19:12.36crich1999Trazz: not yet i think, there's a page on voip-info.org regarding this question
19:12.39*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)
19:12.55[TK]D-Fenderinfinity1 : Polycom doesn't hand out its firmwares to just anybody.  You can ge older versions here : http://www.freedomphones.net/polycom/files/
19:13.12[TK]D-FenderTrazz : No Skype....
19:13.29infinity1[TK]D-Fender: i just got this 601 and it has more current firmware than what i can download. but my other polycom is still old
19:13.42infinity1can someone dcc me the latest polycom firmware ? :)
19:13.52[TK]D-Fenderinfinity1 : What version do you have on it?
19:14.12infinity1[TK]D-Fender: br 3.1 and siup 1.6.2 on the new one
19:14.20[av]banii'm so disappointed polycom doesnt support xml on the 501 >:(  wtf is the point of the bitmapped lcd?
19:14.23infinity1[TK]D-Fender: br 2.6.1 and sip 1.5.2 on the old one
19:14.53*** join/#asterisk justinu (n=j2@72.18.13.34)
19:15.16[TK]D-Fenderinfinity1 : You can get 1.6.2 off the site I gave you, and just ask your reseller to give you 1.6.3
19:15.28[TK]D-Fender[av]bani : a pretty logo :)\
19:15.34[av]bani>:(
19:15.43[TK]D-Fender[av]bani : Thats why my setup is 100% 60x :D
19:15.50[av]banii get a pretty logo with gxp-2000 :P
19:15.58infinity1[TK]D-Fender: atacomm should just email it when you buy a fone. heh
19:16.11Lotsjust curious if there is anyone here who is in the voip solution provider business that i could ask a few questions?
19:16.22[TK]D-Fenderinfinity1 : yeah and keep in mind how many mailboxes would reject an attachment that size....
19:16.32infinity1[TK]D-Fender: yes yes. :P
19:16.43[TK]D-FenderLots : just ask the question and spin the wheel!
19:16.52justinuspin the bottle
19:17.35infinity1ahah ...atacomm runs * ...
19:17.38infinity1go figure
19:17.41justinuwhy not?
19:17.45TrazzTK, any webbased gui's that work ?
19:17.57fugitivoanyone using a commercial ASR?
19:17.58jbalcomb[TK]D-Fender ok. thanks. just for educational purposes though, would there be some place to learn about its use?
19:17.58[TK]D-FenderTrazz : GUI for what?
19:18.14[TK]D-Fenderjbalcomb : ... huh?!
19:18.15Trazzgui to configure system and administer it
19:18.21Lotstkd uhh ok, well, i'm just trying to get business owners interested in voip, and was wondering if anyone knew of some good roi calculators i could use when selling voip to clients.
19:18.32[TK]D-FenderTrazz : AMP.  Though I don't suggest it or any other....
19:18.37[TK]D-Fender(for you)
19:18.53Lotsinfinity, is atacomm a good reseller?  I haven't read any good reviews about them.
19:19.07[TK]D-FenderLots : Just get pricing for other solutions and you'll have enough to show how much they can save.
19:19.13infinity1Lots: they get recommended a lot. i have no complaints
19:19.16[TK]D-FenderLots : You need to learn about the competition.
19:19.17TrazzTk, i was hoping the gui would help to visualize the config for extensions, voice mail and ivr setup
19:19.40infinity1Lots: descent prices. and they don't try to fuck you at the end of your online order with additional fees
19:19.49[TK]D-FenderTrazz : Forget GUI... a basic setup is dead easy.  you just need to take it on 1 file at a time.
19:20.25iCEBrkrYay! More 1wk deadlines
19:20.41[TK]D-FenderiCEBrkr : But are they retro-active?! ;)
19:20.47infinity1Lots: i'm on the fone with atacomm right now. they are giving me the ftp login for the firmware
19:20.57justinunice
19:21.28[TK]D-Fenderinfinity1 : While you're at it can you ask for admin guides for the Citel gateways? ;)
19:22.00infinity1[TK]D-Fender: shit. i just hung up before i looked at the screen agin.  i was trying the ftp
19:22.04[TK]D-Fender:O
19:22.10infinity1[TK]D-Fender: did you check their ftp site?
19:22.17[TK]D-Fendernope, credentials?
19:22.47infinity1[TK]D-Fender: i msged ya
19:25.13Lotstkd well i was just trying to find a average comparison chart somewhere on the net that had the comparison of a traditional TDM system vs. a VOIP system.
19:25.22[TK]D-FenderI may just phone them up for it.... since I'm going to need that info to promote them to other clients who can't change their wiring...
19:26.40[TK]D-FenderLots : I think there has beena  few like that floating around, but get some specific comparisons and remember cost isn't everything.  Its what they get vs what it costs vs what they're prepared to spend.
19:26.47[av]banihttp://bani.anime.net/phonez/
19:26.49[av]baniupdated \o/
19:26.50jbalcomb[TK]D-Fender just looking for some reference material that explains the use of that 'X'
19:27.06[TK]D-Fenderjbalcomb : was it the one in the GotoIF?
19:27.14infinity1Lots: feature wise its probably the same.
19:27.30infinity1Lots: but you get SIP with VoIP, which is where things get interesting
19:27.38jbalcomb[TK]D-Fender yes'm
19:27.55Lotsmostly i just want a price chart as part of my selling points, not the entire presentation.
19:28.01Assidhrmm.. remind me never to get a cisco phone
19:28.08Assidi just tried my friends 7960
19:28.12*** join/#asterisk svenna_ (n=svenna@p548D1EDB.dip0.t-ipconnect.de)
19:28.15Jammyhey guys got a quick question... for incoming calls i want zapteller to answer, forgot how to enable that in extensions.conf and cant find documentation on that particular function.. i have exten => s,1,Zapteller so far...anyw ideas whats wrong?
19:28.22iCEBrkrI dunno man, this sucks.. I've kinda slacking on these projects cuz I was informed we don't need the Asterisk box.. Now all of a sudden.. We need it
19:28.23Assidthe polycoms are like soooooo easy to provision
19:28.25jbalcombcisco is the most awesome ever.
19:28.37iCEBrkrjbalcomb: Liar
19:28.39*** join/#asterisk [Sanem] (n=LoGo@stjhnf0112w-142162201040.pppoe-dynamic.nl.aliant.net)
19:28.43Assidjbalcomb: the craziest to configure
19:29.07jbalcombiCEBrkr HA! just because you have to have a brain to understand Cisco doens't mean it ain't the most awesome.
19:29.26jbalcombAssid Agreed but therein lies the power.
19:29.43iCEBrkrjbalcomb: I dunno man, some of their router stuff isn't the bestest out there :P
19:29.57jbalcombPeople who think Cisco is too complicated probably don't watch movies you have to read either,
19:30.40jbalcombiCEBrkr hrmm.. such as? Juniper touches them in the highend 7xxx/12xxx market but otherwise I haven't seen a contender.
19:30.52iCEBrkrAll I'm saying is to look into Foundry
19:31.03*** join/#asterisk chapeaurouge (n=chap@user-85-201-81-201.tvcablenet.be)
19:31.18jbalcombiCEBrkr Foundry is indeed decent, next in line I'd say.
19:31.35iCEBrkrIt's one thing to say you have a 24 port gigabyte switch but the problem is the backplane can't really handle the through-put of all 24 ports at the same time..
19:31.52jbalcombiCEBrkr part of the joy of Cisco is in having their entire product line available.
19:32.02iCEBrkrDeep pockets..
19:32.39jbalcombiCEBrkr often cheaper to only spend life learning one CLI/IOS than mad labor hours trying to keep track of three or four product lines.
19:32.51[TK]D-Fenderjbalcomb : there is nothing to say about that X.  Unneccesary
19:33.15iCEBrkrjbalcomb: I hear that, but like I was saying.. A switch just isn't a switch and a router just isn't a router.
19:33.18jbalcomb[TK]D-Fender okidoki. I'll drop it from my thoughts. Thanks again.
19:33.29[av]banipart of the grey hair going bald of cisco is they DONT FREAKING REGRESSION TEST RELEASES  >:( >:(
19:33.35Lotstkd you a solution provider?
19:33.54[av]baniin fact i suspect cisco has no test methods at all...
19:33.55[TK]D-Fenderjbalcomb : glad to help
19:33.58*** join/#asterisk [hC-] (n=hardcore@209.153.195.139)
19:34.00jbalcomb[TK]D-Fender is the best Asterisk consultant I've ever seen.
19:34.07*** join/#asterisk squinky86 (n=ASGjon@unaffiliated/squinky86)
19:34.20mog_workthree cheers for fender!
19:34.20[TK]D-Fender[av]bani : You should add that there IS PoE for Polycom IP 30x, 50x with an add'l adapter cable
19:34.27jbalcombhighly recommended if you need lengthy expert assistance.
19:34.30iCEBrkrjbalcomb: Oh yeah, and your 'X' question.. It's just a place holder.. It's not a command.  and you don't use it in your DB() function.
19:34.45iCEBrkrjbalcomb: [TK]D-Fender is a frick'n nerd man..
19:34.47[av]banifender, or $15 if you use any off the shelf poe adapter setup
19:35.01[av]banii dont include optional poe if they're just silly kits
19:35.09*** part/#asterisk Davey|Work (n=davey@unaffiliated/davey)
19:35.10fugitivojbalcomb: there's always paypal to help people that helps
19:35.13[TK]D-FenderiCEBrkr.... don't make be break you :D
19:35.25iCEBrkr:P
19:35.33[av]banii include them if they're weird nonstandard poe like cisco
19:35.36[TK]D-Fender[av]bani : Its a selling point thoug, just reverse of the Cisco method...
19:35.42jbalcombiCEBrkr frick'n nerds are my favorite people to contract. they waste unbillable hours fixing things for me and take such pride in the Quality of thier work that I don't have to hire someone else to doub check it.
19:36.00[av]banifender, why include a $50 when i can do it myself with off the shelf for $15
19:36.01iCEBrkrjbalcomb: We know this!
19:36.05[av]baniself defeating
19:36.20[TK]D-Fenderjbalcomb : And for the good stuff my rates are very acceptable! ;)
19:36.46[TK]D-Fender[av]bani : You have to mention that the input requirements ALLOW you to use such methods.  Not all do....
19:36.54Lotsjbal yep thats a habit i have to get away from in my business.
19:36.55[av]banithe great thing about fender's advice here is that if you're dissatisfied you can get a money back guarantee
19:37.00jbalcomb[TK]D-Fender :) perhaps I'll delve into a terrible conflict of interests and contract you through my company to this place. ;)
19:37.14Lotsi find myself doing the same thing then i think to myself wait... am i getting paid for this crap?
19:37.15Lotslol.
19:37.16[TK]D-Fenderjbalcomb : Paypal works....
19:37.33mog_worklol
19:37.47*** join/#asterisk slak- (i=slak@rewted.biz)
19:37.49jbalcombLots yeah, I don't do /extra/ work. We agreed to this, its done, I recommend fixing this too, new contract.
19:37.52slak-hi :)
19:37.54[TK]D-FenderI did a big "heads up" for a guy just for free and he was rather thankful and paind me anyways...
19:38.00slak-where should i begin looking for a voice t1 line..
19:38.02*** join/#asterisk brookshire (n=nubb@gateway.digium.com)
19:38.06slak-already have a data t1 here
19:38.07[av]banisnom would be an easy choice if they did xml...
19:38.20tzangerslak-: contact your local RBOC, then after the sticker shock wears off contact a bunch of the ILECs in your area
19:38.29*** join/#asterisk g4m (n=g4m@dsl253-014-003.sba1.dsl.speakeasy.net)
19:38.48[TK]D-Fender[av]bani : Too true!  If you don't like my free advice you get DOUBLE your money back!
19:38.50jbalcomb[TK]D-Fender you'll get no 'thank you' money from me. ;) my lawyer and accountant would not approve.
19:39.00iCEBrkrLOL
19:39.11rob0Double your money back, less a restocking fee of course
19:39.16znoGis there a list of options available for the VoicemailMain "0" option? (change pass, etc)
19:39.18Lotstkd, i'm new to this voip stuff, i quoted a guy for a 24-phone system with phones cabling and configuration about 10,500.00 for a hosted ip-pbx solution..
19:39.19[TK]D-Fenderjbalcomb : I meant in compensation were I to completely renovate your config :)
19:39.28[TK]D-FenderiCEBrkr : I think I've done PLENTY for you already ;)
19:39.37Lotsthats seems on the cheap side to me though.
19:39.40jbalcombdo keep in mind that if someone does work for you and you do not pay them they are not responsible by law, atleast in Ohio.
19:39.42iCEBrkrlol
19:40.04jbalcomb[TK]D-Fender I think we've all /done/ plenty for iCEBrkr
19:40.10[TK]D-FenderznoG : its all on the WIKI...
19:40.18slak-how much can i expect to pay for a ~10 channel t1 line
19:40.26slak-i already have a data t1 can i use that?
19:40.27iCEBrkrjbalcomb: oh come'on
19:40.31[TK]D-Fenderslak- : Depends where and what company....
19:40.37tzangerslak-: depends entirely on your company and location
19:40.37iCEBrkrjbalcomb: you havent' done anything for me but give me a headache
19:40.38*** part/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com)
19:40.40slak-okay roughly?
19:40.52[TK]D-Fendertzanger : tag!
19:40.53tzangerI, for example, am in a Rate 4 group with Bell Canada.  That's the "don't use the lube" rate group
19:40.54jbalcombslak- if you can setup Multi-Link PPP i beleive can run data/voice on the same PRI
19:40.56tzanger[TK]D-Fender: haha
19:41.01iCEBrkr<-- phone
19:41.13tzangerjbalcomb: you don't need MLPPP for that
19:41.20slak-jbalcomb: is that something that my current data t1 provider would have to agree and do for me
19:41.33jbalcombiCEBrkr *sniff* don't you remember all those 'GET * FROM *' queries I killed for you?
19:41.37[TK]D-Fendertzanger : Mine is at $700 or so for full PRI.  I'm in a great area though
19:41.42slak-tzanger: who or what assigns phone numbers to t1 channels
19:41.44tzangerslak-: yes, and remember you'd only have 11x64kbps availbale for data then
19:41.53tzangerslak-: your provider.
19:41.55slak-okay that sucks
19:42.05slak-so its best to just get another line
19:42.16tzangerwwell maybe 12... depends on whether you're getting CAS or CCS T1
19:42.34tzangerCAS T1 = 24 56kbps channels.  CCS T1 = PRI = 23 64kbps + Dchan
19:42.45[TK]D-Fendertzanger : CAS?  You can do split signalling with PRI can't you?  CAS blows...
19:42.47slak-how much better is this voice t1 config to what i'm doing with sip accounts
19:42.53*** join/#asterisk [}btorch] (n=kvirc@208.63.19.172)
19:43.01slak-s/to/than
19:43.02*** part/#asterisk [}btorch] (n=kvirc@208.63.19.172)
19:43.36*** join/#asterisk }btorch{ (n=kvirc@208.63.19.172)
19:43.38[TK]D-Fenderslak- : Depends, what are you doing with "sip accounts"?  incoming lines?
19:43.52slak-yes incoming for conferencing
19:44.06}btorch{hey anyone here knows a good and cheap USB phone that I can use with * (SIP)
19:44.08[TK]D-Fender[av]bani : USe atacomm pricing for your snom entries... a fair bit cheaper...
19:45.00[av]banilast time i checked atacomm didnt have snom
19:45.02[av]banii'll check again
19:45.15[TK]D-Fender[av]bani : THEY HAVE FOR A long TIME...
19:45.40[TK]D-Fender[av]bani : And for BLF you can add Polycom to the list for those... Lines - 1
19:45.43[av]bani[TK]D-Fender: YOUR caps lock IS BROKEN
19:45.58[TK]D-FenderWhAt ArE YoU tAlKiNg AbOuT?!
19:46.11[av]baniif you refresh page you'll notice blf is listed on all the polycoms
19:46.17[TK]D-Fender[av]bani : I work a lot in caps here so if I slip up, just chill....
19:46.41[TK]D-Fender[av]bani : not yet....
19:47.06wunderkinya fender has that problem a lot :P
19:47.25[av]banithere, got the new prices in
19:47.26[TK]D-Fenderwundaboy : I prefer to think of it as a "feature" :F
19:47.47TrazzTK, i cant seem to find this script usr/src/asterisk/addmailbox
19:47.52Trazzyet it says its supposed to be there
19:47.55[TK]D-Fender[av]bani : and I mixed up the columns...
19:48.04[av]bani\o/
19:48.08[TK]D-FenderTrazz : says who?
19:48.09*** join/#asterisk samueltc (n=sam@69.156.67.214)
19:48.12samueltchi
19:48.15[av]banicolumnar mixture is <3
19:48.24Trazzi looked all throgh my directories
19:48.35fugitivov
19:48.35fugitivoj
19:48.51[TK]D-Fender[av]bani : You need to update the "B/L" for Polycom...
19:48.56wunderkinfender types with his toes and his eyes closed hehe..
19:49.03samueltcanybody got app_prepaid_* working with postgresql?
19:49.10[av]banifender, provide #'s
19:49.23[TK]D-Fender2 hands tied behind my back.... in winter... 20' of snow... uphill ..... BOTH WAYS!
19:49.38*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
19:49.39samueltcI've got the one from commoveo
19:49.40[TK]D-Fender[av]bani : BLF & B/L = [Lines -1]
19:49.59[TK]D-Fenderso if you are on a Polycom 601, you can have 5 BLF / programmable keys
19:50.03[TK]D-Fendermax
19:50.08[av]baniwhy -1 ?
19:50.13[TK]D-Fenderunless you add a side caddy
19:50.22wunderkingood thing they aren't called milf ...
19:50.29iCEBrkro/~ BLF! Uh-huh.. o/~
19:50.39TrazzTk, where should that addmailbox be ?
19:50.49[TK]D-Fender[av]bani : the way line keys work on them.  You can give a single registration 1 line key supporting multiple call/key, and use the remaining 5 for BLF / speed-dials
19:50.59[av]bani:/
19:51.10jbalcomb[TK]D-Fender iCEBrkr What is the bestest phone known for Asterisk/VoIP?
19:51.12[TK]D-FenderTrazz : there is no "magic" script for that, you just write an entry into voicemail.conf!
19:51.35Trazzok i did that.. i was reading where it said it make some sample greetgins. etc
19:51.36[TK]D-Fenderjbalcomb : "bestest"..... Grammar very is english good yours!!!
19:51.45[av]banispeak like yoda fender does
19:51.53iCEBrkrjbalcomb: I dunno, I prefer ATA's :P
19:52.07[TK]D-FenderDo or do not... there is no tryhhhmmmMMMM!!!
19:52.16wunderkin[TK]D-Fender, sucky suky looong time!
19:52.17jbalcomb[TK]D-Fender =) I have a small fascination with slightly silly or rediculous character.
19:52.38iCEBrkr[TK]D-Fender: in other words, he retarded
19:52.46jbalcombiCEBrkr would i think that clever or funny if i knew what ATA's were?
19:52.55[TK]D-Fenderjbalcomb : I'd say the Cisco 7960G is probably the best phone, but the Polycom is a better value.
19:52.56iCEBrkrjbalcomb: oh yeah
19:52.59Lotsanyone know of a good "VOIP Business" irc channel or a good forum to look at for the VOIP Business?
19:53.12jbalcombiCEBrkr yes, 'he retarded' ... you picking up on the ebonics?
19:53.15[av]baniwhat's the lcd rez of the 7960g?
19:53.24iCEBrkrjbalcomb: www.sipura.com  SPA2000
19:53.39[TK]D-Fenderjbalcomb : SPA-2002 <- Be current!
19:53.52iCEBrkrYea yeah
19:53.54iCEBrkr:P
19:54.18iCEBrkrI got three of these SPA-2000's
19:54.21iCEBrkrNot sure how I got three.
19:54.39[av]banio_O
19:54.52[TK]D-FenderiCEBrkr : I run SPA-2000, 3000, 941 at home and have 2000,2001's here at work
19:55.06jbalcombany feelings about the SNOM phones?
19:55.16[av]banifender, any idea how to increase the dialing speed of the 3000?
19:55.39[av]banilike you can do with eg modems, ATS11=40 or whatnot
19:56.13[TK]D-Fender[av]bani : Another update the IP 30x's LCD is measured in characters, not pixels.  you may want to comment that in...
19:56.45[TK]D-Fender[av]bani : not the diaplan timeout, but rather the DTMF transmission rate to PSTN?
19:56.54[TK]D-Fender[av]bani : if so, not offhand
19:56.56[av]banifixed
19:57.02[av]baniyes, the dtmf rate
19:57.09[av]baniits pretty lethargic
19:57.40[av]banipage updated for 301
19:57.46docelm0hay iCEBrkr is the geek convention on the 26th?
19:58.10[av]banibummer no backlight on any of the polycoms. a tilted display doesnt help if the phone is in shadow
19:58.16XIN01OZsup fellow asterisk boxers
19:58.20ruud_org[av]bani: are you talking about the SPA-3000?
19:58.25[av]baniruud_org: yes
19:58.40ruud_orgif so, the option you want is called "PSTN Dial Digit Len:" in the PSTN Line page
19:58.51ruud_org(i believe)
19:58.54jbalcombhrmm.. no backlight is a bummer. some of the people here pitched a fit to get us to turn on the backlight on thier grandstreams
19:58.56[av]baniits .1/.1 which should be pretty fast, but its not...
19:59.07ravenpi[av]bani: Yeah, that's the -one- thing I wish they did differently with the Polycom.
19:59.07[TK]D-Fender[av]bani : I think SNOMS's have backlight as well as GS's, but thats it...
19:59.10[av]banifender, aastra
19:59.11*** join/#asterisk acidfoo (n=acidfoo@Kitchener-HSE-ppp3578579.sympatico.ca)
19:59.19[TK]D-Fender[av]bani : Ah yes...
19:59.39ruud_orgon my spa-3000, i don't hear it dialing on the pstn, so i don't know whether the dalay in call set up is a result of slow dtmf or something else
19:59.50[av]baniyou can hear it muting the digits as it dials out
19:59.52XIN01OZwhat is suggested for voip to make termination to the PSTN cost wise .. a T3?
19:59.54ruud_orgi guess i could listen in with a second phone in parallel on the pstn line
20:00.04[TK]D-Fender[av]bani : Hey, why don't you break upthe LCD column into multipl colums?  res, type, backlight, etc?
20:00.08[av]bania bit faint but you can still hear some bleed through
20:00.14XIN01OZand be able to easily profit
20:00.29ruud_org[av]bani: interesting... never noticed that, but to be honest never tried to pay attention to it either
20:01.05[av]baniruud_org: it's kind of annoying the spa-3000 connects the voip to pstn during hte dialing.. i wish it wouldnt do that till it was finished
20:01.09*** join/#asterisk dalabera (n=dalabera@pmr.pmrtechnologies.com)
20:01.12[TK]D-Fender[av]bani : Cisco 7905G supports Cisco PoE <-
20:01.28[TK]D-Fender[av]bani : yeah the muted DTMF bit annoys me too...
20:01.48[av]banithats how i was able to tell it was dialing sluggish
20:02.02[av]baniyou can blast out dtmf on modems and most switches will pick it up
20:02.15[av]baniut the spa-3000 dials out all pokey and slow
20:02.26*** join/#asterisk RussCC (n=face@216.157.205.211)
20:02.33[TK]D-Fender[av]bani : I'll check it out when I get home
20:02.35ruud_org[av]bani: strange, never noticed that it did... i hear the spa-generated dial-tone, dial the number, wait a while, and then finally hear the ringing... i don't hear anything regarding the PSTN dialtone or DTMF tones, but maybe this is all so faint that i didn't notice it
20:02.54[av]baniits possible if you've dorked with the gain settings
20:02.58[av]banithat you wont hear it
20:03.13jbalcombiCEBrkr For "Unable to handle indication 3" am I looking at a HANGUPCAUSE issue?
20:03.54[av]bani7905g updated
20:04.11ruud_org[av]bani: those are default (and call quality is OK), but if i turn up handset volume in my phone, i can hear it faintly in the background, now that you mention it
20:04.16[av]baniwhy wont cisco say the lcd rez of the 7960g...
20:04.19ruud_orglearn something new every day :)
20:04.22[av]bani:)
20:04.37RussCCHello, I have a question about Cisco configurations to allow traffic coming from my provider any ideas?
20:04.54RussCCCisco Pix
20:05.37RussCCI am able to make outbound calls just fine
20:05.52[av]baniyay .05/.05 works
20:06.11[av]banidials 7 digits in under a second :)
20:06.23*** join/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net)
20:06.40RussCCand when I have trafic completly open it works but once I start to lock it down I lose the ability to make a call
20:07.31RussCCthe number I am calling rings but when the handset is picked up the line is quiet and if hung up the line will ring again
20:08.06fugitivoRussCC: udp 5060, udp 10k to 20k
20:08.15fugitivo(for sip)
20:08.24*** join/#asterisk Katonka_ (n=Katonka@p54BEE46D.dip.t-dialin.net)
20:08.32jbalcomb[TK]D-Fender For "Unable to handle indication 3" am I looking at a HANGUPCAUSE issue?
20:08.51[hC-]This is odd, when i dial local calls on my pri, it generates ringing tones, and so does asterisk (double ring) without specifying the r option to dial.  However, calling 800 numbers on it, i get no ringing at all
20:08.56RussCCfugitivo: thank you I will try that out
20:09.03[av]bani[TK]D-Fender: wonder if i should list skinny phones on there. cisco has some nice non-sucky 79xx models (eg do real 802.3af, not crippled, etc) but they only do skinny
20:10.03[av]baniand you can get used nortels for cheap, which speak the proprietary nortel (for which there is a * driver :)
20:10.05RussCCso I need to allow ports 5060 and 10k - 20k? Is that correct?
20:10.15fugitivoRussCC: for sip, yes
20:10.19fugitivoRussCC: udp
20:10.26RussCCfugitivo: thank you
20:10.48[TK]D-Fenderjbalcomb : That seems to be a ringing indication mismatch
20:11.09[TK]D-Fender[av]bani : Screw Skinny.... experimental and proprietary...
20:11.11TrazzJan 16 14:23:03 NOTICE[13437]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
20:11.12TrazzJan 16 14:23:03 WARNING[13437]: app_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for '2001'
20:11.23[av]baniexperimental? i'm sure cisco would take issue with that :)
20:11.24Trazzi have edited my voicemail.conf, extensions.conf and sipconf
20:11.31[TK]D-Fender[av]bani : Experimental in * !
20:11.51fugitivoTrazz: you didn't do it correctly thn
20:11.52[TK]D-Fender[av]bani : Don't make me "trout" you.....
20:11.52fugitivothen
20:12.40TrazzFugitivo, i added it [local]
20:12.41Trazz2000 => 1234,Joe Blow,joe@gmail.com
20:12.47Trazzthats in the voicemail.conf
20:12.52Trazzdid reload
20:12.59Trazzit was already there
20:13.23fugitivoTrazz: it should be [default], not [local]
20:13.32Trazzok
20:13.34fugitivounless you specify @local when calling voicemail
20:14.23[hC-]is there a defacto way to handle/troubleshoot double ring/no ring issues over pri?
20:14.33[hC-]should i be checking tone zones, or... ?
20:14.36[hC-](im in canada)
20:19.33tzanger[hC-]: first off
20:19.41tzangeruse "progressinband=no" in zapata.conf
20:19.47tzangerstop/restart * and see if that fixes it
20:19.52tzangerwho is your PRI provider?
20:20.01[hC-]okay. Allstream (formerly at&t canada)
20:21.17[hC-]its gonna be tricky to restart this thing, i gotta do it over lunch when nobody's using it
20:21.48tzanger[hC-]: yep.  or just issue a "stop when convenient" and wait, setting your terminal to beep on silence.  :-)
20:21.56[TK]D-Fender[hC-] : I'm with Allstream, and there was nothing funny to set here....
20:21.59tzangerI don't trust restart when convenient because I'm not sure that a restart is enough
20:22.48[hC-]modifying zapata.conf should be sufficient to use restart, no?
20:23.03tzanger[hC-]: well as I said I'm not sure.  that's why I am suggesting a good solid shutdown
20:23.26[hC-][TK]D-Fender: hmm. I get double-ring it seems on local 604 calls, *seemingly* only from certain brands of phones (i hear it on sipura, but not cisco) and on 800 calls, i get no ringing tones at all.
20:24.10[TK]D-Fender[hC-] : this is after it connects the 2 endsand begins ringing?
20:24.17Katonka_crich1999, installed asterisk03, but the misdn.conf is missing
20:25.10[hC-][TK]D-Fender: for the double ring? yeah, after it passes it off to the zap channel, i hear two ringing tones that over lap one another, slightly off-timed
20:25.28*** join/#asterisk l1nux (i=moi@214.138.103-84.rev.gaoland.net)
20:25.37l1nuxhi :)
20:26.08[TK]D-Fender[hC-] : Sounds a bit like echo....
20:26.28[TK]D-Fender[hC-] : You you using "r" in your dial command?
20:26.34[hC-][TK]D-Fender: nope.
20:26.39l1nuxasterisk-xmpp ready for testing ?
20:26.46[hC-][TK]D-Fender: and i dont think its echo, neither parties notice echo in their voice during the call.
20:27.02l1nuxsvn-8106 !
20:27.37[TK]D-Fender[hC-] :/
20:27.44*** join/#asterisk katakefalos (i=katakefa@194.214.77.65.in-addr.arpa.ethernext.com)
20:28.53katakefalosi am looking for support for a newly purchased TE110P
20:30.39katakefalosi dont see anyone chatting its hard to believe with so many users online there must be something wrong can someone msg me directly?
20:30.52[hC-]arg. brb.
20:31.00[TK]D-Fenderkatakefalos :It gets quiet at time.  How about you just ask your question....
20:31.05katakefalosoh ic :)
20:32.07*** part/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
20:32.30katakefalosi need to configure my TE110P to provide a PRI to a legacy PBX system and am on hold for about 20 minutes on digiums phone support sp i logged in here
20:32.33[TK]D-Fender[hC] : Care to pastebin your zapata.conf?
20:32.51[TK]D-Fenderkatakefalos : While continuing to hold I hop?
20:33.14katakefalossure one sec
20:33.46katakefalosis it pastebin.org or so?
20:34.23[TK]D-Fenderkatakefalos : Wasn't a message for you.  What is that actual problem?
20:34.48[TK]D-FenderAnd for general reference :
20:34.49[TK]D-Fender~pb
20:34.51jbotrumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
20:35.09RussCCfugitivo: If I have a static map and I am allowing any UDP traffic from my VOIP provider I still am unable to make a call
20:35.36[TK]D-FenderRussCC : You're behind NAT I take it?
20:35.37katakefalosmy zapata.conf : http://pastebin.com/508676
20:36.04Katonka_is it possible to use misdn with asterisk@home?
20:36.22[TK]D-Fenderkatakefalos : You'll find AMP support hard come-by....
20:36.29*** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk)
20:36.30RussCC[tk]d-fender: yes I am using nat
20:36.49katakefalosand my etc/zaptel.conf : http://pastebin.com/508679
20:36.55[TK]D-FenderRussCC : You need to fill in either EXTERNIP or EXTERNHOST, and LOCALNET.  Have you don't all of these?
20:37.31RussCC[tk]d-fender: I don't quite follow you
20:38.35[TK]D-FenderRussCC : You need to set those values in the [general] section of SIP.CONF for it to work.
20:38.39*** join/#asterisk Davey|Work (n=davey@unaffiliated/davey)
20:38.54RussCC[tk]d-fender: ok thank you
20:39.16[TK]D-Fenderkatakefalos : Your channel declaration looks wrong, it should be "channel => 1-13".  You were missing the ">"
20:39.23Davey|WorkHi there, I'm having issues with one of our SIP clients, when I call him, he gets the ringing, but I get one ring and straight to his voicemail, he cannot dial out or to voicemail, any suggestions where to look?
20:40.08[TK]D-FenderDavey|Work : Did his setup ever work?
20:40.31*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
20:40.32TrazzTK, whats the easies way to test music on hold?
20:40.48katakefalosFende: tehy just picked up
20:41.15[TK]D-FenderTrazz : exten => 1,1,MusicOnHold
20:41.25Davey|Work[TK]D-Fender, for outbound only
20:41.50[TK]D-FenderDavey|Work : You said he can
20:42.02[TK]D-Fender't "dial out".  What is this "outbound" that is working?
20:42.04Davey|Workthe only change since then is a new router/DSL modem, so I'm guessing the problem is there. But he has opened up all the ports for both TCP/IP and UDP traffic
20:42.17Davey|Work[TK]D-Fender, he *could* dial out, now he can't
20:42.22RoyKdrumkilla: ping
20:42.33[TK]D-FenderDavey|Work : So he's behind NAT, and what about you?
20:42.51Davey|Work[TK]D-Fender, Asterisk is on our gateway machine
20:43.06*** join/#asterisk [hC-] (n=hardcore@209.153.195.139)
20:43.13[TK]D-FenderDavey|Work : Pastebin his sip account info
20:43.24Davey|Work[TK]D-Fender, sure, anywhere in particular?
20:43.24[TK]D-Fender[hC] : Care to pastebin your zapata.conf?
20:43.29[TK]D-Fender~pb
20:43.30jbotwell, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
20:43.42[hC-][TK]D-Fender: yeah sure, give me just a sec.
20:43.43TrazzTk, no music on hold just silence with that
20:43.47RoyK~seen drumkilla
20:43.54jbotdrumkilla is currently on #asterisk. Has said a total of 15 messages. Is idling for 4h 51m 23s, last said: 'backblue: you can use DUNDi in combination with using the regexten feature in sip.conf.  That will, the extension will only exist when the peer is registered'.
20:44.04[TK]D-FenderTrazz : you may have to specify a context with that.
20:44.25[TK]D-FenderTrazz : You may also have to shove an Answer in front, etc...
20:44.31Trazzok
20:44.41Davey|Work[TK]D-Fender, http://pastebin.com/508689
20:44.53Davey|Work[TK]D-Fender, essentially the same as all our other clients
20:45.09[TK]D-FenderDavey|Work : What is he using as a phone?
20:45.16Davey|WorkSipura something or other
20:45.30[TK]D-FenderDavey|Work : Sipura should use INFO for the DTMF mode
20:45.35Davey|WorkOK, let me try that
20:45.42[TK]D-Fenderand get rid of defaultip
20:46.06Kattymister fender.
20:46.10Davey|Workdoes he need to re-connect his phone for Asterisk to find it?
20:46.14[TK]D-FenderMs. Katty.
20:46.35[TK]D-FenderDavey|Work : perhaps, and he should go into the ATA's web config to set the DTMF mode.
20:46.36TrazzTK, i did that and nothing on console and call answers but its dead air
20:46.48Davey|Work[TK]D-Fender, OK
20:46.53Davey|Workthats the web config, right?
20:46.57[TK]D-FenderTrazz : Do you have MP3's in the appropriate folder and MPG123 installed?
20:47.21Trazzi installed mpg123 yes
20:47.26Trazzlet me check on files
20:48.54argentasTrazz: what does 'mpg123 --version' show?
20:49.37Trazzmpg123: Unknown option "version".
20:50.07argentasok, that's cool, just wanted to check it wasn't your distro installing mpg321
20:50.17Katonka_what has asterisk -r to say which version ist is if i installed the mqueue misdn?
20:50.41Trazzyep. i downloaded it and installed it seperately
20:50.47argentasok, good.
20:51.09Trazzfile is installed in /usr/local/bin
20:51.24argentaswhen you dial the extension, does 'ps ax' show that mpg123 is running?
20:51.39[TK]D-FenderDavey|Work : yes, the SPA's web config
20:51.54Trazzthis is on console now
20:51.56TrazzJan 16 15:00:45 WARNING[13369]: res_musiconhold.c:336 spawn_mp3: /var/lib/asterisk/mohmp3 is not a valid directory
20:51.56TrazzJan 16 15:00:45 WARNING[13369]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player
20:52.12[TK]D-FenderTrazz : I think that error pretty much explains itself...
20:52.22Trazzmy install is not in /var/lib its in /opt/voip
20:52.42Davey|Work[TK]D-Fender, no luck
20:52.43[TK]D-FenderTrazz : non-standard... bad start.
20:53.09[TK]D-FenderDavey|Work.  And you reloaded your config, had him make the chages and reboot his device?
20:53.18Trazzi was told i could make everything run from 1 directory so i changed install_prefix
20:53.46fugitivowhere can i get cheap soundpoint 501 and 601?
20:53.53*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
20:53.54Davey|Work[TK]D-Fender, correct
20:54.03[TK]D-FenderDavey|Work : repastebin his config
20:54.16[TK]D-FenderTrazz : then you picked the wrong folder in musiconhold.conf
20:54.38[TK]D-Fenderfugitivo : www.atacomm.com has the lowest price I've seen anywhere
20:54.43argentasTrazz: you'll need to edit the path in etc/asterisk/musiconhold.conf
20:54.46Trazzhehehe i just found that in my grep for mohmp3 :)
20:54.48Davey|Work[TK]D-Fender, http://pastebin.com/508712
20:54.50Trazzthanks
20:54.55crich1999Katonka_: you'll need to either make samples again or just copy asterisk03/configs/misdn.conf.samples /etc/asterisk/
20:54.58fugitivo[TK]D-Fender: thanks
20:55.33Katonka_crich1999, rgr
20:55.45[TK]D-FenderDavey|Work : pastebin the [sip] context in extensions.conf
20:56.42Davey|Workuh, I don't appear to have one.
20:56.53[TK]D-FenderDavey|Work : that would be a rather severe problem then :)
20:57.08Davey|Workbut we have like 6-8 phones that work just fine :)
20:57.14Davey|Workso uhm. eck.
20:57.48[TK]D-FenderDavey|Work : I think you should take a closer look for aminute....
20:58.21Davey|Work[TK]D-Fender, looking :)
20:58.46*** part/#asterisk rue_work (n=not@h24-207-96-50.cst.dccnet.com)
20:58.54Davey|Workgrep -R "\[sip\]" ./ in the asterisk config dir brings up nothing. eeek
20:59.14Trazzargentas, ok thats fixed but still when i call i get dead air when connected..
21:00.07*** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk)
21:00.30*** join/#asterisk areski (n=areski@156.Red-83-44-65.dynamicIP.rima-tde.net)
21:00.32CurusIn the asterisk realtime voicemail table, what is customer_id used for?
21:00.46argentasTrazz: ok, now check whether 'ps ax | grep mpg123' shows anything whilst you are listening to the dead are
21:01.03jpabloanyone has experience using unicall ?
21:01.27Davey|Work[TK]D-Fender, OK, so essentially this is entirely b0rked and the fact any of our phones work is entirely a fluke?
21:01.31Trazz14203 ?        S      0:03 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
21:01.31Trazz14208 ?        S      0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
21:02.09[TK]D-FenderDavey|Work : Dunno, pastebin everything and maybe I'll notice something you're not...
21:02.44Davey|Work[TK]D-Fender, thanks, but well, I don't know what half of this stuff is and I don't want to reveal anything I shouldn't :)
21:03.59TrazzArgentas, i actually hear a tiny bit of music for a split second when i first connect and then dead air the rest
21:03.59Katonka_misdn.conf ports are my cards or the channels of the isdn cards... like 1-3 card 1 and 4-6 card 2 (2 B-Chan, 1 D-Chan)
21:04.05*** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net)
21:04.29*** join/#asterisk klictel (n=klictel@207.107.208.137)
21:05.10argentasTrazz: hmm, you're not maybe starting to play the moh then hanging up are you?
21:05.21Trazznope
21:05.22*** join/#asterisk [hC-] (n=hardcore@209.153.195.139)
21:05.25Trazzits connected the whole time
21:05.30argentask
21:05.39Trazzexten => 5000,1,Answer
21:05.40Trazzexten => 5000,2,MusicOnHold
21:05.42Trazzthats teh test exten
21:07.12argentasok, add 'exten => 5000,3,Wait(10)' to that
21:07.29*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
21:07.54Trazzsame thing
21:07.55[TK]D-FenderDavey|Work : Sure, whatever :)
21:08.17CurusHmm maybe it's telling that a recursive grep for customer_id in the * source comes up empty
21:08.19argentasok, couldn't remember where MusicOnHold blocks or backgrounds
21:08.34CurusI wonder if deleting the column will work
21:09.34crich1999Katonka_, what means rgr ?
21:09.39argentasTrazz: came into this part way through, have you already put the console output in pastebin?
21:09.54jpablocrap, i have a problem with unicall, when i search in google for the error i find a paste bin: http://fr.pastebin.ca/comments.php?bin_id=14873
21:10.07jpablothat's my problem, anyone knows how to fix that ?
21:10.08Trazznot yet.. do i need to turn on something first?
21:10.11Katonka_crich1999, Roger=I will do so.
21:10.17Davey|Work[TK]D-Fender, you have been most helpful, but really, why does this have to be so freaking complex? Heh
21:10.20Katonka_crich
21:10.35Davey|Workwhere the GUI dammit? :D
21:10.36crich1999Katonka_, i see
21:10.49argentasTrazz: 'set verbose 20' first.. and we'll see what that shows us
21:10.51Zodiacalanyone know the # to call for verizon lines that tell you what number your calling from?
21:11.07Trazzok
21:11.42Katonka_crich1999, but i am confused with the ports. i got my 2 hfc cards found by misdn-init config . now in misdn.conf i have to choose port 1 for extern and port 2 for intern?
21:12.12Trazzgonna pastebin now
21:12.36SwK[Work]anyone polycom certified around?
21:13.06Trazzargentas - http://pastebin.com/508749
21:13.15Trazzi am still connected btw
21:13.46crich1999Katonka_, you can choose the section name like you wish. It's just a sample, name it like you think its best
21:14.01Trazzargentas, this is after i manually hang up http://pastebin.com/508751
21:14.36Katonka_crich1999, but i got 2 ports or 6 with 2 isdn-cards?
21:15.53crich1999Katonka_, you got 2 ports with 2 hfcpci based cards
21:16.09[TK]D-FenderDavey|Work : this is not complex.... trust me.. its peanuts...
21:16.15argentasTrazz: taking a look now...
21:16.18*** join/#asterisk clive- (n=pirch@dsl-165-158-250.telkomadsl.co.za)
21:16.19[TK]D-FenderSwK : Whats your question?
21:16.21Trazzok
21:17.34Davey|Work[TK]D-Fender, heh, you apparently have a lot more time on your hands than I :)
21:17.56Katonka_crich1999, that misdn.conf is used by asterisk after a reboot? or how is asterisk now using my hfc-cards? is zaptel and bristuff still used when misdn is installed? are you speaking german? ^
21:17.56[TK]D-FenderDavey|Work : I jsut read the WIKI.... and I've been playing around with * for almost 2 years....
21:18.10[TK]D-FenderDavey|Work : Books aren't worth much....
21:18.15Davey|Workyeah, see that 1 years 11 months and 2 weeks more than me ;)
21:18.30[TK]D-FenderDavey|Work : You need better samples and a decent teacher...
21:18.30crich1999Katonka_, yes i do
21:18.40Davey|Work[TK]D-Fender, want a job? LOL
21:18.46SkramXlol
21:18.47[TK]D-FenderI can provide both at a very reasonable price! ;)
21:19.05crich1999Katonka_, join asterisk-misdn
21:19.12SkramXGod, I had a AOL employee approach me for Asterisk consulting last night.
21:19.14*** join/#asterisk MatsK (n=mk@3.80-203-81.nextgentel.com)
21:19.26argentasTrazz: is it definately a moh issue, what happens if you replace MusicOnHold with for example 'SayDigits(123456789)' ?
21:19.31Davey|WorkSkramX, *For* AOL?
21:19.37SkramXDavey|Work: Maybe..
21:19.54SkramXLOL, no.. it was for a company he wanted to start himself.
21:20.11DrukenAOL is a pain in the ass
21:20.27Trazzya its says the numbers
21:20.48SwK[Work][TK]D-Fender is about their certification program...
21:20.51SkramXThis employee wanted consulting for about 10 dollars an hour
21:20.52SkramXOh well.
21:20.58SkramX<-- Not dirt.
21:21.21Davey|Work$10/hr? wow, thats like... crap
21:21.29Davey|Workmy wife for more than that answering phones for UPS.
21:21.32Davey|Workquite a bit more
21:21.38SkramXYeah
21:21.39SkramXheh
21:21.47SkramXSpeaking of UPS.. im waiting for my package!
21:22.14shmaltzanybody ever heard of stratasoft?
21:22.27jdv79what is brown doing for you?
21:22.36jdv79i think they're off today - somme of them
21:22.54Davey|WorkI got an unexpected package in the mail today, I heard when Ic alled my wife at lunch, she opens it up and I get all excited... turns out my mom had just ordered some stuff she couldn't get shipped to the UK and wanted it to go via me first. Hinges.
21:23.10Davey|WorkSkramX, its a bank holiday, they may be off
21:23.16Trazzargentas, it says the numbers ok
21:23.35SkramXhmm
21:23.51SkramXim gonna jet for a bit
21:23.54*** join/#asterisk lesouvage (n=lesouvag@82.74.11.143)
21:24.00SkramXtootles, feel free to pm me for whatever
21:25.08znoGwhats the best way, from an AGI script, to find out of a particular EXTEN has a voicemail configured_
21:25.11znoG?
21:25.35znoGrunning "show voicemail users" and doing a grep on that would work, but from an AGI script how can I run such command?
21:25.46znoGunless I do an asterisk -rx and run it that way
21:26.08[TK]D-FenderznoG : you can run CPI commands from AGI direct....
21:26.29znoGhow so?
21:27.41[TK]D-FenderCLI*
21:27.52[TK]D-FenderznoG : Don't recall the exact command, but its there...
21:28.02jbalcombhrmm.. significant lack of information specific to 'Unable to handle indication 3'
21:28.13[TK]D-FenderznoG : you can also parse voicemail.conf direct as well..
21:28.27TrazzTk, any ideas on why my music on hold would be broken?
21:28.30znoG[TK]D-Fender: yea, could do that too. Was hoping via AGI
21:29.59*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
21:30.50[TK]D-FenderznoG : znoG : You can use either a System call, or AMI, or parse the file direct.  take your pick.  I use AMI on my Polycom Idle scripts for queue stats & voicemail.
21:31.20znoGAMI?
21:31.28znoGoh
21:31.31znoGasterisk manager interface
21:31.51[TK]D-Fenderyup, ok, gtg, back later
21:32.02*** join/#asterisk oogle_ (n=oogle@63.215.127.17)
21:33.26*** join/#asterisk nicknick (n=nicknick@81-86-107-241.dsl.pipex.com)
21:33.37oogle_i'm having a big problem parking sip calls with moh-native.  I initiate the transfer to 700 and the person being parked starts hearing music on hold, then when the channel is parked, the music on hold stops and starts again really loud and distorted
21:33.45*** join/#asterisk wulf814 (n=lorentz@216.48.0.4)
21:34.59Trazzargentas you there ?
21:35.49Trazzcan someone review this http://pastebin.com/508809
21:35.51wulfy814ok, silly question I just changed ISP's at the office
21:36.00wulfy814and my GS2000 won't register
21:36.04wulfy814with my * at home
21:36.13wulfy814it is still showing the old IP from the old ISP
21:36.27wulfy814how do I kill it? so that it can register from the new ISP
21:42.22*** join/#asterisk a1fa (n=a1fa@207.210.210.202)
21:42.23a1fahi
21:42.57*** join/#asterisk devel (n=devel@wiggum.digitalcoven.com)
21:43.45a1faon 1.2.1
21:44.39justinufunny, i named a computer wiggum also
21:45.30a1fasue them
21:46.01a1fatrademark infirngement
21:46.03a1fa!!!!
21:46.05a1fayo yo yo
21:46.09a1faz;)
21:46.30develcool, justinu
21:47.46a1faaiiiighhhhhhhhhhht
21:48.11*** part/#asterisk oogle_ (n=oogle@63.215.127.17)
21:48.28*** join/#asterisk gml (n=rm@66.193.229.9)
21:49.41gmlhey when i do "zap show channel 1" it says "Dialing: no" and i get Don't know what to do w
21:49.41gmlntrol frame 15
21:49.41infinity1can someone familiar with polycoms tell me the point of the server tab in the web UI when there is a server associated with each configurable line?
21:49.41gmldon't know what to do with control frame 15
21:49.46gmlis that the CO side or is that me?
21:50.47justinuinfinity: in case the line doesn't have a server associated with it, it'll fall back to the "generic" server section
21:51.20lesouvagetazz: check your /etc/asterisk musiconhold.conf. There shoudl be a class default under [classes]
21:52.02a1falol sucks that asterisk wants termcap
21:52.25*** part/#asterisk RussCC (n=face@216.157.205.211)
21:52.38infinity1justinu: ic thanks :)
21:53.53lesouvagegml: I have the same while calling over the internet. I guess it has to do with the provider.
21:54.23infinity1justinu: should register be set to Yes for the default server? i'm don't follow the significance. some poeple say set it to no.... confusing
21:56.01sivanaanyone here in 902 area code?
21:56.03VyepermanDoes anyone have a recommendation for a cheap ip phone that I can throw in all of kids rooms, I have two nice ones for my business, but I'd like to be able to stick on in each room, and I'm not willing to pay $160 a piece for it.
21:56.37IkaruszszszszVyeperman: budgetone
21:56.58justinuinfinity: register=0 means that the phone can call out even if it can't register with a proxy
21:57.15justinuregister=1 means that if there are no succesful registrations, it won't allow outbound calls
21:57.23*** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)
21:57.29Trazzany good information on how to build IVR ?
21:58.15infinity1justinu: is all this documented someone? or are you the only one that knows? heh
21:58.21clive-trazz, use asterisk
21:58.23infinity1er s/someone/somewhere/
21:58.30clive-:0
21:58.32jarrodhmm
21:58.41jarrodon all my calls im getting this 'ssssshhhh'
21:58.44jarrod'sssshhh'
21:58.53Trazzclive- yes i am messing with it now. i would like to see some samples configs / examples on how to build IVR
21:59.00justinuinfinity: it's in the polycom admin guide.
21:59.05clive-trazz check the wiki
21:59.23VyepermanIkaruszszszsz, thnx
21:59.24justinuafk
21:59.55Ariel_Trazz, there is a great sample you can fallow of an ivr in your setup now. it's located at /usr/src/asterisk/configs/extensions.conf.sample
21:59.56infinity1justinu: i swear i searched the admin guide for the word 'register' and didn't find anything good. i'll look again.
22:00.48TrazzAriel, great your back..i can't music on hold working
22:01.30*** join/#asterisk libila (n=vye@ip68-8-174-154.sd.sd.cox.net)
22:01.38infinity1justinu: yer right. i just had to dig harder.
22:02.00*** join/#asterisk franck (n=franck@tikiwiki/franck)
22:02.01Ariel_Trazz, how did you configure the moh
22:02.08*** join/#asterisk skamp|afk (i=skambar@p5484080E.dip0.t-ipconnect.de)
22:02.13Trazzi dl it , make linux and then make install
22:02.17blitz[class]tzanger: oh happy day... Canada is moving right wing on Jan 23rd it sounds like :(
22:02.34Beirdobah
22:02.34libilaI have 3 phone lines, and probably about 8-10 ip phones. Which card would fit my needs best?
22:02.41Beirdowe shall see, blitz[class]
22:02.57Ariel_right wing left wing.. just a bunch of bs if you really think about it..
22:03.02Beirdoand if we get a change of lying scum, that's a good thing in my mind, time for new lying scum
22:03.05*** join/#asterisk Umaro (n=umaro@68.142.142.105)
22:03.14Umarohey guys.. anyone here using asterisk with telasip?
22:03.19jarrodwhy do calls thru asterisk suck
22:03.24TrazzAriel, i dl it , make linux and then make install, when i dial that extension i get no music
22:03.36*** join/#asterisk saitech (n=admin@85.235.237.14)
22:03.47Trazzyet it says its playing with verbose on 20
22:03.58tzangerblitz[class]: ?
22:04.01Ariel_Trazz, did you install mpg123 or what are you using for moh
22:04.12Trazzyes i installed mpg123
22:04.23tzangerI'm voting green, I've had enough of the liberals, even though they're the only financially sound choice.  the PC will never get my vote and hte NDP's a joke
22:04.43Beirdooh, and blitz[class], you can't blame me this time, I voted in advanced polls...  for the pinko bastard NDP as they actually have a real good chance of taking my riding.
22:05.08saitechi'm having a problem on my asterisk production box. It keep spams we with "chan_sip.c: failed to grab lock... trying again" in my debug log. The actually problems is, that it makes my box hang, while this debug is posted in the log.Can anyone help, or does anyone identify this problem ?
22:05.14BeirdoLiberals financially sound?  have you slept through the scandals, tzanger? :)
22:05.24Ariel_wow ndp, pinko's, green sounds like fun games for kids
22:05.52Beirdovoting Green Party's a good thing :)
22:06.02tzangerBeirdo: no.  what I am saying is that they are *excellent* at turning a country around.  However they take all that surplus and pocket it.  That's the downside.  The PC or NDP could never do what the libs did in terms of fiscal responsibility on a national level.
22:06.14Beirdoheh
22:06.19Beirdodunno about that
22:06.24tzangerso I figure it's time to let some fresh blood in and if they fuck up the country, vote the liberals back in to clean up but get 'em out before they keep all the money
22:06.32Beirdoheh
22:06.36Beirdosounds like a plan
22:06.38jarrodanyone had a problem with receive 'hiss' noises?
22:06.39*** join/#asterisk rue_work (n=not@h24-207-96-50.cst.dccnet.com)
22:06.52TrazzAriel, when i dial my extension i get a split second of medicine and then silence
22:06.53rue_workquick!? how do I find out how many active calls are on a machine????
22:07.02jbalcombjarrod I am having troubles with hiss, static, and echo across the board
22:07.11jarrodjbalcomb: asterisk 1.2.1?
22:07.13rue_workI think our pstn machine jsut went fooie, that or all our lines are in use...
22:07.25tzangerbut no I'm voting green because I want to get fresh blood and new ideas in.  They haven't got a hope of getting majority so it's "safe" and if I can get ONE seat green then I'm hoping ot see the greens at the next debates and really see what they can do.  They are hellishly disorganized but let's give their crackpot ideas some voice and see what happens
22:07.30tzangerjarrod: no not me
22:07.42Beirdohehe, cool
22:07.54Beirdowonder if the Marijuana Party will land a seat?
22:08.02rue_workANYONE!?
22:08.16blitz[class]rue_work: how about 'show channels'
22:08.30tzangerrue_work: nobody can tell but you
22:08.47Trazzbrb
22:09.11rue_workshow channels might be right....
22:09.29jarrodasterisk quality is sucking
22:09.31rue_workI need to know if I need to reboot the pstn mahine or if the lines really are all used up
22:09.54tzangerrue_work: show channels
22:10.00rue_workhoooo, looks like we just used up all our lines
22:10.03blitz[class]well... the PC scares the hell outta me... and I'd rather the Liberals get back in since everyone should be watching exactly what they are doing -- would be pretty hard to steal money I would hope
22:10.06rue_workthat freaked me
22:10.19tzangerblitz[class]: nah it's as easy as ever
22:10.27saitechi'm having a problem on my asterisk production box. It keep spams we with "chan_sip.c: failed to grab lock... trying again" in my debug log. The actually problems is, that it makes my box hang, while this debug is posted in the log.Can anyone help, or does anyone identify this problem ?
22:10.56blitz[class]tzanger: although I'm not too impressed with the Liberals trying to make handguns illegal -- criminals don't register their guns
22:11.20a1fablah
22:11.23blitz[class]you know what... if you think the last minority gov't was a joke, this time around is going to be even worse
22:11.42Umarono one here is using telasip? :/
22:11.48Beirdothis time will be FUN, blitz[class]
22:12.04BeirdoI hope we get NDP minority so they have to bend over for everyone to get anything done
22:12.06tzangerblitz[class]: yeah that's a fool's errand used to placate the old and dumb
22:12.14tzangerI LIKE minority governments
22:12.20blitz[class]tzanger: ditto
22:12.24tzangerbut I fucking hate the bullshit tactics that the conservatives did to end it
22:12.25JohnnieHmmm...
22:12.29Beirdome too, they tend to be more accountable
22:12.30bkw_saitech, those are bad messages.. I can do a literal translation of the message if you lilke?
22:12.31Beirdobah
22:12.31tzangerfucking waste my fucking money just ot get the exact same thing
22:12.40JohnnieAnyone here intimately familiar with zaptel sources and ztdummy?
22:12.43blitz[class]tzanger: if the PC's get a majority gov't I might as well just move to the US :)
22:12.51Beirdoand bah again
22:12.57sivanatzanger: they all suck and just fall into the same rut, year after year
22:13.01blitz[class]tzanger: actually... I'm considering moving to SF some August
22:13.01Beirdoand I will be anywyas
22:13.04tzangerblitz[class]: you won't have to, they basically said they would become a state
22:13.12blitz[class]tzanger: thats true
22:13.15tzangersivana: which is why, like diapers, government needs frequent changing
22:13.28tzangerI want to get rid of the two-party system
22:13.32bkw_too bad it never happns
22:13.41bkw_do they make Governmental Drano?
22:13.42Beirdowe don't have a 2-party system, silly
22:13.43saitechbkw_ i have looked in the code, and it seems like its trying to retrieve some owner-lock from a sip call. but failed all the time. im not sure, but it seems that the lock is the header .
22:13.44tzangerI fully expect if the PC win for Harper to unzip his skin and Preston Manning to jump out
22:13.47Beirdowe have a 4-party system
22:13.52tzangerBeirdo: yeah right.
22:13.56*** join/#asterisk Jzalae (n=sk@dsl-66-63-110-48.gwi.net)
22:14.03sivanatzanger: I'm voting green party, what else do we have to lose
22:14.06Beirdoand Preston Manning has a lot more integrity than Harper :)
22:14.08saitechbkw_ if you have a better perspective i would be glad to hear so
22:14.12tzangersivana: exactly :-)
22:14.27libilaDoes anyone have anything against this interface card: http://www.voicetronix.com.au/openpci.htm  I have 8-10 ip phones & 3 phone lines I want to hook up, and of course I'll be using asterisk(FreeBSD).
22:14.34Beirdoor Stockwell Day.  HE was a joke
22:14.39tzangerhaha
22:15.00tzangerthe PC are jokes... they don't have a platform nor a personality... they are the "we are not the liberals" party
22:15.06tzangerthat's their entire platform
22:15.11Beirdonot true
22:15.18tzangerand the NDP is "we'll give money to everyone and everything to get in" party
22:15.24blitz[class]Kim Campbell was my hero
22:15.28Beirdothey are just chosing to not state any policy to not piss people off
22:15.42tzangerBeirdo: which indicates to me that they have no platform
22:15.47Beirdono
22:15.49tzangerif you're not gonna tell me then why the fuck would I vote you in
22:15.50Beirdothey have one
22:15.56sivanait's just not public
22:16.04Beirdobut one that many people who are sitting on the fence may not like
22:16.09blitz[class]"vote us in and we'll tell you what we're going to do after we have already done it"
22:16.15Beirdoand one that half the people IN the party won't like
22:16.21sivanahehe
22:16.25blitz[class]Beirdo: thats terrible, irresponsible politics
22:16.41Beirdomore honest than the Liberals who promise piles of stuff and don't do any of it :)
22:17.00blitz[class]I hate consevative values
22:17.03*** join/#asterisk SarahEmm (n=sarahemm@MTL-HSE-ppp159791.qc.sympatico.ca)
22:17.11Beirdowe still have the GST that Chretien promised to get rid of
22:17.15blitz[class]I'm pretty much pro everything they are against
22:17.39blitz[class]Beirdo: whatever... you replace the GST with higher income tax -- no matter what happens you still don't get that money
22:17.45blitz[class]at least with GST I can choose not to buy stuff
22:17.46SarahEmmhiya
22:17.58*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)
22:18.10blitz[class]SarahEmm: hoi-oh
22:18.17BeirdoSarahEmm: ahoy!
22:18.23sivanaSarahEmm: yo
22:18.32SarahEmmhi :)
22:18.34Beirdoevil pirate wench :)
22:18.36Beirdoheh
22:18.40SarahEmm:)
22:18.46*** part/#asterisk clive- (n=pirch@dsl-165-158-250.telkomadsl.co.za)
22:18.51Beirdowe STILL spend too much time in htat damn game
22:19.02Beirdocome buy some black cloth from us on DN
22:19.04Beirdoheh
22:19.07tzangerI'm addicted to Civ4
22:19.14blitz[class]thank goodness I don't play games or I'd never get anything done
22:19.16*** join/#asterisk Naturalblue (n=Kay@195.26.12.229)
22:19.17SarahEmmDN is home for me :)
22:19.20tzangerblitz[class]: that's what I said too
22:19.27Beirdome too, SarahEmm.
22:19.32SarahEmmanyone remember who the one here working on TTY/TDD stuff is, other than me?
22:19.38bsdfreakHEH
22:19.40Beirdosorry, not off hand
22:20.22Beirdowell, I should go home anyways.  got some cleaning, puzzle pirates, etc to do :)
22:20.24saitechbkw_: what would you translate the message into ?
22:21.09Beirdoseeya all.  Don't forget to vote, regardless of which lying scum ya vote for :)
22:21.13Beirdohehe
22:21.25*** join/#asterisk x[Girl] (n=FunGirl_@stjhnf0112w-142163115151.pppoe-dynamic.nl.aliant.net)
22:22.07SarahEmmheh :)
22:24.00*** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com)
22:25.56*** join/#asterisk devel (n=devel@wiggum.digitalcoven.com)
22:26.00gmlhey i'm trying to dial out on a Zap interface and i keep getting "Unallocated (unassigned) number" back
22:27.27*** join/#asterisk cpm (n=Chip@68-66-23-191.chvlva.adelphia.net)
22:28.11infinity1how can you make the polycom register on using a random poort? when i do 'sip show peers' its always 5060
22:28.31SarahEmmwhy would you want to, infinity1?
22:28.58libilahttp://www.voicetronix.com.au/openpci.htm that interface card says Bus Type: PCI2.2, is that the common 32-bit PCI bus?
22:29.08infinity1SarahEmm: because i have multiple polycoms behind nat with asterisk on the other side
22:29.33SarahEmmlibila: one of the common types. your slots could be 1.x or 2.x...
22:29.40SarahEmminfinity1: ahhh...
22:29.50SarahEmminfinity1: the source port is 5060 you mean?
22:30.39libilaSarahEmm: any easy way to tell short of looking up the mobo?
22:30.48SarahEmmdoes the mobo have ISA slots too?
22:30.56*** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt)
22:31.06*** part/#asterisk ping1 (n=DLBaker@67-133-167-72.dia.cust.qwest.net)
22:31.24infinity1SarahEmm: yes. i'm assuming thats what is shown in sip show peers. all the other phones are random
22:31.29SarahEmmahh..
22:31.35libilaSarahEmm: I don't believe so, it's not that old.
22:31.45infinity1SarahEmm: except the polycom.
22:31.49SarahEmmokay. it's likely 2.2 then, not sure how you can be sure.
22:31.59sivanaSarahEmm: you aren't registered to my server
22:32.47*** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
22:32.48SarahEmmsivana: my box is down with a hardware failure right now
22:32.54sivanaoh ok :)
22:32.55libilaSarahEmm: if you had 3 phone lines to connect to 8-10 ip phones running asterisk on FBSD would you buy this card? http://www.voicetronix.com.au/openpci.htm
22:33.12*** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com)
22:33.38blitz[class]SarahEmm: unfortunately there isn't really any easy way to tell if its 2.2 or 2.1 unless you check the MB manual
22:33.53SarahEmmi'm not sure libila, i don't have much experience with analog fxs/fxo cards :)
22:34.08libilaSarahEmm: Neither do I, and I don't have a clue.
22:34.13blitz[class]fxs/fxo....ewww :)
22:34.19blitz[class]analog is gross :D
22:34.25SarahEmmi'd go with something a bit better supported than voicetronix.. they need exsternal binary-only drivers, no?
22:34.32libilablitz[class]: how else do you hook it up to your phone lines?
22:34.44infinity1justinu: how do you make the polycom use a random source port, instead of the default 5060?
22:36.28Augheylibila: I'd suggest http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P
22:36.46SarahEmmanyone here familiar with the internals of fsckmodem.c?
22:36.48SarahEmmfskmodem.c rather
22:37.11*** join/#asterisk saftsack (n=saftsack@p54A7F6F8.dip.t-dialin.net)
22:37.16saftsackhi
22:39.12MstlyHrmlsinfinity1: volpProt.local.port
22:39.22MstlyHrmlsinfinity1: in sip.cfg
22:39.35infinity1MstlyHrmls: thanks!
22:40.15infinity1MstlyHrmls: it says if you put 0, it still uses 5060.
22:40.20*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
22:40.36MstlyHrmlsinfinity1: yes, the Polycoms don't do the random source address thing
22:40.43*** join/#asterisk pigpen2 (n=mark@fw.seamans.cc)
22:40.51infinity1MstlyHrmls: hm. what do you suggest?
22:40.55MstlyHrmlsinfinity1: you'll have to assign a unique port to each phone behind the NAT
22:41.30infinity1:/ ..k
22:42.10*** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com)
22:42.13*** join/#asterisk Flauto (n=zhao@c-71-194-194-48.hsd1.il.comcast.net)
22:42.16libilaAughey: Alright, well since I don't know what FXS is I don't think I'll need it, I'll probably get the TDM04B package.
22:42.18ManxPowerWHERE does that myth come from????
22:42.25Flautohello everyone
22:42.28ManxPowerThere is no reason to assign each phone a different port.
22:42.29Flautohow you guys doing
22:42.35saftsacksome iaxmodem experts here?
22:42.40ManxPowerUnless your NAT router sucks, of course.
22:43.18*** part/#asterisk a1fa (n=a1fa@207.210.210.202)
22:43.20Flautois there anyone who can help me with a2billing?
22:43.24Flautoi tried a couple of times
22:43.28Flautobut did not work out
22:43.39ManxPowerFlauto, Sorry, I don't bill for calls.
22:43.46Flautohehe
22:43.48ManxPower~fxofxs
22:43.51jbotwell, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this.  An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this.
22:43.51Flautomanxpower
22:43.52SarahEmmManxPower: meow!
22:43.53Flautoi dont' either indeed
22:43.58Flautobut i just wnat to play with it
22:44.05ManxPowerhellop, SarahEmm
22:44.11*** join/#asterisk |omni| (n=rob@net98.limelyte.net)
22:45.42infinity1argh. when using the mac to assin the phones names in sip.conf, you can't use the default buttons to connect to voicemail. hmm
22:45.55infinity1if its not one thing, its another. hah
22:48.14dpryoSomebody know how to use the feature buttons on Avaya 4620, under asterisk/sip?
22:48.27jarrodi get hiss noises on all channels, zap, sip, mgcp
22:50.15*** join/#asterisk Katonka (n=Miranda@p54BEE46D.dip.t-dialin.net)
22:50.29libilaif you just have ip phones but your using a normal phone line you jus need fxo ports right?
22:50.39ManxPowerinfinity1, Huh?
22:50.41fugitivo~fxsfxo
22:50.42jboti heard fxsfxo is An FXO port expects to receive dialtone and receive ring voltage.  An FXS port expects to provide dialtone and provide ring voltage.
22:50.44ManxPowerinfinity1, Um, what phone?
22:51.37libilak so you do need both...
22:51.50fugitivolibila: no
22:52.03fugitivolibila: you already have ip phones
22:52.21*** join/#asterisk zotz (n=zotz@24.231.47.175)
22:52.22fugitivolibila: you only need fxo ports to connect the line
22:52.45*** join/#asterisk HuSeyiN (n=_Yalniz-@stjhnf0112w-142163117148.pppoe-dynamic.nl.aliant.net)
22:53.16*** join/#asterisk Strom_C (n=strom@adsl-69-221-57-46.dsl.chcgil.ameritech.net)
22:53.23Lotswhatd the diff between asterisk@home and regular asterisk?
22:53.40fugitivoLots: well
22:53.49ManxPowerLots, Asterisk@HOME is some lame ass GUI that people like.
22:53.53fugitivoLots: asterisk@home is a complete distribution, that includes asterisk
22:54.06fugitivoManxPower: no, it's not a gui
22:54.12fugitivoamp is a gui
22:54.20ManxPowerIn nay case, we don't really talk about it here.
22:54.24ManxPower~amp
22:54.25jbotsomebody said amp was NOT supported here! people using it should join #amportal
22:54.39fugitivoLots: asterisk@home is a distro that includes linux + asterisk + amp (gui) and it sucks
22:55.05fugitivoLots: regular asterisk is the way to go
22:55.31fugitivoLots: you pick the distro you like more, compile asterisk from source and configure it using plain text files
22:55.43*** part/#asterisk l1nux (i=moi@214.138.103-84.rev.gaoland.net)
22:55.56saftsackfugitivo, hi, do you know iaxmodem?
22:56.03Lotsfug thats where i'm at right now, running the most recent asterisk on debian
22:56.14Lotsbut reading this article about asterisk@home.
22:56.50fugitivosaftsack: no
22:57.08infinity1ManxPower: polycom
22:57.20*** join/#asterisk saftsack (n=saftsack@p54A7F6F8.dip.t-dialin.net)
22:57.32fugitivoLots: asterisk@home is a try to bring a pbx to regular users, and "I" think, regular users shouldn't mess with a PBX
22:57.40Lotslol
22:57.48ManxPowerinfinity1, You're smarter than you look.  Using Polycoms and setting up the phones with their MAC as their sip.conf username.
22:57.49*** join/#asterisk Snooker (n=klayton@201.4.213.114)
22:58.07ManxPowerinfinity1, as a special bonus, do you want my config files?
22:58.17infinity1ManxPower: for sure!
22:58.25ManxPowerhold on.
22:59.33*** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com)
23:01.28ManxPowerinfinity1, I'll include my config files, one set of config files for one of my phones, as well as the sip and bootrom that I use.
23:01.44infinity1ManxPower: k. email? dcc?
23:01.55ManxPowerinfinity1, I'll put it on a web site.
23:02.05infinity1ok. thanks!
23:02.20ManxPowerwww.fnords.org/~eric/tmp/poly.tar.gz
23:02.25infinity1for my current problem, i think setting a "callback attribute" is necessary.
23:02.28ManxPowerit's about 11MB with all the polycom stuff
23:03.38ManxPowerinfinity1, most of my users are too stupid to try anything fancy like pressing the voicemail key.
23:03.57infinity1hah
23:04.09crich1999do the Polycom phones allow Sendtext ?
23:04.19ManxPowercrich1999, not when I tried it.
23:04.36ManxPowerinfinity1, one of my users actually told me they don't use text messaging because "it's too complicated"
23:04.45*** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com)
23:04.55crich1999ManxPower, hm .. the Snoms do, but you need to press the Message button to retrieve the text info, which really sucks
23:05.19crich1999i search a nice sip phone which directly displays the text sended by sendtext
23:05.52umayi can't find one of those, either
23:05.57Nuggetmake one that has backlight and I'll order 100.
23:06.05crich1999hehe
23:06.08SarahEmmhehe
23:06.29umayADSI allowed for it ages ago
23:06.31Darwin35ast_load became ast_conf_load right
23:06.41crich1999isdn phones can do that  too
23:08.06*** join/#asterisk ToTo (n=ToTo@host16-146.pool872.interbusiness.it)
23:08.28*** join/#asterisk backblue (n=moo@87-196-15-214.net.novis.pt)
23:10.10Darwin35did ast_load  become ast_conf_load in odbc
23:10.53Darwin35trying to get dbodbc to work
23:12.56*** join/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com)
23:13.03darwin_35afetrnoon all
23:13.35darwin_35<PROTECTED>
23:13.36darwin_35Jan 16 16:13:41 WARNING[94919]: loader.c:554 load_modules: Loading module app_dbodbc.so failed!
23:13.43darwin_35neeed help with this error
23:14.26file[laptop]I've been over this before
23:14.31file[laptop]use ast_config_load
23:14.50infinity1ManxPower: so 3509 is your voicemail app?
23:15.27darwin_35there where 2 lines to change I could not find them
23:15.34darwin_35File thats why I asked
23:16.38file[laptop]well, you'll have to find them... they're in app_dbodbc's source code
23:16.39darwin_35what whas the destroy line file ?
23:16.57file[laptop]ast_config_destroy?""
23:17.19saitechi'm having a problem on my asterisk production box. It keep spams we with "chan_sip.c: failed to grab lock... trying again" in my debug log. The actually problems is, that it makes my box hang, while this debug is posted in the log.Can anyone help, or does anyone identify this problem ?
23:17.26ManxPowerinfinity1, correct
23:17.45*** join/#asterisk Jaxx[18f] (n=SiGn@stjhnf0112w-142162207029.pppoe-dynamic.nl.aliant.net)
23:18.18*** join/#asterisk bjohnson (n=bjohnson@i216-58-48-187.cybersurf.com)
23:19.13Zodiacalanyone know the advantage asterisk can give you by using a fax machine on an fxs port? insted of just wiring a fax machine to the phone line directly?
23:20.02darwin_35thanks file
23:20.12darwin_35I will treat you to dinner
23:21.02Skumlinghumm, I just had a very odd problem with *... when trying to call out using zap/g1 it kept failing with "Channel 0/1, span 1 got hangup, cause 42" - I was able to ring in to asterisk via the same zap-interface using my cellphone
23:21.37*** join/#asterisk lesouvage (n=lesouvag@82.74.11.143)
23:22.28Skumlingrestart/reload of asterisk including ztcfg -vv didn't help... then I tried changing extensions.conf to zap/1 and it worked... also it worked with zap/2, and when changing back to zap/g1 that also works - looks like something in the dynamic channel allocation screwed up?
23:22.46darwin_35it worked
23:24.08*** part/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com)
23:24.19*** join/#asterisk corposant1 (i=corpo@dyn-shp-225-67.dyn.columbia.edu)
23:24.57SkumlingIf I should hook up a SIP-phone to Asterisk, would the Grandstream GXP-2000 be a decent choice?
23:27.27corposant1anybody here have a moment to give me some input on an asterisk issue?
23:27.55corposant1i'm having trouble with a phone being "UNREACHABLE" from the asterisk server, but it's able to make calls
23:28.01corposant1just not receive them
23:28.22SkumlingiDunno: humm okay, looks cool, but pricy too...
23:28.24Corydon-wcorposant1: SIP phone?
23:28.30corposant1yessir
23:28.36corposant1asterisk has a public ip
23:28.39corposant1as does the sip phone
23:28.40Corydon-wcorposant1: set qualify=no
23:28.42corposant1no nat
23:28.45corposant1ah...k
23:29.06Corydon-wcorposant1: the problem is that your SIP phone is not properly responding to an OPTIONS request from Asterisk
23:29.33Corydon-wIf it doesn't respond to OPTIONS, Asterisk sees it as unreachable
23:29.50*** join/#asterisk jpablo (n=jpablo@200.94.130.194)
23:29.52Corydon-wEven an error response to OPTIONS works... but no response is bad
23:30.03corposant1that was indeed it
23:30.07corposant1it's working
23:30.18corposant1thanks!
23:30.43corposant1next question would be how would I make it work if the phone were behind a nat?
23:31.09corposant1doesn't the qualify setting act as a keep-alive for the nat?
23:33.16*** join/#asterisk tmccrary (n=tmccrary@d47-69-35-227.try.wideopenwest.com)
23:36.03[av]banihttp://bani.anime.net/phonez/
23:36.37*** join/#asterisk areski (n=areski@192.Red-83-60-99.dynamicIP.rima-tde.net)
23:40.56Kattyhi.
23:42.15Kattythe vonage commercial is annoying.
23:42.51xachenVonage sucks
23:42.59xachenThey are much like Microsuck
23:44.14twisted[asteria]woohoo woo hoo hoo
23:44.17*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
23:44.44jdv79uhm, was it that good for you?
23:46.33Darwin35yes it all works
23:46.48Darwin35new fax machine rocks
23:46.55Darwin35this kicks ass
23:51.17Kattyoh boy, gateway media center!
23:51.21Kattyjust what i /always/ wanted!
23:51.28twisted[asteria]lol
23:51.49infinity1anyone have a polycom configured with something like voxee? asterisk is easy, but i added a button for voxee, and it doesn't work
23:52.19twisted[asteria]ow
23:52.24twisted[asteria]my eye!
23:52.57xachenfax over Voip? :S
23:53.06*** join/#asterisk m_a_g_o (i=maxgluck@201.243.103.247)
23:53.26*** part/#asterisk tmccrary (n=tmccrary@d47-69-35-227.try.wideopenwest.com)
23:53.29Kattytwisted[asteria]: i'd never poke you in the eye silly.
23:54.18twisted[asteria]Katty, mmkay.
23:54.57iDunnotwisted[asteria]: that should make you ask where Katty *is* poking you.
23:55.07Kattygosh.
23:57.05iDunnoowww.
23:57.11iDunnothat 'urt.
23:57.25iDunnowas that really neccessary?
23:57.56[av]banihow good is voxee?
23:58.48justinuyou aint' all that beautiful either
23:58.55justinuso what does that leave...
23:59.13Kattybe nice.
23:59.44justinuok, ok... it's just that I'm not all that attracted to him.
23:59.48twisted[asteria]justinu, you're quite the mirror breaker yourself from what I hear.
23:59.58Kattywoah

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