irclog2html for #asterisk on 20060714

00:00.43flynuxok, i'll have a look, thanks
00:03.38Dr-LinuxCunningPike: ever you tried sphinx?
00:03.59CunningPikeDr-Linux: No
00:04.10Dr-Linuxok
00:04.27*** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
00:04.54Dr-LinuxCunningPike: actually i'm trying to install sphinx voice recognition with asterisk .. but not sure if anyone already done.
00:05.36CunningPikeDr-Linux: Did you search the wiki?
00:05.55Dr-LinuxRoyK[uk]: thanks
00:05.57Un1x•Dr-Linux• fucker i asked u for help on msn
00:05.59Un1xand u didn't help :S
00:06.02Un1xdamn gayvision
00:06.23Dr-LinuxCunningPike: i didn't found any help
00:06.43CunningPikeUn1x: See below
00:06.48CunningPike~thebook
00:06.53jbotthebook is probably a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
00:06.54Un1x?
00:07.02CunningPike~thewiki
00:07.04jbotmethinks thewiki is at http://www.voip-info.org/wiki-Asterisk
00:07.05Un1xthis aint, for sipra lol
00:07.20CunningPikeUn1x: Sorry! Wrong person lol
00:07.26Un1xoh ok
00:07.30Un1xCunning pike
00:07.31CunningPikeUn1x: Do you have the admin guide?
00:07.34Un1xtell me soem voip providers
00:07.35Un1xi can get
00:07.52CunningPikeUn1x: I found that very useful for the SPA-3000
00:07.58Un1xno only the litrle booklet it came with has 4 pages 3 of wich are how to plug in wires lol and 1 of wich just tells u how to find out, wich ip address is being used etc
00:08.07CunningPikeUn1x: As for ITSPs, I have no idea - we use a PRI
00:08.24*** join/#asterisk eKo1 (n=bernd@190.4.7.90)
00:08.25CunningPikeUn1x: You can get the admin guide from the Sipura web site - it's well worth a read
00:08.32Un1xno Cunningpike i need a provider for my Spa-3000
00:08.34Un1xu dont know one?
00:09.00CunningPikeUn1x: Um - your SPA-3000 connects asterisk to a POTS line,,,,,,,
00:09.15*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
00:09.17Un1xno
00:09.19Un1xjust to the net
00:09.21Un1xno Asterisk
00:12.28Un1xman
00:12.31Un1xcrap no help here
00:12.46*** join/#asterisk test34- (n=test34@unaffiliated/test34)
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00:17.25flynuxCunningPike: ok, found out, there seems to be jumpers on the TE4XXP cards but setting is bad, fortunately it's overridable by software they give the solution there: http://www.asteriskguru.com/tutorials/wildcard_te405p_wildcard_te410p.html
00:18.42CunningPikeOK - great
00:18.55Un1xCunningPike you dont know any SIP providers?
00:19.08CunningPikeUn1x: No - we do direct to the PSTN
00:19.47knarflyUn1x: try myvoice.splitinfinity.com
00:20.00Un1xknarfly thanks
00:20.06Un1xanyone use it the spa-300 tho :P?
00:22.42knarflyUn1x: tell them user #49 sent you...they have DID's and their tech support is very * friendly.
00:23.04Un1xknarfly i wanna know that they suppora SIP meaning, Spa-3000 stuff..
00:23.08Un1xwell nvm i got my answer :P
00:23.19Un1xok thanks knarfly wanna help me a bit on configuring the spa as well?
00:23.50hadsUn1x: Try a Sipura support forum.
00:23.58Un1xok
00:24.01knarflyUn1x: I'll try...I'm still pretty new at this but I got several systems working so far
00:24.34knarflyUn1x: Yes http://myvoice.splitinfinity.com does sip
00:24.42Nivex<PROTECTED>
00:24.42Nivex`
00:24.56Un1xokay thanks :)
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00:28.12Un1xknarfly how long do spitfinity take to get ur account setup
00:28.14Un1xso u can use it?
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00:30.10knarflyUn1x: instant
00:30.20knarflyUn1x: instantly that is
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00:31.05knarflyUn1x: I was up in minutes...if you want DID you'll need to catch them during normal biz hours...they're in San Diego, CA
00:32.04Un1xwhy do they take soo long for the DID?
00:33.53knarflyUn1x: Can't say but they do have a new interface that looks like you might get DID without their help...I haven't tried it but their outgoing stuff works clear as a bell.
00:34.50knarflyUn1x: I have a toll-free DID with them. I just called them during regular biz hours and they set it up.
00:35.45knarflyUn1x: They have 24/7 tech support that's been a big help.
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00:45.24Un1xknarfly
00:45.27Un1xu there
00:45.42*** join/#asterisk DasTech (n=DasTech@c-67-176-28-65.hsd1.co.comcast.net)
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00:50.41Un1xsomone help man
00:50.46Un1xshit man
00:50.52Un1xknarfly coz of u i got charged 80$ :|
00:51.05Un1xfrom spitfinity but now i dont know how to use the info they gave me with, Spa-3000
00:51.05Un1x:S
00:53.08Un1xshit thanks to knarfly im fucked now lol
00:53.31riddleboxcan someone help me with this error? http://pastebin.ca/87497
00:53.52Strom_CUn1x: what did knarfly tell you to do?
00:54.01Un1xhe told me use spitfinity
00:54.05Un1xi got a account there
00:54.24Un1xthey charged me 80$ lol wichi dont care about
00:54.25Un1xbut now
00:54.25Un1xi dont know how to set it up with Spa-3000
00:54.25Un1xL:S
00:54.25Strom_Cdoes "lol"
00:54.30Strom_Cactually appear on your credit card bill?
00:55.32Un1xwhat?
00:55.39Un1xofc man if they charged me 80$ obviously
00:55.45Un1xthatsd 1 day of work gone to them :S
00:55.51Strom_Cwell you said they charged you "$80 lol"
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00:57.03Un1xoh no
00:57.14Un1xStrom_c comon man you use spitfinity with Sipra?
00:57.23Strom_CI've never heard of spitfinity
00:57.46Strom_Cbut I would personally be wary of any telco with "spit" in the name
00:58.13Un1xlol
00:58.21Un1xhttp://splitinfinity.com/aboutus_contactinfo.html
00:58.22Un1xsplit sorry
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01:01.59knarflyUn1x: sorry I missed that...was in another room... let's try it again. what was your question?
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01:02.16Un1xknarfly
01:02.24Un1xi need help setting up the Spitfinity with Spa-3000
01:02.24Un1x:S
01:02.28Un1xi dont know where what goes
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01:06.47Avochelmdoes anybody know how asterisk generates a uniqueid for a call? i need to be able to generate my own uniqueids.
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01:07.37*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
01:07.52Docelm0haha ya unix timestamp
01:08.11*** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
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01:12.42Avochelmah, it is too... but there's the funny decimal point on the end
01:13.04wunderkinummm what are you really trying to do
01:13.16Avochelmsomething awfull
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01:32.16livindedI know that asterisk has support for streaming a conference to icecast, but is there any support or a 3rd party script for shoutcast?
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01:42.07knarflyAvochelm: you just want to spoof your callerid?
01:44.45livindedare the police/other law enforcement starting to crack down on spoofing, i heard a few months back it was being looked into?
01:44.56livindedat least within america?
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01:48.45Avochelmknarfly, i'm copying a mysql table of call records into another table (already populated) and want to generate new uniqueid for them to prevent clashes.
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01:51.36Un1xfuck man
01:51.38Un1xfucking splitfinity sucks
01:51.42Un1xthere tech support sucks too
01:52.04mogwhat you need un1x
01:52.16Un1xman i been trying to get help to setup my spa-3000 with them
01:52.21Un1xand they say hello
01:52.24Un1xon msn and elave for 2 hours
01:52.31Un1x:S
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01:52.49russellb~striplsd
01:52.55russellb~striplastdigit
01:52.56jbotstriplastdigit is probably ${EXTEN:0:$[${LEN(${EXTEN})} - 1]} , will remove the last digit from EXTEN, making 5551212 become 555121.  Change the "1" to remove more digits.
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01:58.35skraelings001hi
02:00.53riddleboxcan someone help me with this error? http://pastebin.ca/87497
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02:09.13dlynes_laptopUn1x: have you tried the sipura user group forums on voxilla?
02:09.41dlynes_laptopUn1x: there's a number of really good threads there, detailing how to set up a sipura 3000 to work with asterisk
02:09.56Un1xim not setting it up with asterisk
02:10.55dlynes_laptopUn1x: oh
02:11.12dlynes_laptopUn1x: what exactly are you trying to do with it, then?
02:11.41dlynes_laptopriddlebox: looks like someone wrote an asterisk module, and never finished writing it
02:11.49dlynes_laptopriddlebox: or it was written for a really old version of asterisk
02:14.51riddlebox<PROTECTED>
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02:15.56dlynes_laptopriddlebox: ummm....res_smdi doesn't come with asterisk, afaik
02:17.43dlynes_laptopriddlebox: put a noload => res_smdi.so in your modules.conf file
02:18.11filemeep
02:18.11riddleboxdlynes_laptop, I deleted the /usr/lib/asterisk/modules folder, and I am going to compile from source
02:18.12dlynes_laptopi have no idea what ubuntu dapper is
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02:18.52livindeddlynes_laptop: its the next release of ubuntu
02:19.03dlynes_laptopthe next release after the current one?
02:19.13livindedyes, dapper is currently in beta
02:19.20dlynes_laptopcool...so what's the current one? :P
02:19.22riddleboxwhat
02:19.28riddleboxdapper is the current release
02:19.30livindedbreazy beaver or something
02:19.41livindedwhen was dapper released?
02:19.42dlynes_laptoplivinded: that sounds kinky
02:19.57dlynes_laptopa breazy beaver
02:20.00riddleboxlast month
02:20.10dlynes_laptopi thought that's when a girl was wearing a really short skirt
02:20.33livindeddlynes_laptop: its not really breazy beaver, its breazy something else, i always call it breazy beaver becuase i can't remember the real animal and i thought it sounded better
02:20.34riddleboxok how do I make absolute sure that asterisk is removed from my system?
02:20.54Un1xlol sounds like you used someon elses system as a PBx....
02:21.02Un1xand wherent allowed to and now trying to clean ur tracks :P
02:21.25dlynes_laptopfor dirfile in `find / -name asterisk`; do rm -rf $dirfile; done
02:21.37riddleboxno, I have had so many problems with this machine
02:22.05livindeddlynes_laptop: its breazy badger
02:22.15dlynes_laptopi like breezy beaver better
02:22.23livindedme too
02:22.32dlynes_laptopso it's going to stay breezy beaver
02:22.41dlynes_laptopwho ever heard of a breezy badger, anyways?
02:22.49livindedexactly
02:23.01livindedbut i was wrong, dapper drake was released
02:23.10riddleboxjust for fun I did apt-get install asterisk on my laptop, it worked fine, which is also dapper, but on my main machine it doesnt work
02:23.25livindedriddlebox: never ever use an asterisk package
02:23.30dlynes_laptopriddlebox: maybe you're grabbing an unstable package
02:23.48dlynes_laptopriddlebox: i always compile from source
02:23.56dlynes_laptopriddlebox: then i know exactly what i'm getting myself into
02:24.23livindeddlynes_laptop: its not like its even hard, you wget 4 files, untar, and typ make && make install
02:24.48livindedassuming you already got the dependencies
02:24.54dlynes_laptoplivinded: ummm...it's a bit more than that..I use sangoma hardware :)
02:24.55riddleboxthats the thing, I did apt-get build-dep asterisk, to get every package I needed to, compile from source but yet there are errors
02:25.09livindedsangoma?
02:25.17dlynes_laptopyeah
02:25.21dlynes_laptopwww.sangoma.com
02:25.35dlynes_laptopthey make some really good pri and tdm cards
02:25.55skraelings001how can i avoid peers from being unreachable or too lagged?
02:25.59livindedare they cheaper than digium?
02:26.11dlynes_laptoplivinded: retail, around the same price
02:26.24dlynes_laptoplivinded: however, sangoma doesn't screw over their distributors like digium does
02:26.39dlynes_laptoplivinded: so you can get the cards cheaper wholesale than you can retail
02:26.41livinded:( i can't justify paying for a tdm card for a pbx that wont be connected to my punchdown block when i don't have a stable job
02:27.06hadsdlynes_laptop: What do you mean by screwing over the distributors?
02:27.35dlynes_laptopdlynes_laptop: i've heard digium's reseller network is all fubar because they sell the same price or cheaper than their distributors
02:27.44riddleboxno I get this, after doing what dlynes_laptop said to remove asterisk, and then recompiling it, http://pastebin.ca/87614
02:27.50dlynes_laptopi.e. they compete with their middlemen
02:28.05livindedwow that sucks
02:28.35hadsAh, yeah I guess they do compete a bit in the US, but it doesn't apply to me as I'm too far away.
02:28.53dlynes_laptophads: ah
02:29.04dlynes_laptopriddlebox: you didn't install it properly
02:29.10livindedi'm just waiting for my digium screwdriver to come
02:29.16dlynes_laptopriddlebox: read the docs on how to install it properly
02:29.19livindedthose are the coolest things ever!
02:29.20hadsThe stuff they sell is "retail boxed" and all the distributed stuff is OEM packaged. Not that it's a mojor difference.
02:29.27dlynes_laptophads: if you're an end consumer, you notice no difference
02:29.29riddleboxdlynes_laptop, I untarred make sudo make install
02:29.39dlynes_laptopriddlebox: yeah, iow, you didn't install it properly
02:29.50hadsdlynes_laptop: I sell gear :)
02:29.55dlynes_laptopriddlebox: you forgot make samples
02:30.11livindeddlynes_laptop: you don't need samples
02:30.14riddleboxwhy do I need make samples?
02:30.22dlynes_laptoplivinded: you do if you deleted all your config files
02:30.28livindedriddlebox: do you want to make your own configs
02:30.41riddleboxyes
02:30.52riddleboxI have all of my .conf files backed up
02:30.57dlynes_laptopriddlebox: type ls -al /etc/asterisk
02:30.58livindedthen don't make samples, but you need config files to start asterisk
02:31.04*** part/#asterisk ph|ber (n=phiber@slackwaresupport.com)
02:31.22riddleboxahh that may be it then
02:31.24dlynes_laptophads: i suppose screwing over is too strong of a term, though
02:31.50hadsProbably, I understand where you are coming from though.
02:31.53dlynes_laptophads: their tech support is pretty good
02:32.08dlynes_laptophads: but I have not had very good experience with digium hardware
02:32.12livindedunfortunatly i've never got to deal with their tech support
02:32.15hadsTBH, I've never talked to Digium direct.
02:32.24riddleboxthat was it
02:32.25dlynes_laptophads: you shouldn't need specialized boxes in order to run their hardware
02:32.34dlynes_laptophads: any off-the-shelf server should work
02:32.48livindeddlynes_laptop: since when do you need specialized hardware to use a digium tdm card?
02:32.57livindedaren't they normal pci cards?
02:32.59dlynes_laptoplivinded: who knows...never tried
02:33.07dlynes_laptoplivinded: i got tired of all the bs with their pri cards
02:33.10hadsdlynes_laptop: Huh? I know :) I sell the gear and install it.
02:33.12riddleboxwohoo I am sooo happy now
02:33.32livindedriddlebox: now go setup a miliwat and listen to it
02:33.39dlynes_laptoplivinded: and so when it came time to buy tdm cards, I went straight to sangoma instead
02:34.03livindeddlynes_laptop: i'll keep that in mind when i can afford a card, i need to buy a new box first though
02:34.06riddleboxlivinded, miliwat?
02:34.19dlynes_laptoplivinded: besides...you cannot get hwec on a 4 port tdm card from digium
02:34.27dlynes_laptoplivinded: you can from sangoma...even two port
02:34.30livindedhwec?
02:34.32hadsYeah, that's a little annoying.
02:34.35dlynes_laptophardware echo canceller
02:34.39livindedoh
02:34.40Luke-JriConnectHere sucks
02:34.48hadsSangoma is nice for that.
02:34.49dlynes_laptopand sangoma's are all carrier grade from Octasic
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02:35.01livindedooooo awsome wont eat up my cpu using asterisk
02:35.05dlynes_laptopfor me that's a double bonus
02:35.12dlynes_laptopsangoma and octasic are both Canadian companies
02:35.27dlynes_laptopI'd prefer to support Canadian business, as I'm also Canadian
02:35.50hadsFair enough.
02:36.13dlynes_laptopI'd like to support Digium because they help finance the development of asterisk too
02:39.58*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
02:39.58*** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.9.1 and 1.0.11.1 released with a critical security fix for chan_iax2, please upgrade immediately (June 6, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx
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02:40.04livindedi think the server is broke, thats the same one
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02:40.08Un1xi wish networks would get motre money and then theyd buy new servers and wouldn't have that much load
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02:40.16dlynes_laptopUn1x: in my case, it was my server chopping my connection
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02:40.18Un1xmy network never used to split
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02:40.32Un1xsame here i think
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02:40.54livindedmy network never splits, but we only have maybe 120 at any given time and 4 servers
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02:41.12dlynes_laptopyeah...but in this case, the server went completely down; it wasn't a netsplit
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02:41.35dlynes_laptopit was chat.freenode.net
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02:42.56livindedi need an "i <3 netsplits" sticker for my laptop
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02:43.43RageMaxyou guys know the site to get a free DID?
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02:43.52livindedRageMax: ipkall
02:43.53RageMaxI can't seem to find the link
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02:44.09RageMaxI think it was freedid or something similar
02:44.16livindedtheres that one too
02:44.36brut-question: it is possible to host the asterisk box behind NAT and make it work..., right? :<
02:45.06livindedbrut-:sure, just forward the ports
02:45.09RageMaxbrut-: yeah, if you use the IAX protocol, SIP gets kind of messy
02:45.41brut-aye, ok..., then I've got something mis-configured... thanks. :)
02:45.42brut-I was just hoping i wasn't trying to do something that's known to not work
02:45.42livindedi've never had a problem with sip behind my nat but there are problems with it
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02:46.49RageMaxlivinded: is there one that works with IAX?
02:47.22livindedRageMax: a free did service? not that i know of.
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02:53.05dlynes_laptopaol offers free dids, too
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02:56.11dlynes_laptopsipgate.co.uk, sipgate.de also offer free didds
03:02.41danphow much is a 4-port FXO sangoma card with echo cancellation?
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03:04.17RageMaxis there one with "free did" in its name?
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03:05.19dlynes_laptopdanp: retail?
03:05.35rob0www.ipkall.com ?
03:05.55dlynes_laptopyeah.."ip kall" looks a lot like "free did" :)
03:06.04RageMaxdanp: those types of cards are usually around $500
03:06.12rob0someone mentioned ipkall up there
03:06.26dlynes_laptopRageMax: ummm...that's retail, and that's without the hwec
03:07.31rob0ipkall is a great service. In fact I think I might hang out a virtual shingle to do business in the Seattle area. :)
03:14.17danpso they're about the same as a similar digium card?
03:14.45hadsdanp: Where are you located?
03:15.00danparizona, US
03:15.06RageMaxhrm
03:15.12dlynes_laptopdanp: yeah, except digium doesn't have hwec on their tdm cards until you get into the tdm2400p
03:15.39RageMaxrather than using FWD as the sip proxy for ipkall, can you just use a dyndns address directly to your asterisk box?
03:15.58hadsI'm sure someone in here will be able to point you to some online stores in the US, that's the easiest way to get prices.
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03:16.15danpi was just looking for a ballpark
03:16.56hadsdanp: http://www.google.co.nz/search?hl=en&ie=UTF-8&oe=UTF-8&q=sangoma+a200+price
03:17.02RageMaxlike I said, ballpark is about $500
03:17.03RageMaxUS
03:17.05danpcool
03:18.39rob0RageMax: that's what I do, 'cept it's my own DNS.
03:19.07RageMaxrob0: so is it setup like a normal sip extension?
03:19.35rob0Yes, there's a HOWTO page on the Wiki. I followed it, it works.
03:20.11RageMaxthe voip wiki?
03:20.27RageMaxvoip-info
03:20.51rob0right, I think that's what's normally known as "the" wiki here. :)
03:21.26RageMaxfigured
03:21.41RageMaxI've been using it for almost 4 years, when asterisk was in early beta ;)
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03:23.33danpi remembered that we get our stuff through voipsupply.com...that card's about $660
03:23.41danp4 port FXO with echo cancellation
03:23.55danpin case anyone was curious :P
03:25.07wunderkinfroogle.google.com
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03:35.16shmaltzdanp, what card was 4 port fxo with echo can?
03:35.56shmaltzfor around 660?
03:36.14danpthe sangoma a20002d
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03:36.45SarahEmmhihi
03:37.12shmaltzdanp, where are you using it?
03:37.22danpi'm not, i was just curious about it
03:37.30danpwe use 4-port digium cards at work
03:37.53shmaltzwell, the TDM400P from digium is something I stopped using a while ago
03:38.05danpyeah, TDM400P is what we use
03:38.08shmaltzI now use T1 card with channel banks
03:38.33danpwe don't have many analog lines...we deploy mainly small offices
03:38.37danpbbiab
03:39.14shmaltzhttp://www.voipsupply.com/product_info.php?products_id=1339
03:39.26RageMaxyou'll probably need the T1 sooner or later
03:39.32shmaltzso what do you have in small offices?
03:40.17RageMaxyou guys remmeber that really cheap win/fax modem that supposedly worked with asterisk with a small hack
03:40.48SarahEmmRageMax: yep. i use one here
03:40.49SarahEmmx100p clone
03:41.07RageMaxis that pretty much the only one that works?
03:41.22SarahEmmerr
03:41.26SarahEmmthere's a bunch that work
03:41.35SarahEmma couple intel chipsets, an ambient chipset
03:41.58SarahEmmhttp://www.voip-info.org/wiki/view/X100P+clone
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03:44.31SarahEmmRageMax: a lot of people have issues with x100p's with echo and other issues tho
03:44.37SarahEmmjust a warning if you care about audio quality...
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03:45.31fileyay relaxing
03:45.40russellbyay
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03:46.00filerussellb is totally like, 800km away from me
03:46.16russellbi.e. 10 feet
03:46.26hadsmust have a long poking device
03:46.33file:D
03:46.55filerussellb: I can't tell if that's a guy or a girl, despite knowing it's a guy :(
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03:48.04wunderkinyour poker device?
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03:48.32filerussellb just tried to poke me
03:48.32russellbow ... pain ....
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03:48.47wunderkinoh, russellb's poker..
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03:58.27Luke-Jrso anyone know a good origination provider?
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04:01.26mitchelocLuke-Jr, i'm looking for one right now as well, i need 9K minutes and it seems like nobody has targeted that sub-bulk amount =/
04:01.56mitchelocrm -rf /file
04:01.59mitcheloc;)
04:02.46hadsYay, weekend time!
04:03.33filerussellb: this TV is so very nice
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04:07.52russellbfile: indeed
04:08.03russellbwe have invaded the house of kpfleming!
04:08.48filezomg
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04:10.21CunningPikewb [TK]D-Fender
04:10.53mitchelochaha yea he has a nice tv, are you guys watching coblert report?
04:11.05fileyeah, he's gone out... just me and russellb
04:11.18mitchelocnice, loud crickets eh?
04:11.30filequite
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04:12.05mitcheloc:)
04:13.46trelanefile, quick run up the phone bill!
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04:14.11[TK]D-FenderI went *b00m*
04:14.43trelanefile, call the pope!
04:15.34fileno phone!
04:15.40trelaneno phone?
04:15.41trelanehow!?
04:15.46SarahEmmerr.
04:15.52filecrazy things called cellphones
04:15.55SarahEmmaren't you at a house with a bunch of VoIP people?
04:15.55trelanecall the dalai lama!
04:15.57SarahEmmthere's no phone?
04:16.01trelaneSarahEmm, no kidding
04:16.07trelanefile, gank his cellphone and call HHDL
04:16.09fileno analog house phone ^_^
04:16.18trelanevoip house phone?
04:16.22filewhat kind of cellphone would it be if he didn't take it with him?
04:16.24trelanehe's gotta have some snoms or some cisco's or even polycoms?
04:16.41SarahEmmheh :)
04:16.49trelanegrandstream?
04:16.50mitchelocfile, there is no phone, look over to your left, on the round coffee table, i'm pretty sure there is
04:16.53trelane3com?
04:16.59fileit's not hooked up
04:17.12mitchelocah, well nevermind then
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04:20.06filesleeeeeeepy
04:20.51russellbmhm
04:20.57russellbfile: this is a silly tv show
04:21.07SarahEmmheh :)
04:21.08SarahEmmoops
04:21.52mitchelocirc d-o-r-k-s, you are in the same room!
04:23.59russellb...or are we
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04:32.46CunningPikeWe're all in the same room.........
04:32.57Qwell[laptop]or are we?
04:33.02CunningPikelol
04:33.05SarahEmmlol
04:33.16fileQwell[laptop]: we're all just 800km apart
04:33.27Qwell[laptop]file: exactly
04:33.48Qwell[laptop]CunningPike: all the cool people (and file) were in the same room earlier
04:33.59filenub
04:34.03Qwell[laptop]:D
04:34.05mitchelocdorks...
04:34.17Qwell[laptop]mitcheloc: You should've stayed an extra week :p
04:34.23CunningPikemitcheloc is just jealous.......
04:34.32Qwell[laptop]nice pseudo-meeting you, btw, heh
04:34.41Qwell[laptop]for all of 30 seconds...
04:34.44mitcheloci know! damn
04:35.00Qwell[laptop]I'm thinking "wtf...that isn't mog..."
04:35.25mitchelocyep, the security guard gave me a hard time, haha, i could totally see that in your face, you were like wtf why is this guy talking to me...
04:35.36Qwell[laptop]heh, why'd he give you a hard time?
04:35.51fileQwell[laptop]: because you're a known terrorist and he was talking to you...
04:35.52mitcheloci was ignoring him while i was talking to you, and he was trying to check my boarding pass
04:35.53fileor at least known nub
04:35.55Qwell[laptop]file: ahh
04:36.01mitchelocthat too
04:36.02Qwell[laptop]mitcheloc: heh
04:36.17mitchelocyou all will be at astericon in boston anyway right?
04:36.19Qwell[laptop]mitcheloc: definitely interesting timing...
04:36.41mitcheloc** dallas
04:36.43Qwell[laptop]my flight was 20 mins late getting in too
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04:37.08filemitcheloc: dunno
04:37.31mitchelocoh yea nice timing for sure, good guess on my part too as to picking you out
04:37.53Qwell[laptop]mitcheloc: not hard. ;)
04:38.10Qwell[laptop]There was probably something obvious that tipped you off :P
04:38.35mitchelocwell when he said ugly, i didn't know you would be *that ugly* so yes it was easy =P
04:38.38Qwell[laptop]heh
04:38.44mitcheloc(just kidding)
04:38.54fileha
04:40.54Qwell[laptop]Kevin gave you a bed?
04:40.58Qwell[laptop]lame
04:41.05mitchelocQwell[laptop], where are you staying?
04:41.11Qwell[laptop]mitcheloc: residence inn
04:41.20mitchelocouch, lame
04:41.22filevery lame
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04:41.41Qwell[laptop]nah, it's a nice place, I think
04:41.58mitchelocbut you don't get to use the nice tv at kps
04:42.03Qwell[laptop]ahh
04:42.10fileQwell[laptop]: you haven't been here, have you?
04:42.16Qwell[laptop]nope
04:42.21mitchelocsomebody was there a week or two ago, and told me to try the waffle house, which i did, so i recommend you all to try it out, good food :)
04:42.23fileeep
04:42.28Qwell[laptop]mitcheloc: already did
04:42.31filewe did the waffle house
04:42.32mitchelocnice!
04:42.44filei bit my tongue while eating my lunch there >.<
04:42.45mitchelocomg did you all see kram's new car?
04:42.53mitchelocwe got stuck in the rain in it!!!!!!!!
04:42.58mitcheloci have pictures too, i'll develop them tomorrow
04:43.01fileha
04:43.15mitchelocit wasn't funny!
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04:47.30Luke-Jrmitcheloc: I just need something :/
04:47.49Luke-Jrmitcheloc: any non-bulk seems to be trash
04:48.08Luke-Jreven iConnectHere, which seemed to be a well-established company
04:48.54Luke-Jrthey just randomly forgot my DID # sometime recently
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04:56.15batttHello, I am a bit confused. according to all the doccumentation I found, in order to hook asterisk up to a voip service I have to use either t1 or ISDN. Isn't there a way I can use my plain jane dsl connection?
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05:06.15CunningPikebattt: Yes, if you subscribe to a service provider that connects to the PSTN, or do not wish to use the PSTN at all
05:06.52copantlany body  use a2billing?
05:06.56batttwhat kind of hardware would I use? surely not a wildcard or t1 card, would I just use an extra nic?
05:07.38CunningPikebattt: Provided what I said applies, you can use your existing NIC
05:08.46batttah, okay, thanks for clearing that up for me. the asterisk book made me think I had to use a t1 or an ISDN connection, no ifs ands or buts.
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05:12.03copantlany body  use a2billing?
05:12.14CunningPikebattt: No, only for connecting to the PSTN - but you will probably want to connect to the PSTN at some point.......
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05:53.53twislamelerisme roxor.
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05:59.40mostyi have some DID's being routed to be via iax2, in my * console i see "Rejected connect attempt from <IP>, request '<DID>@incoming_voip' does not exist, but my incoming_voip context does exist. what could be wrong?
05:59.53mostyrouted to me, rather
06:00.34drrayI've only dealt with PRI did's
06:00.50drraybut don't you need a channe= > xxxxxxxx for it?
06:00.54drrayer, channel
06:01.23drrayfor example, only did's that I sepcify get into my dialplan the rest are rejected
06:01.45mostywhere do you specify that?
06:01.52drrayextyensions.conf
06:01.57drrayminus the y
06:02.17mostywell i have an incoming_voip context, which simply sends all calls to incoming,s,1
06:02.36drrayexten => 1074,1,Background(/etc/asterisk/main-menu)
06:02.51dlynes_laptopmosty: [incoming_voip]\nexten => <DID>,1,dothis()
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06:02.59dlynes_laptopmosty: do you have something like that?
06:03.02drray1074 being the last 4 digits
06:03.08drrayof the DID
06:03.15drraylisten to dlynes..
06:03.22dlynes_laptopmosty: i.e. you have to specify the extension in that context
06:03.30dlynes_laptopmosty: it doesn't come in on extension 's'
06:03.30mostydlynes_laptop: no, but i have exten => s,1,Goto(mainmenu,s,1)
06:03.34mostyoh ok
06:04.01mostyi was sure this worked the other day. i will specify the did and see what happens
06:04.16*** join/#asterisk Corydon76-home (i=three@pdpc/supporter/sustaining/Corydon76-home)
06:05.01mostythat worked- thankyou both
06:05.13dlynes_laptopnp
06:05.34drraythank dlyne more:)
06:05.37dlynes_laptopmosty: you might even be able to do exten => _X.,1,...
06:05.57dlynes_laptopmosty: but I wouldn't use that approach myself, in case someone's just trying to find a way into your system to spam your customers
06:06.08mostydlynes_laptop: that's what i did. i would do _. if only * wouldn't warn me not to
06:06.09drrayisn't that insecure
06:06.10drray?
06:06.10dlynes_laptopmosty: i.e. the voip version of spammers
06:06.57mostydlynes_laptop: well i have the iax2 ports firewalled to only let a specific voip provider send me calls
06:07.09dlynes_laptopah
06:07.15drraysounds like you are done then
06:07.20mostyyes
06:07.33mitchelocmosty good idea, what if they change ips though?
06:08.50mostymitcheloc: a friend is the admin for the provider, he will let me know if that happens
06:09.40drraychances are your users will let you know :)
06:10.15mitchelocwell, it'd be nice if there was a more reliable way to set that up
06:10.35mitchelocdns would be more accurate? but it's not as reliable
06:11.02dlynes_laptopmitcheloc: well, i'm sure his voip provider is on a dynamic ip
06:12.23drrayyou could allocate the subnet
06:14.22pdtmobileanybody in here done much with FastAGI?
06:15.50_GuhitI'm trying to get asterisk setup and everything is installed and started.  I can call in an I get the demo, but I can't seem to figure out how to disable the demo and start adding my own dialplan.  I'm running FreeBSD and installed it via the ports.
06:16.05pdtmobileI have a very simple setup currently a hello world of FastAGI if you will...  AGI(agi://127.0.0.1)
06:16.26pdtmobilethe server never sees a connect but asterisk is saying everything is A OK
06:17.03pdtmobileif I telnet to the port the server acknowledges i connected, that doesn't happen when I call into the dialplan I just get a AGI Script agi://127.0.0.1 completed, returning 0
06:17.03Nuggettelnet is eeeeeeevil!
06:17.19pdtmobilei am gonna have to start saying netcat in here
06:17.30*** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it)
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06:20.16dlynes_laptoptelnet
06:20.16dlynes_laptoprules
06:20.17dlynes_laptopnugget
06:20.17dlynes_laptopdrools
06:21.59CunningPike_Guhit: You need to modify extensions.conf to create your own dialplan
06:22.36_GuhitCunningPike: yeah, I just found the include => demo bit
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06:30.25dlynes_laptopCunningPike: have you gotten around to writing any agi scripts yet?
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06:30.48CunningPikedlynes_laptop: No - not yet. Are you?
06:30.53dlynes_laptopthinking about it
06:31.01dlynes_laptopwas just curious how easy it was
06:31.22CunningPikedlynes_laptop: I attended a presentation at Astricon last year and it didn't seem hard
06:31.32dlynes_laptopah
06:34.40CunningPikeI'm smashing my head against SugarCRM atm
06:35.53mitchelocdon't use it?
06:36.20CunningPikemitcheloc: My head, or SugarCRM?
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06:38.10CunningPikedlynes_laptop: Got FOP working - it's quite cool
06:38.12mitchelocSugarCRM
06:38.35CunningPikemitcheloc: It's not that bad - I'm importing data from their previous system, which is a PITA
06:38.44dlynes_laptopCunningPike: ah, you did?
06:38.55dlynes_laptopI'm having issues with it, myself
06:39.10dlynes_laptopgot the layout working fine, but the status and stuff like that doesn't seem to be working
06:39.21CunningPikedlynes_laptop: Really? I've only got a barebones installation running, but it works just fine
06:39.25dlynes_laptopand the drag 'n drop seems to be not dragging and dropping the way it's supposed to
06:39.43dlynes_laptopCunningPike: can you transfer calls to voicemail, and monitor the status of extensions and lines?
06:40.00CunningPikedlynes_laptop: Yes - I haven't got that working yet - none of clicks and drags work, but the status works like lightning
06:40.16mitchelocCunningPike, regarding fop, are you actually pleased with it as an interface? in so much as you accept it as being the best out there (and yet not as good as it could be?)
06:40.31loopthi! somebody can tell me what is this warning and how can i fix it?
06:40.32looptWARNING[5359]: channel.c:2221 ast_write: Thread 1092921696 Blocking 'mISDN/2-1', already blocked by thread 1095051616 in procedure ast_waitfor_nandfds
06:41.22CunningPikemitcheloc: It is, ahem, adequate - and certainly the best I've found. I tried HUDLite, but can't get it working and anyway its configuration looks way complicated
06:41.26drrayFOP is not open
06:41.36mitchelocdrray: not open?
06:41.45CunningPikedrray: Correct - not OSS
06:41.55drrayyou can't modify the flash component
06:42.09CunningPikeHUDLite is a binary also......
06:42.13mitchelocCunningPike, i thought it was... it's just flash, tough to use
06:42.21drrayplus I have too many lines for FOP to work with
06:42.32drrayand flash/firefox does not play nice
06:42.53mitchelocwell if you just need your personal queue you can use snap ;)
06:43.04CunningPikeWhat I would like to do with it is have different pages for different departments - not sure if that's possible without having multiple instances
06:43.18mitchelocbut yes i'm curious as to feedback along the lines for software like FOP, like what would you improve?
06:43.23hads|homefrom the FOP FAQ: "You will find the perl source to compile the swf under the ming-source directory"
06:43.25dlynes_laptopCunningPike: yeah, it's possible
06:43.36dlynes_laptopCunningPike: fop comes iwht the perl code to generate the flash applet
06:43.44mitchelocso it is open source,nothing wrong with that then? but flash code is cryptic heh
06:43.49dlynes_laptopCunningPike: you can modify the flash applet perl generation code
06:43.55CunningPikedlynes_laptop: Ugh
06:43.59dlynes_laptopCunningPike: i think it needs to be run under windows though
06:44.09CunningPikedlynes_laptop: What does?
06:44.17mitchelocthe perl generator
06:44.24CunningPikemitcheloc: Ah, I see
06:44.25dlynes_laptopCunningPike: the perl swf generator
06:44.54*** join/#asterisk spooky7 (n=spooky@ipa121.208.tellas.gr)
06:44.55dlynes_laptopCunningPike: because you probably need adobe flash studio or whatever it's called installed
06:45.04CunningPikeWell, it's certainly the best of what's out there.......
06:45.06dlynes_laptopCunningPike: so that it has the creation routines
06:45.08spooky7good morning asterisk users
06:45.26dlynes_laptopCunningPike: however, the latest version does dhtml, too
06:45.32drrayI wonder if a SNMP setup would work for that
06:45.53dlynes_laptopdrray: yeah, but if cp's anything like me, he needs something a  secretary can handle, not a system administrator
06:45.54CunningPikedlynes_laptop: I couldn't get that to display - might be my browser
06:46.00spooky7need some help with asterisk and druid . Can anyone help me ?
06:46.09CunningPikedlynes_laptop: Precisely ;)
06:46.10dlynes_laptopCunningPike: are you using a non-standards compliant browser like internet explorer?
06:46.20CunningPikedlynes_laptop: Nope - Omniweb
06:46.25dlynes_laptopCunningPike: try firefox
06:46.29dlynes_laptopCunningPike: it displays it just fine
06:46.30mitchelocdlynes_laptop, ie set the standard, everyone else just didn't want to follow it!
06:46.45dlynes_laptopmitcheloc: standards are documented
06:46.46CunningPikedlynes_laptop: I'll try it - thanks
06:46.53dlynes_laptopmitcheloc: half the crap ie does isn't documented
06:47.18mitchelocwho put the standards body in charge anyway?
06:47.25mitchelochave you looked at msdn?
06:47.38dlynes_laptopbut if that half of it actually was documented, spammers and virus writers would have even more free reign on windows machines
06:47.44mitchelocthere is also a lot not documented in FF, but the excuse is that it's open source so you can go look
06:47.53hads|homedeja vu, didn't this conversation happen yesterday?
06:47.59dlynes_laptophads: which convo?
06:48.13hads|homeIE vs FF etc.
06:48.31mitchelochads|home, i'm just playing devil's advocate :)
06:48.41CunningPikeI have to say that I'll take the IE documentation in MSDN over the pitiful Mozilla documentation any day
06:49.16hads|home:) Fair enough, I guess people voucing for IE would be the minority in an open source channel :)
06:49.27mitchelocpersonally i think microsoft has been doing a really good job of cleaning up their act lately
06:49.56CunningPikeBut - big caveat - when I code web stuff, I always seem to have to put 'if browser==IE' stuff in my code. I also have to put a fair amount of 'if browser==firefox' too
06:50.30dlynes_laptopCunningPike: i don't put any of that stuff in
06:50.35spooky7need some help with asterisk and druid . Can anyone help me ?
06:50.41dlynes_laptopCunningPike: i let a web jockey handle all that headachy stuff :)
06:50.55dlynes_laptopspooky7: isn't druid a disk volume manager for redhat?
06:51.02CunningPikedlynes_laptop: I am a web jockey :D
06:51.15dlynes_laptopheh
06:51.20spooky7no it is a web interface for asterisk
06:51.26x86hmm doesnt asterisk-sounds contain all the month names spoken out?
06:51.28dlynes_laptopah...never heard of it
06:51.43dlynes_laptopx86: probably /usr/lib/asterisk/sounds/months/january.gsm or something
06:52.02mitchelocif you guys are trying web interfaces try out phone call from their svn repo, it's pretty sweet
06:52.04x86i have a months.gsm, but no months/
06:52.18dlynes_laptoperm /var/lib i mean
06:52.21dlynes_laptopguess not though
06:52.24hads|homespooky7: Druid's a commercial thing?
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06:52.37spooky7yes i have bought it
06:53.01hads|homeDo you get support with it?
06:53.02mitchelocspooky7, why did you pick it over freepbx or phonecall or any other one?
06:53.32dlynes_laptopmitcheloc: because it wasn't free?
06:53.49spooky7I like the web interface of druid
06:53.52mitchelocdlynes_laptop, if i was going for non-free i'd have taken a serious look at scopserv
06:54.01dlynes_laptopbut seriously...I think i've seen another commercial offering out there that was tonnes better than freepbx
06:54.04spooky7I have paid fro druid
06:54.20mitchelocdlynes_laptop, scopserv i think, and phonecall is pretty good
06:54.32dlynes_laptopyeah...scopserv i think was the one i looked at
06:55.07dlynes_laptopyeah...it was scopserv
06:55.41dlynes_laptopI'm guessing scopserv is the offering packaged with Allworx' PBX?
06:55.42mitchelocyep theirs looked very professional
06:56.03dlynes_laptopcause i see Allworx' PBX pictured on scopserv's website
06:57.09x86dlynes_laptop: nope
06:57.22x86dlynes_laptop: niether asterisk-sounds or asterisk-sounds-extra contain month names
06:57.28spooky7so can anyone help me ?
06:57.29x86dlynes_laptop: know where i might find them?
06:57.35CunningPikeWe looked at a few GUIs, but didn't like any of them
06:57.48CunningPikeI find the conf files much easier to deal with
06:57.49dlynes_laptopno idea...maybe talk to digium to get them to get allison to record them for you?
06:58.07CunningPikex86: Talk to Allison directly
06:58.10CunningPike~thevoice
06:58.15dlynes_laptopCunningPike: yeah, but eventually i have to look for a gui config tool or a telnet config tool or something
06:58.32hads|homeI heard telnet was eeeeeeevil.
06:58.33dlynes_laptopCunningPike: she won't talk to you directly, apparently...you pay for her through digium
06:58.37CunningPikedlynes_laptop: I'm building those scripts I was telling you about  - for our help desk
06:58.50drrayit's cheaper to just buy voice credits from digium
06:58.50dlynes_laptopwhich scripts were those again?
06:59.19x86CunningPike: eh, too much money ;)
06:59.29CunningPikedlynes_laptop: Some shell scripts that allow them to add a new phone, reset passwords, change names etc
06:59.45dlynes_laptopah...like cgi scripts that they can activate from a web page?
06:59.56CunningPikedlynes_laptop: In a kind of ./configure style from a CLI
07:00.08CunningPikedlynes_laptop: Text based menu
07:00.16dlynes_laptopah...and secretaries have enough gray matter to figure stuff like that out?
07:00.28CunningPikedlynes_laptop: Not secretaries - help desk
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07:00.57dlynes_laptopok...help desk people then
07:01.04dlynes_laptopsame gene pool i think
07:01.07mitcheloci think that phone call has that
07:01.49CunningPikedlynes_laptop: lol - sometimes I wonder
07:02.21dlynes_laptopCunningPike: ever try talking to shaw tech support?
07:03.00CunningPikedlynes_laptop: Did once - needed a lobotomy afterwards
07:03.13CunningPikedlynes_laptop: They are a waste of God's good air
07:03.17dlynes_laptopheh
07:03.35dlynes_laptopthey really put the tech in tech support, don't they? :)
07:03.44dlynes_laptopright up there with the sales people at Can Computers
07:03.56*** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com)
07:03.56dlynes_laptoperm actually
07:04.14dlynes_laptopthe sales people at can computers make shaw tech support guys look like brain surgeons
07:04.19CunningPikedlynes_laptop: Their role in Shaw is to prevent anyone from actually talking to anyone in there with clue
07:04.35dlynes_laptopoh...you're running Linux?
07:04.50dlynes_laptopWell, that must be why it's not working; Linux is not a supported platform!
07:05.14dlynes_laptopor there's the other ones that don't even know what Linux is :)
07:05.16drraylinux is for sucks and squares
07:05.30NuggetLinux is poo.
07:05.31CunningPike"You have a Mac? Oh, we don't support Macs. I don't think they have the Internet anyway"
07:05.35CunningPikeVerbatim
07:05.38dlynes_laptophahahahahah
07:05.45dlynes_laptopnow that's hilarious :)
07:07.09CunningPikeI nearly had a stroke
07:07.53x86http://cafe.bevocal.com/libraries/audio/female1/en_us/datetime/
07:07.55x86w00t :)
07:09.19mitchelocmac? internet?
07:09.19CunningPikemitcheloc: Oh, yes - since 1984, actually ;)
07:09.19*** join/#asterisk littlejohn (n=little@host63-66.pool8716.interbusiness.it)
07:09.26mitchelocwell even if they do windows is better
07:09.28dlynes_laptopCunningPike: or 602-555-1212 is even funnier
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07:09.47dlynes_laptopCunningPike: You're calling from Canada?  Which state is that in?
07:09.56CunningPikemitcheloc: You're entitled to your opinion........
07:10.09mitcheloc=P
07:10.10CunningPikemitcheloc: Even if you are deluded
07:10.12CunningPike;)
07:10.35CunningPikedlynes_laptop: That's Canada, Texas, ma'am
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07:14.37CunningPikeYay - all done!
07:15.21SimoAmihi there
07:16.00SimoAmilittle quite in here
07:16.02SimoAmi;)
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07:16.55SimoAmiwhat digium card to use for a pri or fractional pri line
07:17.39dlynes_laptopte110p, or te105p(?)
07:17.46dlynes_laptopbasically a single port pri card
07:19.18SimoAmiok, thanks
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07:29.00cy3o3word
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07:53.18chazmanhello
07:54.03chazmanis anyone able to assist me here with Asterisk?
07:57.37chazmanlol is anyone here gunna talk?
08:01.22SimoAmihi again. Is a ISDN  PRI line a flat rate or per usage ?
08:03.38pnlarssonISDN PRI is just a way to get you a connection, then you can get all kind of deals with the telco. But normally a montly cost + usage
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08:05.12SimoAmihow much would verizon charge for a pri in the new york area ?
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08:06.19pnlarssongive them a call...
08:08.36SimoAmiok, how many simultaneous calls one can receive with a fractional pri
08:09.46hads47
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08:11.12SimoAmiwow 47!
08:11.50chazmanDoes anyone know where I can go to get the cards I need for my machine to allow me to connect my incoming phone line and handsets to the Asterisk server?
08:12.45SimoAmiin your opinion, does it make sense to order a fractional pri line to satisfy that the client can dial and receive up to 6 calls (incoming, outgoing or mixed)
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08:13.57SimoAmichazman: http://www.digiumcards.com, http://www.voipsupply.com
08:14.44chazmanI am needing cheap ones, if there are any
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08:18.43SimoAmichazman: get a Linksys SPA-3102 NA 1FXS / 1FXO Analog VoIP Gateway for $89.95
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08:19.01SimoAmilink: http://www.voipsupply.com/product_info.php?products_id=1646
08:19.11drraycan you use that with asterisk?
08:19.17chazmanOk is it possible to connect more than one handset to it? and can they be a standard household phone?
08:20.04SimoAmiit's for one handset only, unless you use ip phones
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08:20.43drrayit's not locked into vonage or some other provider?
08:22.21SimoAminope you can use it with asterisk
08:22.43SimoAmiit's mentioned in the Asterisk guide here:
08:22.45SimoAmihttp://nerdvittles.com/index.php?p=123
08:23.17SimoAminote that this is a new replacement for the popular SPA-3000
08:24.31drrayI saw
08:24.48drraymaybe i can start dumping IAxy
08:24.49drrays
08:25.05chazmancan I put a splitter on it and use mroe than one?
08:25.30chazmanor have it automatically wire my house by connecting it to an existing phone line?
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08:28.11chazmanalso, I was looking for a PCI card?
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08:29.21SimoAmiyou could use a splitter, however it is recommended to use self powered phone terminals
08:29.43MrChimpygood morning telemonkeys
08:30.20chazmanOk. I need support for one phone line and 5 handsets. do I need special phones like the Nortel phones to work with Asterisk, or can a standard phone found in the Average house work?
08:30.50SimoAmiyou have 2 choices
08:31.36chazmanWhat are those choices?
08:31.53SimoAmietheir invest in ip phones and wire them to an ethernet network (they're for less than $100 each)
08:32.19drrayyou can use a splitter, but only one phone will work at a time, and yuo have to watch out for ring equivalency rating
08:32.21SimoAmiasterisk will communicate with them through tcp/ip
08:32.55chazmanok I am trying to use my current phones that I have. They are not Digital phones, but Panasonic cordless phones that the average joe would get
08:33.09SimoAmithe other way is to get 2 tdm400 pci cards with appropriate modules in them
08:33.46drraychazman - that should work, assuming the phones don't take too much power
08:34.02chazmanBut how would I program them with their own extensions?
08:34.15drrayto do that, you'd need a tdm card
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08:34.21drrayor one magic box per phone
08:34.32chazmanI have a server room, which is where all my stuff is, and I want to have easy communication with all rooms in my gome
08:34.32drrayor a tcp/ip voip phone
08:34.33chazmanhome
08:35.20chazmanOk I looked at a TDM card, i found one for under 80
08:35.28chazmanhttp://www.voipsupply.com/product_info.php?products_id=290&searchid=38413
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08:35.51chazmanwould I just install that onto my existing fxo or fxs card?
08:36.18chazmanthe card I was looking at getting is: http://www.voipsupply.com/product_info.php?products_id=1388
08:36.24drraythat's a module, for the tdm400
08:36.26chazmanThat is, once Ive got the money and a job
08:36.27SimoAmithat's a module that you plug in a tdm pci card
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08:36.41SimoAmiso you need to buy the card first
08:36.46SimoAmithis is what you need
08:36.47SimoAmihttp://www.voipsupply.com/product_info.php?products_id=290&searchid=38413
08:36.53SimoAmioops wait
08:37.00SimoAmiDigium TDM40B - (4) FXS VoIP SIP IAX H.323 Asterisk  $342.90
08:37.00SimoAmi+
08:37.00SimoAmiDigium TDM11B - (1) FXS & (1) FXO VoIP SIP IAX H.323 Asterisk
08:37.15chazmanOk what if I ordered the Nortel phones that I was looking at? They are digital i think
08:37.59SimoAmithat should work but you stil need to connect the analog  phone line itself
08:38.00chazmanhttp://products.nortel.com/go/product_content.jsp?segId=0&catId=null&parId=0&prod_id=47365&locale=en-US
08:38.07chazmanthat is the kind that I would get if I had to
08:40.00drrayhow much per phone?
08:40.18chazmanMSRP 139.99 US
08:40.18SimoAmidoesn't look like an ip phone to me
08:40.29chazmanThey have those exact phones at my school
08:40.35drrayit's an analog
08:41.02chazmanok, but I take it a school doesnt have a virtual PBX, and a physical unit that does all the work
08:41.50chazmanhttp://www.voipsupply.com/product_info.php?cPath=95_106&products_id=1057
08:41.53chazmanwhat about that?
08:42.26SimoAmiif you wanna connect 5 phones as you stated earlier, and you want intercom feature, then you need a pbx
08:42.49*** part/#asterisk littlejohn (n=little@host63-66.pool8716.interbusiness.it)
08:42.51chazmanIs Asterisk not a PBX?
08:43.02chazmanI have a machine running Asterisk@home
08:43.09SimoAmiwell that's what I'm coming to
08:43.37chazmanso then do I need an actual PBX unit? will Asterisk not work as a PBX system?
08:44.04SimoAmiso yes, look for cheap ip phones, it could be a good option for you
08:44.16SimoAmiyes
08:44.21drrayasterisk can work as a pbx
08:44.26SimoAmiyes
08:44.43SimoAmiyou need a pbx and asterisk is a pbx
08:44.44chazmanOk well http://www.voipsupply.com/product_info.php?cPath=95_106&products_id=1057 appears to be an IP phone. Is Asterisk a IP PBX or can it support normal phones as well?
08:45.06SimoAmiit does both
08:45.19chazmanOk so for that Nortel phone, how would I use those?
08:45.21drrayI run 180+ normal phones in a hotel with asterisk
08:45.34drrayfor that nortel phone you'd need FXS ports
08:45.53SimoAmito connect  normal phones you need an adapter (they call it a FXS)
08:46.14drrayit can either be in the function of a PCI card, or a MTA
08:46.27drraybut if you are going to buy a MTA you might be better off with a good IP phone
08:46.36chazmanOk so if I get only an FXS PCI card, can I just use that to establish a small internal PBX network? At the moment I do not want to extend it to support incoming calls and outgoing calls
08:47.00drrayyes, you'd need 1 fcs port per extension in asterisk
08:47.03drrayfxs
08:47.15chazmanOk, and do I program the actual FXS port and not the phone?
08:47.22drraycorrect
08:47.29SimoAmiok, I'm going to bed
08:47.31drrayyou make the fxs port behave how you want
08:47.36drray(Within reason)
08:47.51SimoAmigoodnight guys
08:47.58drraynight
08:47.58chazmanWell thats wierd. How would you program a FXS port? I will have some extensions here, eventually making an auto attendant answer incoming calls,
08:48.36drrayif you use the fxs pci cards, you put a line in zapata.conf and zaptel.conf for it
08:48.53drraythen you put a line in extensions.conf point to it say Zap/01
08:49.24chazmanalso, I have a small problem with Asterisk. I test everything by using a piece of software called Express Talk. The music keeps cutting out on me. Also, can Asterisk do the programming for me? I am using Asterisk@home so theres a GUI interface
08:53.10chazmanWould that do all the editing for me since thats also how I setup extensions?
08:55.56drrayI don't use asterisk@home
08:56.22mitchelocyes you do!
08:56.34drrayI do no!
08:56.37drrayshut your mouth!
08:56.40drraylies!
08:57.22chazmanOh, well it gives me everything I need and installs everything to start me off, including FreePBX, Some Flash thing, SugarCRM, ARI, etx
08:57.25chazmanetc(*)
08:59.26chazmanIt adds the extensions for me, so I assume it will do all the config for me
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09:01.12chazmanI just went into the config, and I can add what is called a Trunk
09:01.23chazmanThe default one was called Trunk ZAP/g0
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09:02.33chazmanI dont know if that is what I would do or not
09:04.50chazmanhttp://www.freepbx.org/trac
09:04.57chazmanThat is the freePBX Web Config program
09:08.13tzafrir_laptopchazman, try #freepbx
09:08.37*** join/#asterisk ghenry (n=ghenry@80.229.93.1.plusnet.pte-ag2.dyn.plus.net)
09:09.29tzafrir_laptopanyway, are you sure you're not confusing FXS and FXO?
09:09.40tzafrir_laptop~fxsfxo
09:09.42jbotmethinks fxsfxo is an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it.
09:11.46chazmanYou use a trunk to carry a call (or any number of calls) to a VSP or a device that cares about what number you send to it (eg, another Asterisk/FreePBX Machine). There are 5 types of trunks supported
09:11.52chazmanIs that what I need to do then?
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09:12.30tparcinahi channel!
09:12.46tzafrir_laptopchazman, how many phones do you need? How far are the phones from the server room?
09:13.53chazmanOk, my server room is on the top floor (2 floors). One of the phones will be 5 feet away. Another will be on the other side of the room, no more than 25 feet. Another will be 50 feet away, and then another 75 feet away. Then I will have one downstairs, approx. 125 ft total
09:14.08chazmani mean on the other side of the wall, not room
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09:15.24chazmanhere is something I found about FreePBX
09:15.25chazmanZap trunks consist of physical hardware in your machine that uses the Zapata interface. This is configured in /etc/zaptel.conf and /etc/asterisk/zapata.conf. Documentation on these files is available on the voip-info wiki.
09:16.54tzafrir_laptopchazman, ask FreePBX questions in the channel #freepbx
09:17.04tzafrir_laptoptry: /j #freepbx
09:17.06chazmannobody is responding in there
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09:18.12tzafrir_laptopI must say that I don't really understand what you're stating and what you're asking
09:20.49chazmanIn the topic bar it says I can get help with FreePBX here too!
09:20.55chazmanoh nvm
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09:23.48qdkchazman: so you read the _entire_ topic now?
09:26.00chazmanlol yeah
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09:44.05tparcinaany interesting DaPrivateeriscusion ghenryoing on hadsere?
09:44.32tparcinaany interesting discusion going on here?
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09:49.34drray"No matter how hot she is, someone, somewhere, is sick of her shit"
09:50.02tparcinayou are talking about Paris? ;)
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09:51.16Blake0psI want to setup a PRI at a business. If I installed the TE110P (single span T1) in the Asterisk box, would I need any other hardware besides phones (they will be SIP phones)?
09:51.34*** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com)
09:52.07tparcinaBlake0ps, no you wan't
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09:54.39kmilitzerHi everyone ... I have a slightly OT question. Does anyone of you know of a "clustered" Filesystem where I can store my Voicemail on, that will continue working if one Node goes down. And I don't want DRDB ... ;)
09:55.05darviriacan anyone help me with a problem of sip phones not getting any audio?  and no i'm not using nat
10:03.22MrChimpywhy not drbd?
10:03.48MrChimpythere's some oracle thingy that's a clustered fs that they've open sourced IIRC
10:07.07tzafrir_laptopgfs?
10:07.28tzafrir_laptopnot anything by oracle. More by RH
10:08.51MrChimpyi must be hallucinating ocfs then
10:09.43MrChimpynot sure if it's FOSS. not looked at it at all really, but it's what oracle punt for doing linux clusters.
10:10.13MrChimpyhttp://oss.oracle.com/projects/ocfs/
10:10.33MrChimpyI think i'd trust oracle shit more than redhat shit
10:10.38MrChimpybut it's a narrow margin.
10:13.16Blake0psthanks tparcina
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10:19.18benjkanybody here using AstLinux?
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10:29.23bikola_chey, guys, was wondering how is it possible to make outgoin calls asterisk box but recieve incoming on my SIP phone
10:29.37Blake0psbleh, I just wrote a 500 word email to a client who is about to seal the deal with me on this PBX
10:30.51bikola_cyeah, anyone lol
10:31.07bikola_cpretty quiet, for a packed room
10:37.49RoyK[uk]benjk: wtf is astlinux?
10:38.56stoffell_hbenjk: i would love to use it, but it's not bristuffed, and i'm to lazy to do it all myself :p
10:39.03stoffell_hRoyK[uk]: put .org behind it ;)
10:39.11stoffell_hand then surf it :)
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10:56.57benjkstoffel_h, that's precisely what I want to do with AstLinux, add BRIstuff
10:57.10benjkbut the damn thing is borken
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10:57.29benjkI had to set up a fake name server just so it would boot
10:57.48benjkcause it wasn't finding it's NTP server
10:57.56knarflybikola_c: I don't know you can do that...the phone will want to register with the sip proxy or the * box....why not bring your sip traffic in thru * too?
10:57.57benjkall sorts of nonsense
10:58.13benjkand the main configuration file where you configure the thing
10:58.19benjkwell, that file is volatile
10:58.28benjkreboot and all your edits disappear
10:58.45benjkand of course there is zero documentation
10:59.00daysmen3can anyone provide some pointers regarding the setup of asterisk for 20 users using sip and pstn lines.
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10:59.36knarflydaysmen3: what you wanna know
10:59.48daysmen3my question is regarding the asterisk server and additional components necessary to run asterisk well -
11:00.28knarflydaysmen3: 20 users will need a little more horsepower...what's under the hood?
11:01.10knarflydaysmen3: 20 users is no problem though
11:01.11daysmen3for instance what tool would you use for monitoring asterisk or what tool would you use to look at call stats etc etc - when of the free guis at there
11:02.04benjkdaysmen3: www.voip-info.org
11:02.14knarflydaysmen3: the free gui's I've seen out there are not much to write home about....the commercial ones offer more if your in beez-ness and need to track calls
11:02.28MrChimpywrite yer own, you big girly-man
11:02.31MrChimpy:)
11:02.49daysmen3wish i could but programming is not my strong point - how about astbill
11:03.56drraywhy not just spend some time learning "awk"
11:04.06daysmen3ok monitoring would not be an issue and scripting for automatic recovery would be would be easy
11:04.07Dr-Linux|workwhat does it mean? Jul 14 02:31:03 NOTICE[24011]: chan_sip.c:6275 check_auth: stale nonce received from '4073<sip:4073@70.89.66.122>'
11:04.43MrChimpymeans a pederast was trying to use your sip server for grooming kids
11:04.57daysmen3i know awk a little im just wondering what the best setup would be -
11:07.17daysmen3knarfly: -do you allow users to customise the dialplan or do you do that yourself
11:10.44MrChimpywhat users?
11:11.07MrChimpyas in user users? the gimps on the end of the phone?
11:11.43MrChimpyi'd expect most people to run away screaming on sight of a dialplan. the ones that don't do that immediately will invariably break it for you.
11:13.15daysmen3i know yu wouldnt let them touch extensions.conf ;-) however would you provide a webgui to a sys admin at the company to do it
11:13.40daysmen3or get them to contract you out at mega bucks to do it yourself -
11:13.51MrChimpyah. well, it's possible.
11:14.00MrChimpydepends how kind you're feeling :)
11:14.31MrChimpybut if you're doing that stuff it's probably best using macros and DB and not altering extensions.conf directly
11:14.54MrChimpythough I haven't done all that stuff. my asterisk systems aren't PBX type things, they're IVR
11:15.02daysmen3ok im new to asterisk and im simply gaging best practices
11:15.18MrChimpygot the o'reilly book?
11:15.40*** join/#asterisk Gunnar (n=gunnar@62.97.242.6)
11:15.45daysmen3yep downloaded the pdf??
11:15.52MrChimpyit's worth a read. it's good.
11:15.54daysmen3a few months ago
11:15.58daysmen3OK
11:16.03MrChimpyi like my hardcopy version. it's well thumbed.
11:16.03daysmen3ill go back over it
11:16.44daysmen3im get my teeth into again  - cool
11:16.50MrChimpyif it's a simple install asterisk@home or one of those things might do and give you a gui with it
11:17.23daysmen3would love to do it manually incase i get into some problems that needs debugging
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11:22.21daysmen3MrChimpy: youve been great - what do you think of Asterisk2Billing - anyone
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11:33.32*** join/#asterisk cvv (n=cvv@212.8.35.34)
11:39.23cvvwhy there may be unidirection connection in local network?
11:39.31cvvprotocol - sip
11:39.42cvvcodec - alaw
11:39.53cvv(G711a)
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11:40.22cvvsip-client: X-Lite
11:40.26CMikehi all
11:41.08cvvHi!
11:50.19tparcinaforwarding of all incoming calls, has anybody done it?
11:51.37tparcinai'm planing to done it puting data into asterisk internal DB, so extension that needs to be forwarded will have something like this in * DB - forward/ext_no : forwart_to_ext_no
11:52.12af_I have I have a tdm400, connected to a legacy pbx with 2 fxo. hangup on thos lines is not detected. what could be?
11:52.37drraythe line type does not support it?
11:52.48af_voice is just fine
11:52.58af_does not support hangup?
11:53.05drraysure, but the tdm can't tell the line has let go
11:53.07tparcinanow, i'm havin trouble puting this data indo asterisk DB
11:53.13af_I think is very basic requirement.....
11:53.21drraywhat type of lines does the PBX have?
11:53.40af_well I did put fxs_ks signaling
11:53.43drrayit could be that it is not set right
11:53.53af_fxs_ls?
11:53.57drrayI don't know
11:54.04af_how could I detect?
11:54.08drrayyou need to find out what your PBX offers
11:54.22af_I have jus the line no doc
11:54.33drrayyou have the internet
11:55.06drraywhat type/brand/model pbx are you hooking up to?
11:55.08*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
11:55.11af_no idea
11:55.23drraymaybe you should find out?
11:55.54af_I can't even enter in the pbx room
11:55.58drrayI mean, you could randomly change signalling types until it worked
11:56.05drrayassuming that is the problem
11:56.10af_oh yeah. I now how to do that
11:56.25af_I think not any is good for this kind of link
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12:02.00af_http://kb.digium.com/entry/1/30/
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12:11.41e-ddieanyone got any ideas of how to make our * server talk the same language, both when calling to the phones directly, and when calling to the queues?
12:12.00e-ddielike, the language is set to the same language in both iax and sip.conf
12:12.10e-ddieand it only happens while in queue
12:12.22e-ddieor when you call the queue nr's
12:13.15E-bolaJul 14 16:11:18 WARNING[6363]: pbx.c:4796 ast_add_extension2: Unable to register extension 'jonas', priority 1 in 'incoming', already in use
12:13.21E-bolais this normal when reloading?
12:13.42E-bolai have 1 extension defined, if i type reload without changing anything it warns me that it already exists?
12:14.48fileis priority 1 already in use in your config...
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12:16.23E-bolafor that extension?
12:16.35[TK]D-FenderE-bola : pastebin the context if you can't see it for yourself.  * doesn't make that sorta stuff up....
12:16.49E-bolawell im reading the asterix book made by the community
12:16.55E-bolaand the first example of a dialplan dont work for me
12:17.07E-bolai made new sip.conf and new extensions.conf
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12:18.33[TK]D-FenderE-bola : It wasn't make by "the community", but rather by a select FEW who happen to be ACTIVE in it.
12:19.05E-bolahttp://pastebin.ca/88008
12:19.19MrChimpyif everyone wrote it it'd be shakespeare written by infinite monkeys
12:19.24MrChimpybut no use for asterisk
12:19.25[TK]D-FenderE-bola : Please observe lines 8 & 9
12:19.36[TK]D-FenderE-bola : BOTH PRIORITY 1
12:19.43[TK]D-FenderE-bola : Didn't take 2 seconds
12:19.47E-boladoh lol
12:19.57E-bolait was so short i asumed i hadnt made errors hehe
12:20.07[TK]D-FenderTrout.. its not just for breakfast anymore!
12:20.08E-bolai read u can just use n instaid of sequential numbering?
12:20.11E-bolais that bad habbit?
12:20.43[TK]D-FenderE-bola : Depends... it lets people who don't pay attention enough as it is take even MORE for granted :/
12:21.07[TK]D-FenderE-bola : I don't use "n", and won't for the forseeable future
12:22.03E-bolak
12:22.30E-bolahmm i got my softphones registered and the 3 line extension as pasted but correct to extension s, and with correct priorities
12:22.41E-bolai thought i could then just dial any number on my softphone and hear the audio file?
12:22.42*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)
12:23.26E-bolathe example is for zap channels, im using sip so maybe im missing something?
12:25.12*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
12:25.12*** mode/#asterisk [+o anthm] by ChanServ
12:25.33[TK]D-FenderE-bola : SIP always dials an explicit #.  If you were to LITERALLY put "s" in your dial line it should work.  if you want a catch-all for NUMBERS, you should use something like _X.
12:25.50[TK]D-FenderE-bola : Go read the book again about dial patterns!
12:26.28memicanybody can tell my why ChanIsAvail doenst check if an iax2 peer is reachable?
12:27.03E-bolathanks [TK]D-fender: _X worked
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12:33.36[TK]D-FenderE-bola : np.  Keep in mind the book is a guide to the concepts of * and not to be hand typed line for line.
12:34.09[TK]D-FenderE-bola. read up on the different line techs and the methods you would use to identify and route calls coming from them.
12:38.37E-bolawell i found out i had to read something heavy
12:38.52E-bolacuz after spending 3 hours looking at stuff it was sill overwhelming
12:39.20E-bolaim quote supprised at how complex * is, its more complex than both apache, samba and nagios to get an overview of
12:39.35E-bolamaybe it would have helped if i knew the least about telephony hehe
12:39.40mut^^^
12:39.54mutholy crap i am pooped
12:40.00muti finally got to move into my house last night
12:40.09mutso i was moving from 2 til 11pm last night
12:40.20*** join/#asterisk ariel_ (n=Ariel@70.46.87.158)
12:40.23mutdidn't get much sleept cause my legs kept cramping up and woke up at 5 for work
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12:43.38ariel_morning everyone
12:45.46tparcinamorining ariel_
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12:46.04*** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.212)
12:46.12Kernel_corehi all
12:46.14tparcinacan anybody explain this to me - http://pastebin.ca/88028 - why this doesn't get into * db?
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12:47.16Kernel_corewhich version of SPANDSP ( soft-switch.org ) Fax solution , is compatible with Asterisk 1.2.9 ?!
12:47.35Luke-Jrso anyone know a good origination provider?
12:47.59Kernel_coreLuke-Jr: I know
12:48.33*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
12:48.38Luke-JrKernel_core: who?
12:48.50tparcinaanybody knows the reason why some data doesn't get into asterisk db?
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12:49.42cvvevening ariel_
12:51.05ariel_damm network is giving me shit today...
12:51.26tzafrir_laptopKernel_core, latest 0.2
12:51.35tzafrir_laptop0.2.26pre
12:51.45tzafrir_laptop0.0.2, that is
12:52.23Kernel_coretzafrir: I just compiled spandsp-0.0.3pre22.tgz and copied apps_rxfax.c and apps_txfax.c to asterisk
12:52.31Kernel_corewhen I compile I get this error :
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12:54.13Luke-Jrso anyone know a good origination provider in the US?
12:54.35tparcinacan I put variable to asterisk DB? I mean Set(DB(forward/${CALLERID(number)})=${FORWARD})
12:55.01Kernel_coretzafrir: I get this error http://pastebin.ca/index.php
12:55.18tparcinathis ${CALLERID(number)} i know it works for sure, but is my problem because of second variable? - =${FORWARD}
12:56.06Qwell[laptop]tparcina: will work fine
12:56.38ariel_Luke-Jr, that is a loaded question. But I use Voicepulse.com and Race.com as my primary providers for customers.
12:57.16Kernel_coretzafrir_laptop : I get this error http://pastebin.ca/88035 ops
12:58.09tparcinaQwell: well i have some problem and i don't know whay it doesnt work.
12:58.32memicanybody knows why ChanIsAvail(IAX2/ast-int:xxxx@ast-rgbg|js) is not working? this gives me "IAX2/ast-int:test1234@ast-rgbg|js") for the channel to check
12:58.43Qwell[laptop]tparcina: set ${DB(blah)}
12:59.13memicbut should be IAX2/ast-int:test1234@ast-rgbg only
12:59.15Qwell[laptop]wait, no :p
12:59.16*** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin)
12:59.39memicthe |js is the option for chanisavail
13:00.52memicideas?
13:01.34Qwell[laptop]memic: Show the exact line you're using
13:01.49memicexten => _9X.,2,ChanIsAvail(IAX2/ast-int:test1234@ast-rgbg|js)
13:02.39memicbut |js is not recognized as option for chanisavail
13:02.42Qwell[laptop]And the line in the CLI says?
13:03.30memic-- Executing ChanIsAvail("IAX2/memic@memic/3", "IAX2/ast-int:test1234@ast-rgbg|js") in new stack
13:03.38Qwell[laptop]So what is the problem?
13:03.49memicJul 14 15:03:06 WARNING[12477]: chan_iax2.c:2215 create_addr: No such host: ast-rgbg|js
13:03.51memicthats
13:04.08Qwell[laptop]That's a little better
13:04.14memichost should not contain options for chanisavail
13:04.23Qwell[laptop]That's interesting
13:04.46memicyea? %)
13:05.04Qwell[laptop]what version of *?
13:05.04Luke-Jrariel_: do they offer origination?
13:05.12Luke-Jr(Race, that is)
13:05.16ariel_yes
13:05.23Dr-Linux|worki have installed Sphinx3
13:05.30memicAsterisk 1.0.9-BRIstuffed-0.2.0-RC8n built by root@asterisk on a i686 running Linux
13:05.34Dr-Linux|workanybody knows Sphinx?
13:05.52*** join/#asterisk beyond (n=beyond@200.192.160.100)
13:05.57Qwell[laptop]memic: and does `show application chanisavail` show s and/or j as valid options in 1.0?
13:06.02Qwell[laptop]I'm guessing no
13:06.12*** join/#asterisk m4rkl4r (n=markp@66.129.95.30)
13:06.14memichu you are asking question :P
13:06.22memicmh
13:06.38memicyou could be right, i should upgrade *
13:06.45Qwell[laptop]Yes you should
13:06.51Dr-Linux|workQwell[laptop]: ever you use Sphinx voice recognition system?
13:07.02Qwell[laptop]Dr-Linux|work: yes, but...no, I can't help with it
13:07.06Luke-Jrariel_: any idea where I can find info on it? their main page only seems to have retail services
13:07.24*** mode/#asterisk [+o Qwell[laptop]] by russellb
13:07.28Qwell[laptop]!
13:07.44russellb:)
13:07.44ariel_give them a call. Ask for Carlos the also have a var user section.
13:07.44Qwell[laptop]russellb: I have to disconnect in a minute though :p
13:07.45Dr-Linux|workQwell[laptop]: i just wanna know, if sphinx3 works
13:07.58russellbQwell[laptop]: lame
13:08.07fileworks is relative
13:08.10*** mode/#asterisk [+o Corydon-w] by russellb
13:08.19Luke-Jrariel_: var user? O.o
13:08.38ariel_they call it partners
13:08.48Dr-Linux|workfile: why i can't find any document for Sphinx?
13:09.06Dr-Linux|workhhm..
13:09.15Dr-Linux|worklooks like i asked wrong question :S
13:09.16fileDr-Linux|work: because nobody has written one? maybe they gave up?
13:09.35*** join/#asterisk franekstein (n=coast@coasta.ca)
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13:10.00[TK]D-Fendertparcina : "db" is a function and is CASE-SENSITIVE.  It must be in lower-case.  That is your problem.
13:10.05Dr-Linux|workfile: file i just wanna make sure if it works just fine with asteirsk, then i can do some effort. but if it is not good, then i should not waste my time.
13:10.26Qwell[laptop][tkYou smoking crack? :p
13:10.45fileit's not a commercial product... so it won't be *that* good
13:10.50filebut apparently it works semi-ok
13:10.50Qwell[laptop]tparcina: I question his sanity...DB is upper case
13:11.07fileand I do believe there is a guide out there somewhere
13:11.17*** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.92.Dial1.SanJose1.Level3.net)
13:11.59franeksteinhey guys I have just done some major version jumping :( and having some voicemail issues are there any updated voicemail docs ?
13:12.47franeksteinI am still using static config will go realtime at a later date
13:12.47*** join/#asterisk ghecken (n=info@pd95b1af6.dip0.t-ipconnect.de)
13:13.16franeksteinbut for now just need to make my existing entries work again
13:15.33franeksteinI have tried may different combonations such as exten => XXXXXXXXX,3,Voicemail,uXXXXXXXX Voicemail(XXXXXXXX@context)
13:15.43franeksteinis there a new proper format
13:16.16*** join/#asterisk CleanerX (n=nix@p54A38C2B.dip0.t-ipconnect.de)
13:18.28russellbfranekstein: you probably need to add searchcontexts=yes in the [general] section of voicemail.conf
13:18.46Qwell[laptop]or use the proper context..
13:18.52russellbQwell[laptop]: or that
13:18.58russellbi was going for the quick solution.
13:19.02Qwell[laptop]heh
13:19.28*** join/#asterisk littlejohn (n=little@host63-66.pool8716.interbusiness.it)
13:19.54franeksteinis searchcontexts=yes new ?
13:20.47*** join/#asterisk Laerte (n=bho@217.221.36.10)
13:21.27franeksteinwell ading searchcontexts=yes worked
13:21.29*** part/#asterisk littlejohn (n=little@host63-66.pool8716.interbusiness.it)
13:21.34franeksteinadding even
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13:29.49jetaway2009hihi
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13:30.48clive-does anyone have any pointers for me how to create a symbolic link to a directory (as opposed to a file)
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13:32.31CleanerXclive-, exactly the same
13:33.38clive-Cleaner hi, I keep creating the directory which doesnt opoint to the required directory, but to one level below
13:34.10CleanerXln -s /home/sourcedir /home/targetdir
13:34.20*** join/#asterisk MikeJ[Laptop] (n=vircuser@c-24-13-240-121.hsd1.il.comcast.net)
13:34.32clive-This is what I typed: ln -s /usr/src/kernels/2.6.9-34.0.2.EL-smp-i686/ ./linux
13:34.35CleanerXls -l targetdir give you contents of source dir
13:34.49CleanerX+s
13:37.15clive-bingo,,,thanks Cleaner
13:38.00*** join/#asterisk kram (n=mark@pdpc/sponsor/digium/kram)
13:38.00*** mode/#asterisk [+o kram] by ChanServ
13:40.14clive-hi Kram, greeting from south africa
13:41.00clive-X-gen...hey, you tsotsi
13:41.16X-Genclive-, i think kram is a bot, never seen IT speak in here
13:41.20X-Gensee
13:41.53clive-kram is the man, he is alive and kicking:)
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13:42.45justlee7does anyone here know the current address of Asterisk's CVS server?
13:43.15clive-justlee does CVS still work ?
13:43.52nortexjustlee7, I think it has been replaced by the svn server
13:44.07justlee7ah, that's why i get a host unknown
13:44.17justlee7bah
13:44.21justlee7ok thanks guys
13:44.51Assidwoohoooo
13:44.56Assidwassup peeps
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13:49.07JffMRIIIhello all
13:49.22JffMRIIIcan anyone help with a cisco upgrade 7960
13:49.44JffMRIIIcurrently on default firmware 5.0 (3.0)
13:52.29Luke-Jradvise: avoid iConnectHere at all costs
13:52.57AssidLuke-Jr: why whats up
13:53.53Assidwhat happened with them?
13:54.09Luke-JrAssid: well, first a few months ago their VoIP was broken and would only forward my calls
13:54.26Luke-Jrnow they've deleted my number (without telling me at all) because I "haven't paid since April"
13:54.33Assidyou had an incoming line with them?
13:54.38Luke-Jryet my bank statement shows the charges and all
13:54.39Luke-Jryes
13:54.44Assidhrmm
13:54.54*** join/#asterisk ph|ber (n=phiber@slackwaresupport.com)
13:55.00Luke-Jroh, and it appears their website no longer functions without Internet Explorer
13:55.12*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
13:55.12Luke-Jr*and* their chat support doesn't even work with XP
13:55.19Assidhhahaaha..  well.. in their defence .. they still love ie
13:55.20Luke-Jrso I had to find a neighbor running 98
13:55.23ph|beranyone run into problems installing a phone on astgui??
13:55.27Assidyou kidding me
13:55.34Assidyou need 98 to run their chat?
13:55.59[TK]D-FenderLuke-Jr : You forgot to mention that its also on on Tuesday nights it its raining ;)
13:56.01Assidokay. you win.. not toucing them..
13:56.19Assid[TK]D-Fender ?
13:56.20Qwell[laptop]Maybe it's just you
13:56.28Assidyeah maybe they dont like you
13:56.35Luke-JrAssid: yep
13:57.00Luke-Jrnow their support guy is trying to say $3 of outgoing calls is paid for by two $10 charges over 2 months
13:57.02Assid[TK]D-Fender: told them about the TDM.. will let you know what they say
13:57.49Assidhrmm.. anyones sipdiscount accepting dtmf?
13:58.48Assiderr.. anyone got an ivr i can test this on?
13:59.31nortexJffMRIII, Can you explain the problem you are having while upgrading?
13:59.46*** join/#asterisk RoyK[uk] (n=roy@83.105.70.179)
14:00.35*** part/#asterisk tparcina (n=tparcina@lns02-1906.dsl.iskon.hr)
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14:01.44GerbilWrkCan anyone recomment the best way to rotate the Master.csv file?
14:01.58Assidcrontab
14:02.10snowy_owlhi fellows. Im using grandstreams devices (HT 486) to make calls, OpenSER and Asterisk like a rtp proxy. Im receiving this message: chan_sip.c:2530 sip_write: Asked to transmit frame type 256, while native formats is 1 (read/write = 256/256)
14:02.15Assid[TK]D-Fender: got a number i could check ivr on?
14:02.23snowy_owlIs this a problem?
14:02.34Assidsnowy_owl: codec issues from what i can see
14:02.55Assid256 is 729
14:03.00snowy_owlim using 729
14:03.09Assidright
14:03.18Assidbut your input is on 723
14:03.55Assideither edit your device to use 729 or .. transcode
14:04.55snowy_owlhummm... so, the HT is using 723 and asterisk is 'translating' it to 729 to send to carrier
14:05.16Assidi guess
14:05.25Assidwhatever your input is .. is on 723
14:05.27*** join/#asterisk DuRaZNo (n=durazno@201.230.129.212)
14:05.32DuRaZNoslds
14:05.40Assidslds?!?!?!
14:05.48Assidso long dumb suckers?
14:06.32nortexsuper long dial signal ???
14:06.37DuRaZNosorry
14:06.53snowy_owli'll see that
14:06.57snowy_owlthanks Assid
14:07.03DuRaZNoi speak spanish and 'slds' means something like say hi
14:07.14DuRaZNoheh :D
14:07.27nortexWell then slds to you too.
14:07.28*** join/#asterisk MikeJ__ (n=vircuser@c-24-13-240-121.hsd1.il.comcast.net)
14:07.30DuRaZNoso, i think my english is not very good
14:07.40DuRaZNohaha hi there
14:08.33*** join/#asterisk trig (n=jb@xob.neospire.net)
14:08.58RoyK[uk]afternoon
14:09.26mogmorning
14:09.27E-bolaul 14 18:07:13 WARNING[6802]: pbx.c:1700 pbx_extension_helper: No application 'MeetMe' for extension (internal, 600, 1)
14:09.27E-bola<PROTECTED>
14:09.32E-bolawhy wont meetme work?
14:09.52ph|berE-bola: an id 10 t problem?
14:10.12E-bolammm sory whats that?
14:10.24E-bolai simply put conf => 600 in meetme.conf
14:10.36E-bolaand made an extension to point to it
14:10.39Qwell[laptop]E-bola: Do you have zaptel installed?
14:10.46drrayor ztdummy
14:10.47E-bolanope, purely sip
14:10.54Qwell[laptop]E-bola: You need zap
14:10.57E-bolaahh ok
14:11.43*** part/#asterisk trig (n=jb@xob.neospire.net)
14:11.49hmmhesaysnice name
14:12.33nortexI have a asterisk server setup as a gateway to my PR?I and then IAX trunks to my call servers. Does anyone know of an easy way to combine the CDR of the asterisk servers so I can show what caller went out what ZAP channel?
14:12.49E-bolacan i get conference functionality without using the zaptel stuff?
14:12.55[TK]D-Fenderph|ber : My users have a lot of id10 t problems here too....
14:12.57nortexid 10 t, I have not seen that in years. Very true though.
14:13.01E-bolai'd prefer not to fiddle with kernel modules atm if i can help it
14:13.05*** join/#asterisk fndude (i=sobeit@63-191.126-70.tampabay.res.rr.com)
14:13.10*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)
14:15.04[TK]D-FenderE-bola : You don't need to mess with the kernel.
14:15.48[TK]D-FenderE-bola : If you're on 2.6 you can just use the RTC timer and you're set.  If not, you will need a UCHI USB interface.
14:16.26E-bolai am on 2.6
14:16.45E-bolagot a page describing how to use the rtc timer, or ?
14:17.29fndudeMy provider tells me to use two accounts on asterisk I must use 'from user' I tried to add 'fromuser=mynumber' to the sip.conf of the trunk, still a pw error, could somebody help me understand where this authentication is going wrong?
14:18.14DuRaZNoI know there are hardware voip solutions offered by Cisco, Avaya, Panasonic, etc... Do you know how much could I pay for those solutions?
14:18.23Qwell[laptop]DuRaZNo: a lot
14:18.55[TK]D-FenderE-bola : no need, it should choose it automatically.  just download zaptel, enable ZTDUMMY in it, compile & install it, then recompile *.
14:19.02*** join/#asterisk Skarmeth (n=Skarmeth@201008240231.user.veloxzone.com.br)
14:19.04Skarmethhi all
14:19.23DuRaZNowell, i know they are expensive...
14:19.27*** join/#asterisk greendisease (n=jack@fedora/greendisease)
14:19.50E-bolatk: thanks
14:19.51DuRaZNobut I don't if that means $1000, $10000 or how much
14:20.32Skarmethanybody using polycom soundpoint ip 301 and sip 1.6.6, bootrom 3.1.3 ? I have updated two phones to last sip and bootrom version, and the phones show's a icon (line icon) as a phone off-hook and like a RJ45 plug, I cant make or receive calls
14:20.59*** join/#asterisk afrosheen (n=test@txprotoa2.august.net)
14:20.59Skarmethjust updating the bootrom, works, but if I update sip software, this problem comes
14:21.47nortexDuRaZNo, When we got ready to deploy our first sites system the Cisco equivilant was around 56,000 compared to the 26,000 asterisk setup, phones and all.
14:22.26*** join/#asterisk viler (i=1000@200.114.70.228)
14:22.28E-bolaWhen ur register attemps with a voip provider times out whats the most likely reason?
14:22.43E-bolainternet works, i can resolve their hostname and ive doublechecked hostname and password
14:23.17DuRaZNo56000? omg
14:23.46afrosheenyeah Cisco charges the big bucks
14:23.50*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)
14:24.07mutcisco is usually rock solid tho
14:24.09DuRaZNoand they use propietary codecs and protocols right?
14:24.09nortexDuRaZNo, And honestly, Asterisk packed more features.
14:24.28[TK]D-FenderSkarmeth : Shat ver were you on before?
14:24.46E-bolaI have an odd request... Do anybody have a place i can register with to test my system?
14:24.49nortexCisco is switching to SIP in their lastest Call Managers and the phones already support it.
14:25.01E-bolaThe VOIP provider i signed up with times out, so im wondering if the problem is on my part
14:25.07DuRaZNoI'm reading about hardware solutions, and a lot of people say that they are limited, and I would need to pay more to get more specific services
14:25.16afrosheenthat's true
14:25.24nortexDuRaZNo, Yup.
14:25.38mutanyone know if the t3 cards will be * supported anytime soon?
14:26.03nortexE-bola, You might try http://www.iaxtel.com/
14:27.59nortexDuRaZNo, An example of that is we ask cisco about voicemail to email, pretty simple in my mind since the voicemail is on a server. The cost to add that feature for a 100 users doubled cost of the entire voicemail server. server
14:27.59*** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd)
14:28.44Assidiaxtel?
14:28.49Assidhrmm
14:28.57Assidwhich one was it which had issues? iaxtel or teliax?
14:29.15DuRaZNoi'm writing a lot of information abou software and hardware solutions to open my boss's eyes
14:29.33DuRaZNothanks everybody
14:29.33nortexiaxtel is the digium test network 700 and toll free access only.
14:29.56nortexDuRaZNo, No problem and good luck :)
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14:33.46Skarmeth[TK]D-Fender, It was 1.6.3 (SIP) and 3.1.3 (BootROM)
14:34.04*** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com)
14:34.08Skarmethwith SIP 1.6.6 I can't receive or make calls
14:34.31*** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158)
14:34.59AssidSkarmeth: does your CLI show your call being placed?
14:35.02*** join/#asterisk daysmen3 (n=primus@host86-141-242-160.range86-141.btcentralplus.com)
14:35.46E-bolanortex: great tip, just signed up
14:36.28*** join/#asterisk my007ms (n=noor@217.139.224.194)
14:38.50E-bolaDo you have something similar with sip?
14:39.03E-bolajust somewhere i can register with my asterix server to test it?
14:39.08*** join/#asterisk Egonis (n=chultay@207.245.14.10)
14:39.32EgonisI am trying to setup a multiple context zapata.conf where channels 1-3 are context1, and channel 4 is context2. How do I do this? is there a sample? I can't find one
14:40.25[TK]D-FenderSkarmeth : I'm betting your config files go mangled.
14:41.14cvvbye-bye
14:41.18*** part/#asterisk cvv (n=cvv@212.8.35.34)
14:43.15*** join/#asterisk Mercestes (n=Merceste@216.54.143.2)
14:43.21[TK]D-FenderEgonis : context = context1
14:43.30[TK]D-FenderEgonis : channels => 1-3
14:43.32[TK]D-FenderEgonis : context = context2
14:43.39[TK]D-FenderEgonis : channels => 4
14:44.09Egonisah, thank you! And that goes in zapata.conf?
14:46.20*** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com)
14:47.57*** join/#asterisk tecnico (n=tecnico@24.96.146.69)
14:48.07[TK]D-FenderEgonis : Correct.  when done in that order the other channels inherit all the characteristcs that remain unchanged (as in ALL of them) and the only thing we override is the context
14:48.43E-bolaHow can i verify a sip proxy is running?
14:48.55E-bolaremotely i mean, i cant connect to my voip providers sip proxy
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14:54.51dwrecktionQuestion: I'm on an office network that already has asterisk installed and configured and working fine.  but i'm trying to set up another asterisk test system that I can eventually move into another location.  Anybody have any recommendations/tips/suggestions on making this work as far as configuration?
14:55.31*** join/#asterisk Gr1ncheux_ (n=devine@AStDenis-105-1-22-82.w81-248.abo.wanadoo.fr)
14:56.20afrosheen<PROTECTED>
14:56.29nortexE-bola, Have you checked you firewall/router to make sure nothing is blocking the request?
14:56.37Luke-JrLooking for origination: US, Kansas City DIDs, respectable pricing (per minute or otherwise), *non* Flash website, preferably LNP
14:58.58afrosheenLuke-Jr: commpartners.us
14:59.18afrosheenor txlink.net
14:59.19dwrecktionI'm not cloning a system.  I'm designing a new system, but for now it is on the physical network
14:59.19*** join/#asterisk psk (n=psk@golia.caltanet.it)
14:59.37afrosheendwrecktion: as long as nothing is trying to register to it, it will be invisible
15:00.01Luke-Jrafrosheen: thanks
15:00.20E-bolanortex: the host is directly on the internet
15:00.30E-bolathe router is passthough
15:00.35afrosheenLuke-Jr: np..commpartners is preferred, they have insanely cheap intl. rates
15:00.46E-bolai just need some sort of tool to verify a given host runs a sip proxy
15:00.53E-bolai found a nagios plugin and that times out too
15:01.35dwrecktionso I can put another Asterisk server and SIP phones on the same network as the existing Asterisk setup and have them work independently?
15:02.06trelane_<3 handset
15:04.55dwrecktionafrosheen: so I can put another Asterisk server and SIP phones on the same network as the existing Asterisk setup and have them work independently?
15:05.42Luke-Jrafrosheen: unfortunately, CommPartners doesn't seem to cover KC
15:05.58jbalcombdwrecktion: absolutely
15:07.06dwrecktionjbalcomb: and can you summarize how?
15:07.39*** join/#asterisk kiddy (n=achu@59.93.32.89)
15:07.53kiddyHow can I connect two asterisk server's ?
15:08.12Luke-JrIAX2
15:08.28jbalcombdwrecktion: did you set up the first system?
15:08.57dwrecktioni didn't, but i feel like i've got a fairly decent understanding of how its set up
15:09.15kiddyLuke-Jr : can you pls give me the url where I can find the configuration of interconnecting two servers ?
15:09.34*** join/#asterisk hohum (n=dcorbe@12.195.58.235)
15:09.46jbalcombdwrecktion: give your secondary asterisk server a different IP and configure the phones with that IP as thier SIP server. done.
15:10.02*** join/#asterisk arguile (i=user224@66.38.201.234)
15:10.51jbalcombdwrecktion: you might consider using a different private class bock for clarity and/or VLANs for traffic segmentation
15:11.14*** join/#asterisk blaylock (n=sfv100@68-69-102-120.chvlva.adelphia.net)
15:11.20dwrecktionjbalcomb: does SIP server = IP Gateway?
15:12.10jbalcombdwrecktion: i would think IP gateway would be your router. where do you see this term?
15:12.17blaylockanyone using the new TE412P or TE407P cards?
15:12.20*** join/#asterisk jetaway2009 (n=asd@218.111.10.56)
15:12.49blaylockwell maybe not new really
15:13.08jetaway2009hui
15:13.10dwrecktionjbalcomb: this is on the phone.  i've trying to figure out how to configure the phones to connect to the SIP server
15:13.35*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)
15:13.38afrosheenLuke-Jr: I just checked commpartners website and I have a whole list of KC area codes
15:14.37Luke-Jrafrosheen: weird... I searched :/
15:14.45Luke-Jrand go "No records were found"
15:15.37afrosheenI clicked service area then the link for kansas
15:16.27Luke-JrWhere is that?
15:16.48*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)
15:16.56*** join/#asterisk SexyKen (n=Ken@c-71-202-149-39.hsd1.ca.comcast.net)
15:17.19mutanyone know if the t3 cards will be * supported anytime soon?
15:17.34SexyKenHey guys -- currently my Asterisk setup has some Voicemail configs to actually allow for over the phone voicemail - but some of them delete the file from the server and have it sent via e-mail.
15:17.46SexyKenI'm wondering if there is anyway to get it sent in MP3 format instead of WAV
15:21.04*** join/#asterisk TeePOG (n=temp@dsl-145-178-200.telkomadsl.co.za)
15:21.18*** join/#asterisk pdtmobile (n=ptinsley@209.12.249.243)
15:21.32*** join/#asterisk Eggplant (i=No@dsl-216-155-214-162.cascadeaccess.com)
15:21.41*** join/#asterisk Precion (n=crhodes@adsl-75-7-75-29.dsl.milwwi.sbcglobal.net)
15:21.49TeePOGhi guys
15:21.53*** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com)
15:22.04Luke-JrSexyKen: MP3 isn't ideal for voice
15:22.19SexyKenMP3 isn't ideal for voicemail?
15:22.27Luke-JrMP3 is designed for music, not voice
15:22.57Luke-Jrand even then, Vorbis is better ;)
15:23.09SexyKenRight but in my situation, MP3 would be useful.;
15:23.15SexyKenSo that still doesn't answer the question I had.
15:23.18Luke-JrWhy not use speex? :)
15:23.20SexyKenOr did I miss your answer?
15:23.32Luke-JrMP3 encoding would probably be a patent violation, so unlikely
15:23.50Qwell[laptop]among other things
15:23.54SexyKenExplain
15:24.08Qwell[laptop]SexyKen: MP3 is a patented tech...
15:24.19Qwell[laptop]You need to have licenses in order to do so (legally)
15:24.49SexyKenOh goodness -- I better call someone and tell them tha 80% of the internet is illegally encoding MP3's!!!
15:24.59Qwell[laptop]SexyKen: Yes, this is quite true
15:25.03*** join/#asterisk Blaze312 (i=Blaze312@24-247-183-114.dhcp.aldl.mi.charter.com)
15:25.10Blaze312just a quick question
15:25.44Luke-Jrlol
15:25.51Blaze312im a total newb
15:25.53*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
15:25.55jetaway2009rhrhr
15:25.58jetaway2009hhehe
15:25.59jetaway2009true
15:26.12E-bolagrrr dammit
15:26.13Blaze312just wondering if avaya ip phones work with asterisk
15:26.17E-bolai finally found a free sip provider
15:26.29E-bolaand it works right away, so it turns out it WAS my payed for voip provider who
15:26.32E-bola's fucked up
15:26.52Luke-Jro.o
15:28.35Blaze312should any IP phone work with asterisk or is it limited to certain ones?
15:28.47SexyKenSIP/IAX phones should work.
15:28.54SexyKenGo look it up you jackass.
15:29.36Blaze312i looked on the site but didnt see a list of supported phones or anything
15:29.46Blaze312this particular phone is a SIP phone
15:30.18jbroomethen you're fine
15:30.22Blaze312im very new to PBX and just learning linux as well
15:30.32*** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca)
15:30.34Blaze312just thought asking here would be the fastest way to get my answer
15:30.45Blaze312im looking to set this up when i move in a month
15:31.27SexyKenI'd be careful if I were you.
15:31.46*** join/#asterisk mmarker (n=mmarker@216.220.209.239)
15:31.50RoyK[uk]hm.. anyone that knows how i can debug asterisk while it's running? it is hanging with some 'active' calls which are not active at all
15:32.12RoyK[uk]this is with some custom patches. i beleive those patches are to blame...
15:32.20rob0Anyone in UK? I am wondering about 0870 numbers. I take it that those cost a premium to call?
15:32.20[TK]D-FenderSexyKen : SHUP YUO.
15:32.46SexyKenSHUP YUO?
15:33.57rob0Blaze312: the learning curve can be steep in places and it goes a VERY long way. In general IRC is not a good substitute for learning the basics. Good luck.
15:35.04muthey tk
15:35.21mutknow if sangoma plans to make drivers for their t3 card for *?
15:35.30*** join/#asterisk I-MOD (i=opticron@68.62.165.168)
15:36.22Blaze312rob0
15:36.31Blaze312thanks for the info. i know what you mean
15:36.39Blaze312i was just coming here to get a couple of quick answers
15:37.01Blaze312ill probably spend time actually learning by looking through the documentation and tutorials
15:37.41MrChimpyhmm.
15:38.01MrChimpyi need to be able to run lots of concurrent perl AGIs
15:38.13MrChimpyis there anything like FastCGI?
15:38.33rob0Blaze312: the Wiki has a lot of good stuff.
15:38.50[TK]D-Fendermut : Already have drivers for linux so * can use that card....
15:39.04mutchannelized?
15:40.33*** join/#asterisk Strom_C (n=strom@gateway.digium.com)
15:43.00*** join/#asterisk matkix01 (n=null@71-209-37-209.bois.qwest.net)
15:43.02matkix01Hey all
15:43.40Strom_Cgood morning!
15:44.34mut[TK]D-Fender: 2 A104D's in one system will work well right?
15:45.44*** part/#asterisk matkix01 (n=null@71-209-37-209.bois.qwest.net)
15:45.50*** join/#asterisk chorlick (n=Chris@63.81.26.126)
15:46.09*** join/#asterisk piper69 (n=piper69@69.155.81.24)
15:46.12[TK]D-Fendermut : Sure, but why not just buy an A108d?
15:46.22mutbecause they're new
15:46.30[TK]D-Fendermut : Channelized?  Now you're being PICKY! ;)
15:46.43[TK]D-Fendermut : Works gread for SIP though ;)
15:46.44mutwell
15:46.54*** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com)
15:47.03[TK]D-Fendermut : Translation : No idea when/if it will be channelized anytime soon
15:47.04muti'm looking to make a class 4/5 switch
15:47.10mutinstead of 100k on a new one
15:47.26muti don't need anything like that right now
15:47.44mut[TK]D-Fender: didn't know they had an a108d
15:48.06*** join/#asterisk sandra78 (n=aerae@200.106.67.49)
15:48.10sandra78hi pls
15:48.15sandra78help!!
15:48.30[TK]D-Fendermut : perfect solution. Though you CAN add 2 A104d without any real problems usually.
15:49.02sandra78i have create a context  exten => s,1,dial(zap/g0/${DNID}#,40,r)
15:49.25sandra78the SIP incomming nummber its paas fine with SIP
15:49.34sandra78but with IAX i get a empy number
15:51.07sandra78in cli console with sip i get zap/g0/639604000#|40|r but in iax i get zap/g0/639604000#|40|r
15:51.13mutwonder what kinda cpu i'de need to do simple switching
15:51.24mutwouldn't think much
15:51.41[TK]D-Fendermut : not much
15:53.42*** join/#asterisk SplasPood (n=jwb@206.252.198.101)
15:54.16sandra78??
15:54.36Strom_Csandra78: those are the same string
15:54.49*** part/#asterisk JffMRIII (n=JffMRIII@c-67-167-202-60.hsd1.il.comcast.net)
15:54.49*** join/#asterisk postel (n=jp@unaffiliated/postel)
15:55.53sandra78no, incomming number don't pass when i use iax
15:56.26Strom_Cwell, you pasted the same string for both examples
15:56.31sandra78with sip i guess dial the channel with the DNID number
15:56.43sandra78ohhh yes sorry
15:57.04sandra78i get this with iax Executing Dial("IAX2/1000-3", "zap/g0/#|40|r") in new stack
15:57.11Strom_Cwhat's on the other end of your IAX trunk?
15:57.25sandra78with sip i get this fine Executing Dial("SIP/77701-69f4", "zap/g0/439565345#|40|r") in new stack
15:57.36sandra78another asterisk box
15:57.52Strom_Cwhat's the dial string on that asterisk box?
15:58.11pdtmobilethis channel doesn't happen to get archived anywhere does it?
15:59.15sandra78the dial string it's fine i get the number filtered
15:59.20sandra78<PROTECTED>
15:59.57Strom_Cpastebin the dialplan and iax.conf on the box you're having trouble with
16:00.01sandra78i'm calling from a internal sip extension and get out with a iax trunk
16:00.05Strom_Cpastebin.ca
16:00.39sandra78i have the last asterisk release
16:00.47sandra781.2.9.1
16:00.50Strom_Cpastebin the dialplan and iax.conf on the box you're having trouble with
16:03.08sandra78what it's this¿?
16:03.20sandra78pastebin.ca??
16:03.29Strom_C~pb
16:03.30jbotsomebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
16:03.40Dr-Linux|workfile
16:03.45Dr-Linux|workguys
16:03.57Dr-Linux|workthis problem is killing me since last day ..
16:03.58Dr-Linux|workJul 14 08:48:01 NOTICE[12346]: rtp.c:564 ast_rtp_read: Unknown RTP codec 104 received
16:04.10Dr-Linux|workother end can't hear me
16:04.15Dr-Linux|workwhat could be wrong?
16:04.29Strom_Cyou screwed up your SIP configuration ;)
16:05.04*** join/#asterisk znoG (n=gs@205-17-235-201.fibertel.com.ar)
16:05.43Dr-Linux|workStrom_C: what do you mean?
16:05.58Dr-Linux|workStrom_C: do you understand what's wrong with my codecs?
16:06.13Strom_Cwhat codec are you trying to use
16:07.00*** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70)
16:07.12jetaway2009could someone pls tell me how to extend T1 cables
16:07.24[TK]D-FenderStrom_C : Looks like video
16:07.41[TK]D-Fenderjetaway2009 : std CAT5E extension.
16:07.42jetaway2009any device to enable the cable for a long distance
16:07.45*** join/#asterisk dandan (i=dandan@pacanka.com)
16:07.51dandanhey all
16:07.52dandan:)
16:07.53Strom_Cjetaway2009: define "long distance"
16:07.55Sonderbladeis it possible to code your whole dialplan in agi?
16:08.03[TK]D-Fenderjetaway2009 : For long distances you need a repeater and will have to change your LBO
16:08.04Dr-Linux|workStrom_C: i tried all codecs one by one
16:08.08Strom_CSonderblade: if you're totally insane, yes
16:08.22dandanI already have a sangoma card and now I need to put in a x100p for overhead paging, how do I cnfigure zapata.conf for multiple cards?
16:08.25jetaway2009i read t1 cable has limitation on the length.
16:08.32SonderbladeStrom_C: why insane?
16:08.35Strom_Cjetaway2009: hence the need for a repeater
16:08.47dandanjetway: yes it does which you compensate on the hardware
16:08.48Strom_CSonderblade: why the hell would you want to do the whole thing in AGI?
16:08.51iCEBrkrjetaway2009: stretch them! :D
16:08.55mutPRAISE THE LORD FOR BACKUPS!
16:08.59Dr-Linux|workStrom_C: on priority bases, like g729, g723, ulaw , alaw and ilbc
16:09.06SonderbladeStrom_C: because asterisk's dialplan language sucks
16:09.09jetaway2009how much for the repeater
16:09.20jetaway2009could u recommendation specific brand
16:09.25Strom_Cadtran
16:09.30[TK]D-FenderSonderblade : Works fine for 99% of needs.  What are you having trouble implementing in it?
16:09.45jetaway2009is repeater=channel bank?
16:09.48Strom_Cno
16:09.51Strom_Crepeater = repeater
16:09.52[TK]D-Fenderjetaway2009 : No idea.  You'll have to call a place that sells them
16:09.55dandanI already have a sangoma card and now I need to put in a x100p for overhead paging, how do I cnfigure zapata.conf for multiple cards?
16:10.02Sonderblade[TK]D-Fender: MOST pbx:es works fine for 99% of needs
16:10.09Strom_Cdandan: don't ask the same question twice in two minutes
16:10.12[TK]D-Fenderjetaway2009 : Channel bank CAN be a repeater, but no, you typically buy a little box to repeat the signal on.
16:10.32[TK]D-FenderSonderblade : So what exactly are you trying to do that make you want to use AGI instead?
16:10.35jetaway2009will i able to google with term repeater and t1
16:10.35dandanstrom: I didn't even hear no idea... go google or anything
16:10.42jetaway2009is it the correct term
16:10.52Strom_Cdandan: if no one wants to help, no one will say anything
16:10.58Strom_Cdandan: welcome to #asterisk
16:11.03Sonderblade[TK]D-Fender: i want to log each incoming and outgoing call in a db
16:11.11Strom_CSonderblade: that's easy
16:11.18Dr-Linux|workStrom_C: no clue?
16:11.19Strom_CSonderblade: asterisk already does that in call detail records
16:11.23dandanstrom: heh
16:11.30Strom_CDr-Linux|work: i dont feel like debugging it right now
16:11.42jetaway2009whats the different between cat5 and T1 cable
16:11.45jetaway2009look the same
16:11.51jetaway2009with exact 8 pin
16:11.55Strom_Cjetaway2009: T1 can run over cat5
16:12.05Strom_Cjetaway2009: ethernet and T1 use different pairs
16:12.17SonderbladeStrom_C: its more advanced than that, i want to log each state change for each sip device connected to the asterisk
16:12.17Strom_Cethernet uses pairs 2 and 3, T1 uses pairs 1 and 3
16:12.47jetaway2009but the cable are the same in size ..but different pairs...is it true
16:12.57Strom_Cjetaway2009: that's what I just said
16:13.21Strom_CSonderblade: what do you mean?  you want to log each keypress?
16:13.47SonderbladeStrom_C: each device can AFAIK be either InUse, Ringing, Unavailable, Idle each time a device changes its state i want to log the change
16:13.53jetaway2009yes...just rechecking my understand...
16:13.58Strom_CSonderblade: /why/?
16:14.00jetaway2009reconfirming
16:14.03jetaway2009:)
16:14.26jetaway2009how is T1 able to accomodate alot extension with only 8 pin..
16:14.41Strom_Cjetaway2009: it only uses four wires
16:14.42SonderbladeStrom_C: to monitor how the pbx is used
16:14.50Strom_Cjetaway2009: look up time division multiplexing
16:15.09Strom_Cjetaway2009: it's this really amazing new stuff that's only been around since the 1950s
16:15.26SplasPoodDoes a module for asterisk exist that can do direct XMLRPC calls to a remote service?  (similar to CURL()..)
16:15.38Strom_CSonderblade: I still don't get what the end purpose of doing so is
16:15.38SonderbladeStrom_C: which extension does the most calls? which extension gets the most calls? etc, you need advanced logging to answer such questions
16:15.47jetaway2009oh..thanks...a will search for that term...u seem to havee alll the answerss...
16:15.56Strom_CSonderblade: you can figure all that out by parsing the call detail records
16:15.56jetaway2009for everythin...amazing..
16:16.08Strom_CSonderblade: have you /looked/ at the call detail records asterisk generates?
16:16.41Dr-Linux|workStrom_C: wtf, again stopping/restarting asterisk resolved my problem :S
16:16.43Dr-Linux|workwhat's this
16:17.05jetaway2009what if the effect if i run T1 over cat5
16:17.14Strom_Cjetaway2009: T1 runs over cat5 cable
16:17.15SonderbladeStrom_C: i had no idea it logged how much a device was ringing
16:17.33SonderbladeStrom_C: nor did i know that it could log attemts to reach a device when that device is unavailable
16:17.37Strom_CSonderblade: the asterisk CDRs are fairly detailed
16:17.40sandra78i have this http://pastebin.ca/88168
16:17.43sandra78issue
16:17.49jetaway2009so...there is no such thing as T1 cable...so when i shop around ..just buy cat5 cable
16:18.06jetaway2009then why in the shop classified as t1 and cat5 cables..
16:18.09SonderbladeStrom_C: or that it can log calls that end with a busy tone etc..
16:18.10Strom_Cjetaway2009: yes, the only thing you need to worry about is getting a T1-specific crossover cable
16:18.21Strom_CSonderblade: yes, that's in the CDR
16:18.38RoyK[uk]jetaway2009: you can use anything for T1 cabling. it's not like it's really high speed
16:19.14jetaway2009oh..thanks...strom C...u had cleared a lot of doubts over the cabling,,bugging me for some time
16:19.26jetaway2009<RoyK[uk]> : thanks men
16:19.27Strom_Cthats why I'm Strom Carlson
16:19.38Strom_Ctelephone deity
16:19.45RoyK[uk]jetaway2009: i just use a normal cat 5
16:19.47SonderbladeStrom_C: and i need to display the data in realtime
16:19.48jetaway2009:)
16:19.48Strom_C[fanfare]
16:19.49RoyK[uk]just another colour
16:19.58RoyK[uk]~strom_c
16:20.16jboti heard strom_c is just some nub
16:20.16Strom_CSonderblade: so you use the asterisk mysql cdr addon
16:20.17*** join/#asterisk kc5cqm (i=mwilliam@2002:a55f:d1d:0:0:0:0:1)
16:20.25jetaway2009oh..i been looking the whole day..the different between them..oculd have just asked here earlier
16:20.31kc5cqmhowdy
16:20.55Strom_CSonderblade: or you write an AMI program that monitors the status of all channels in use
16:21.07*** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca)
16:21.32*** join/#asterisk hess\n (n=hess@2084452.cps.virtua.com.br)
16:21.37Tall-guyAny "eyebeam" users hanging about?
16:21.59dandanI already have a sangoma card and now I need to put in a x100p for overhead paging, how do I cnfigure zapata.conf for multiple cards?
16:22.11kc5cqmis there a way to transfer to another extension without putting the other party on hold?  Something similar to an analog-equivilent of having both phones off hook at the same time?
16:22.25SonderbladeStrom_C: yes, except that polling is very inefficient
16:22.26Strom_Cdandan: it's really easy, but i recommend you use something less abysmally bad than an x100p ;)
16:22.33kc5cqmI want to switch between my cordless and corded phone without the other party knowing.
16:22.44dandanstrom: my valcomm unit has an fxs interface
16:23.00*** join/#asterisk matheusbh (n=pankz@200150009221.corp.wayinternet.com.br)
16:23.03dandani just need to configure x100p on * to once in a while talk to the shop floor :)
16:23.07kc5cqmI'm using an x100p...works fine for home use
16:23.20Qwell[laptop]dandan: You'll need an fxs for that
16:23.22jetaway2009pls suggest a solution..in a 8 floor building..i plan to install asterisk server on the 5 floor.and intend to share with my firends with analog channel bank..the issue is the cabling.how to supply the phone cable to 1st floor without line derioation ...as the cable has limitation in distance..isthe repeater the best solution?
16:23.24kc5cqmactually have it plugged into my packet8 sip adapter
16:23.26matheusbhhello all. somebody knows where i can found a tutorial of how i can use a modem to be a FXO?
16:23.42dandanQwell[laptop]: huh? yesterday I connected a regular phone since valcomm generates a dialtone
16:23.47Qwell[laptop]matheusbh: You don't
16:23.48dandanand was able to talk to them
16:23.51sandra78pls help!!
16:24.00Strom_CSonderblade: well then perhaps you should talk to the guys in #asterisk-dev and see if there's a way you can hook into asterisk more efficiently, because AGI isn't going to save you
16:24.01matheusbhpls help me too!
16:24.06sandra78http://pastebin.ca/88168 i have issues with iax incomming DNID numbers
16:24.08Strom_Coh will you all pls shut up
16:24.16Tall-guyjetaway: I have a 5km connection to my telco, doesn't degrade THAT bad :)
16:24.43dandan~books
16:24.48dandan~docs
16:24.50jbotdocs is, like, 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
16:24.53Strom_C~book
16:24.54jbotmethinks book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
16:25.03dandantx
16:25.04Strom_Csandra78: i'm looking at your pastebin now
16:25.09dandanmaybe there will be something
16:25.11sandra78http://pastebin.ca/88168 i have issues with iax incomming DNID numbers with sip works fine with iax not
16:25.20kc5cqmanyone here intereface a digium analog card to a 2-way radio?
16:25.27SonderbladeStrom_C: yes, but first maybe you should back up why you think writing the dialplan in FastAGI is insane?
16:25.27Strom_Csandra78: please stop whining so I can help you
16:25.30kc5cqmI know there's support for it.
16:25.55sandra78thanks strom_c :*
16:25.55MrChimpywhuh? there is a fastagi?
16:26.00kc5cqmsandra78, you allowing UDP traffic on the iax port?
16:26.20MrChimpygosh there is too
16:26.24MrChimpyjust what I need!
16:26.27Strom_CSonderblade: because the dialplan is not a channel driver
16:26.30sandra78yeah i have ringing in the extensions
16:26.47Strom_CSonderblade: and by writing the whole thing in AGI, you're not gaining the functionality you're looking for
16:26.48jetaway2009Mr.Strom..
16:26.50sandra784569 port
16:26.56*** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com)
16:26.56Strom_Cjetaway2009: what
16:27.18jetaway2009pls suggest a solution..in a 8 floor building..i plan to install asterisk server on the 5 floor.and intend to share with my firends with analog channel bank..the issue is the cabling.how to supply the phone cable to 1st floor without line derioation ...as the cable has limitation in distance..isthe repeater the best solution?
16:27.24sandra78i have trying to get the iax DNID number but i can't get it
16:27.35Strom_Cman, you guys whine a lot
16:27.35jetaway2009need ur expert opinion ..
16:27.59Strom_Cjetaway2009: at those distances, you're fine with regular cat5
16:28.05dandanjetaway2009: supply it digitally and install channel bank on 8 floor
16:28.07CunningPikekc5cqm: I haven't heard of anyone else doing it, but I met "Dude" and his sidekick at Astricon last year and they had a working setup
16:28.09dandanit is not that far
16:28.12Tall-guyjetaway: really dude, it's not that far....
16:28.24Tall-guy10 floors X 10 feet per floor is only 100feet!!!
16:28.29Strom_Cjetaway2009: if you really want to get fancy, do fiber to the floors and then demux to analog on each floor
16:28.30sandra78<kc5cqm> yeah i have 4569 port fordward to my asterisk box
16:28.48CunningPikekc5cqm: I'm trying to dig out his contact details.......
16:28.50dandanport 4569 tcp? udp?
16:28.57kc5cqmCunningPike, interesting
16:29.01SonderbladeStrom_C: im gaining a more convenient way to implement the functionality
16:29.03Strom_Csandra78: you didnt pastebin what I asked for
16:29.08Strom_CSonderblade: good luck with that then
16:29.25kc5cqmsandra78, you might not need to explicitly forward that port...it's UDP based and should behave on its own
16:29.39Strom_CSonderblade: I suggest you go to #asterisk-dev
16:30.39sandra78<Strom_C> http://pastebin.ca/88172
16:30.50*** join/#asterisk file (n=file@neutrino.joshua-colp.com)
16:31.34Strom_Csandra78: no, thats not what I asked for
16:31.36*** join/#asterisk Waverly360 (n=mirc@209.12.249.243)
16:31.49jetaway2009Strom_c..u are one men show...
16:31.52Strom_Csandra78: I asked for the iax.conf AND the extensions.conf of the terminating asterisk box
16:31.56jetaway2009helping all at same time..
16:31.57jetaway2009amazing
16:31.59sandra78what did you asked? you asked me by my iax.conf
16:32.09*** join/#asterisk pengyong (n=lala@218.93.68.246)
16:32.09*** join/#asterisk Waverly360 (n=mirc@209.12.249.243)
16:32.17Strom_Cjetaway2009: in sixteen minutes, I also do magic tricks
16:32.42CunningPikekc5cqm: Here we go - http://www.zapatatelephony.org/app_rpt.html
16:32.42*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
16:32.46jetaway2009great to meeet a person like u
16:32.51jetaway2009multitaskingg...
16:33.06sandra78ok
16:33.26*** join/#asterisk MGSsancho (n=user@adsl-67-127-164-167.dsl.irvnca.pacbell.net)
16:33.30MGSsanchobam!
16:33.55kc5cqmthanks CunningPike
16:34.18Tall-guyjetaway: wait till you see "Revolution" when he's on a crack-induced typing spree
16:34.29CunningPikekc5cqm: You're welcome de VA7IRL ;)
16:34.36jetaway2009:)
16:35.16Waverly360yo
16:35.16jbalcombwhen I put my phone on DND is unavail.gsm the file it plays when someone calls?
16:35.17kc5cqm;-)
16:35.49[TK]D-Fenderjbalcomb : depends on the phone.  On many you can set what SIP status it will return.
16:36.28CunningPikejbalcomb: Provided you have set your dialplan and VoiceMail() options correctly, yes. Some phones don't distinguish between busy and no answer, but most do
16:36.34jbalcomb[TK]D-Fender: hrmm.. we are doing DND server side..
16:37.06CunningPikejbalcomb: Then it's up to your dialplan to call the appropriate VoiceMail() options
16:37.30jbalcomb"DBGet(dnd=dnd/SIP/${ARG3})" and then "GotoIf($["X${dnd}" = "X0800"]?9:6)"
16:37.47*** join/#asterisk trbldwine (i=trbldwin@adam.ur.northwestern.edu)
16:38.07jbalcombIs that saying goto priority 9 if it is in DND but goto 6 if its not?
16:38.21CunningPikejbalcomb: Probably.......... :D
16:38.55jbalcombso then "s,9,Voicemail(su${ARG2})" is whats getting played....
16:39.59jbalcombso it sends them to the Voicemail function with status 'su' which is 'something unavailable?
16:40.19Waverly360So I'm getting a tiny bit irritated with my Polycom phones...
16:41.07Strom_CWaverly360: leave it t [TK]D-Fender to solve all your polycom problems
16:41.12Strom_C:)
16:41.17Waverly360heh...he couldn't last time :P
16:41.23Strom_Cwhat?!
16:41.23Strom_Cgasp
16:41.29Strom_CI, for one, am shocked
16:41.32Waverly360hah
16:41.35Waverly360Well, he was helpful
16:41.36*** join/#asterisk terrapen (n=cjs@166.70.183.108)
16:42.03CunningPikejbalcomb: 'u' is the unavailable greeting - change that to 'b' if you want the busy greeting instead
16:42.04Strom_Cbut yea, I've done installs with polycom phones, and I'm sticking with cisco
16:42.05Waverly360I'm trying to get the MyStat options working on my phones
16:42.17CunningPikeWaverly360: Won't happen
16:42.35jbalcombare do the MyStat options do?
16:42.37Waverly360CunningPike: I called Polycom tech support, and they said that the options should..
16:42.39jbalcombs/are/what/
16:42.43[TK]D-Fenderjbalcomb : Not familiar with that method if its supposed to be something more that just a value YOU set.  Phone isn't supposed to communicate DND except as a response to a call.
16:42.47CunningPikeWaverly360: On Asterisk??
16:43.11Waverly360CunningPike: Not on asterisk persay, but on other pbxs
16:43.25jbalcomb[TK]D-Fender: yeah, we are setting the dnd variable in the asterisk DB so that we can check it in our dialplan
16:43.31CunningPikejbalcomb: They allow you to set 'Out to Lunch' on your phone and have other phones display that status for your line
16:43.35Waverly360CunningPike: But that means the MyStat options are supposed to send something to the PBX
16:44.01Waverly360CunningPike: I just can't get my phones to send anything..and I can't figure out how trick the phones into thinking asterisk supports it.
16:44.22CunningPikeWaverly360: Haven't we had this conversation before? :D
16:44.23Waverly360CunningPike: I'm wondering if maybe there's an option I can set in the phone.cfg or sip.cfg file that'll turn that feature on in the phone..
16:44.23jbalcomb[TK]D-Fender there is some hassle going on right now with are queues setup. they just realized after 9 months that no one set the unavailable message on are rebates dept. call queues.
16:44.34Waverly360CunningPike: Yes..but I haven't given up on it yet ;)
16:44.44CunningPikeWaverly360: Good for you!
16:44.49[TK]D-Fenderjbalcomb : SMRT
16:44.51jbalcombCunningPike ah, that is totally cool. too bad they dont work. :/
16:44.54Waverly360CunningPike: I want to get the phones sending stuff to asterisk, so that I can possibly write a patch for the next version of asterisk.
16:45.15CunningPikeWaverly360: That would be great - you'd make a lot of people very happy
16:45.19*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
16:45.35CunningPikejbalcomb: You get basic status - Online, Offline and Busy
16:45.35jbalcomb[TK]D-Fender yeah, so of course i now have four managers, a supervisor, and some helper bee chewing on my shoes about getting it fixed Right Now(tm)
16:45.53Waverly360CunningPike: It won't happen unless I can figure out how the polycom phones work.  Polycom tech support was less than helpful.
16:46.01terrapenjbalcomb, where do you work
16:46.04CunningPikeWaverly360: You surprise me :|
16:46.08Strom_Cit's Cresl1n!
16:46.18Strom_Cwill you sign my ISDN book?
16:46.32[TK]D-Fenderjbalcomb : thats a step up from my typical Yesterday (tm)
16:46.32jbalcombterrapen: Any Company, Inc. ? ;)  IMT in Brunswick Ohio
16:46.40Waverly360CunningPike: ?
16:46.42terrapenah
16:46.43*** part/#asterisk matheusbh (n=pankz@200150009221.corp.wayinternet.com.br)
16:47.10CunningPikeWaverly360: I've never found their Tier 1 folks particularly useful
16:47.10terrapenmy DTMF recognition problem is STILL happening
16:47.17terrapenupgraded the sangoma drivers and everything
16:47.22*** join/#asterisk Venust1 (n=Emiliano@69-12-128-128.dsl.static.sonic.net)
16:47.32terrapenlast thing to try is to upgrade polycom firmware
16:47.47Waverly360CunningPike: That was my first time talking to them about anything.  They were pretty much just BSing me..I'd like to talk to someone higher up, but I'm not exactly sure how to go about it.
16:47.55Cresl1nStrom_C: that sounds a little kinky
16:48.04Strom_CCresl1n: you have no idea
16:48.19Strom_C;)
16:48.20CunningPikeWaverly360: How many sets do you plan to have?
16:48.22*** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com)
16:48.42Waverly360CunningPike: Sets?  How many phones?
16:48.43terrapenthe worst thing about this problem is that I have no idea what to blame it on...Polycom?  Sangoma?  Asterisk itself?  Configuration error?
16:48.58mutanyone used a GR-303?
16:49.02terrapenand even worse, the problem occurs about 75% of the time
16:49.08Venust1hi
16:49.12terrapeni hate sometimes-it-works-sometimes-it-doesn't problems
16:49.19terrapeninvariably, the hardest to debug
16:49.35CunningPikeWaverly360: Yes - sorry - POTS speak still creeps through sometimes :)
16:49.37*** join/#asterisk marv[work] (n=timr@64.89.118.139)
16:50.01Waverly360CunningPike: Well..any ideas where I can get more info on the inner workings of these polycoms?
16:50.04Venust1sometimes when I'm workin out in my shop I can't hear my SIP phone ringing
16:50.25Strom_CVenust1: get a louder ringer
16:50.26terrapenwaverly, what are you trying to do
16:50.38[TK]D-Fenderterrapen : I specialize on both nd never had problems with either.
16:50.43CunningPikeWaverly360: We've found a chap called Stephen Sprunk quite helpful - he's one of the head tech honchos
16:50.47Venust1Strom_C, since it's not an AC POTS line how do I do that?
16:50.58terrapend-fender, have you ever seen call parking break?
16:51.07Strom_CVenust1: get a separate ringer box on an ATA and then ring both at the same time
16:51.19terrapendfender, as in, the agent dials the code to park the call but instead of parking the call, the caller hears the DTMF?
16:51.28Waverly360terrapen: I want to MyStat functionality on my 501 and 601s to actually do something.  The options are there, but when I sniff the network traffic coming off of the phone, changing my status to busy, or away doesn't actually send anything to the pbx.
16:51.35Venust1Strom_C, that's what I was thinking, just wanted to see if there was a phone already made to do what I want
16:51.45Venust1google doesn't show anything
16:51.45terrapendfender, but sometimes they don't hear the DTMF and the call is parked properly.  i can't explain it
16:51.53Strom_CVenust1: what kind of sip phone?
16:52.05Waverly360terrapen: once I get the phones sending info across the wire, I can start working on asterisk support for those options.
16:52.07Venust1Strom_C, Grandstream GXP2000
16:52.20Waverly360CunningPike: Any chance you could give me a way to get in touch with him?
16:52.30Strom_CVenust1: the cisco phones have loud ringers :)
16:52.31terrapenwaverly, mystat?  like, DND?  I haven't noticed this on my polycoms
16:52.54Venust1Strom_C, I want a ringer light too
16:52.57Waverly360terrapen: You have to enable presence in asterisk for the options to show up.  Once you do that, and reboot the phone, the options magically appear ;)
16:52.59terrapenwaverly, how are you sniffing?
16:53.17Venust1Strom_C, I'm thinking ATA with an add on Flash/Ringer box
16:53.23Waverly360terrapen: I connected my laptop and polycom phone to a hub, and ran ethereal on my laptop
16:53.27Strom_CVenust1: cisco has it
16:53.41Strom_CVenust1: or you can do the ringer box
16:53.47MstlyHrmlsWaverly360: what version of polycom software are you using?
16:53.48Venust1Strom_C, aren't Cisco phones extremely expensive?
16:53.54Strom_C$250?
16:54.03terrapena real hub, eh?  not a switch
16:55.01MstlyHrmlsWaverly360: do you have a capture of the signalling from boot-up and registration?
16:56.21jbalcombdoes the temporary greeting get played first and foremost regardless of the status of you phone?
16:56.40Waverly360MstlyHrmls: I'm using SIP version 1.6.6.0036
16:57.01[TK]D-Fenderterrapen : Never tried call parking actually.
16:57.05Waverly360MstlyHrmls: I didn't get a capture of the boot-up, but I can re-set everything backup and do that if you think it will help.
16:57.14Venust1Strom_C, what model Cisco phone?
16:57.25*** join/#asterisk kpettit (n=keith@adsl-70-241-120-196.dsl.hstntx.swbell.net)
16:57.28Strom_CVenust1: 7940 was 250 for one unit last I looked
16:57.29MstlyHrmlsWaverly360: I'm just looking at the Admin guide for 1.6
16:57.36kpettitcan you allow people to dial options while on hold?
16:57.36CunningPikejbalcomb: Yes - it overrides everything
16:57.53MstlyHrmlsWaverly360: it reads to me like the phone will only send the presence info if another device SUBSCRIBEs to it...
16:57.54[TK]D-FenderVenust1 : Not extremely, just more that they're worth VS Polycom
16:57.54terrapend-fender, really?  you should try it some time, it's quite cool.  when it works.  :)
16:58.03kpettitI'd like them to be able to "press 1 to leave a message" or other options while there waiting on hold
16:58.17terrapenwhat component of asterisk detects the DTMF during a call?
16:58.35Strom_Cterrapen: the channel driver, IIRC
16:58.48jbalcombCunningPike thanks
16:59.14CunningPikejbalcomb: np
16:59.20terrapenso, since the agents are using Polycom SIP phones, I should do some SIP debugging
16:59.39MstlyHrmlsWaverly360: I've gotta muck around with a Polycom and another system later today, I can see if I can play around with the presence stuff then as well
16:59.39*** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com)
16:59.50kc5cqmhey, what are dta310's going for these days, used?
16:59.55kc5cqmI've got 5 of them I want to sell.
17:00.05Strom_Ckc5cqm: I'll do a dance for one
17:00.14kc5cqmthey're a pain in the arse
17:00.14*** join/#asterisk TeePOG (n=temp@dsl-145-178-200.telkomadsl.co.za)
17:00.27fholmesIN the [Demo] section of the extensions.conf file the Starting script uses exten => s,n,Answer.  What does the n do?  Should there be a number there or not?
17:00.31terrapenwhen the agents dial the park-call code, the caller hears a clicking or popping noise....except occasionally, they hear the agent's DTMF....except for sometimes they hear nothing and the call is parked properly.  It's the strangest thing I've ever encountered with *
17:00.35mutanyone here into telcom ever used a Telcordia GR-303
17:00.48Strom_Cmut: isnt GR-303 a specification
17:00.51*** part/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net)
17:01.11rob0fholmes: not completely sure, I think it means "next", just makes it easier to insert/remove priorities.
17:01.17terrapenfholmes, the wiki would probably help you a lot
17:01.21terrapenesp, the extensions.conf page
17:01.23Strom_Cthe n priority means "next"
17:01.27mutyea
17:01.36rob0fholmes: you can use labels to jump to a specific "n".
17:01.44Strom_Cmut: i forget, what does GR-303 specify?
17:01.54fholmesrob0:  So where to the labels go?
17:02.00rob0terrapen: I guess I missed that on the Wiki.
17:02.03mutgeneric dlc requirements
17:02.35mutand some 'next gen' stuff i guess
17:02.38rob0fholmes: there is an example in the sample extensions.conf
17:02.44CunningPikefholmes: exten => 1234,n(label),DoStuff()
17:03.09*** join/#asterisk carl0s- (n=carl0s@compsup.demon.co.uk)
17:03.12muti gotta find out exactly what switch is there..
17:03.26Strom_Cmut: gimme an NPA-NXX and I can look it up
17:03.36*** join/#asterisk docE (n=docelmo@66.237.242.41.ptr.us.xo.net)
17:03.40mut989-507
17:03.40fholmesCool.  Thanks guys.
17:04.00docEwhadup
17:04.21carl0s-I had a chat with a guy today, an seller on eBay who's selling some voip fxo gateways. He said he's given up because he was "terminating gsm lines" with them and the gsm providers (Vodafone etc.) kept discovering this and blocking his simcards. He said he was using 600minutes/day. What exactly does he mean? Is this something naughty that you aren't supposed to do?
17:04.24*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
17:05.31Strom_Cmut: odd, it doesnt show a switchtype
17:05.36mut:P
17:05.54terrapencarlos, he was probably using one of those cellphone->POTS converters and then plugging those into his FXO gateways
17:05.59muti'll just run down to the CO and look at it next week or something
17:06.13Strom_Cmut: DMS-something-or-other would be a safe bet
17:06.14terrapenusing them to make advertisement calls to other cell phone users
17:06.24carl0s-terrapen: yes, he was. but for what purpose? To route calls from his voip service onto the mobile networks?
17:06.28terrapensince in-network calls are typically free
17:06.45terrapenpossibly...but more likely, he was doing some kind of spamvertisement
17:06.56carl0s-terrapen: I wish they were! (here in the UK). They are cheaper than network-network calls.
17:07.08carl0s-terrapen: hmm yes spamming did cross my mind.
17:07.21terrapenthere is a guy in turkey that uses asterisk to make political advertisement calls over GSM gateways
17:07.28terrapenlike thousands of them each minute
17:07.50carl0s-blimey. he must have a big cellphone bill!
17:08.07terrapencarlos, well, here in the states, same-provider calls are typically free
17:08.35mutits an access navigator m1o
17:08.47carl0s-that's outrageous. I knew you had free local calls on your POTS lines. I didn't know you had free same-network calls.
17:09.49carl0s-same-network calls cost 10p/minute here. That's about 18c/min to you.
17:10.51terrapenyeah, we can talk for free to anyone who uses the same provider.  and if we call between 2100-0600, we can call anyone in the US for free
17:10.52carl0s-or $11/hour
17:11.02carl0s-life isn't fair.
17:11.10terrapencome on over :)
17:11.14carl0s-:D
17:11.18carl0s-lol
17:11.40carl0s-I might struggle now. I received a small conviction for buying goods which had been stolen.
17:11.58*** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com)
17:13.15*** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net)
17:13.26terrapen"See these goods--they've never seen daylight, moonlight, Israelite, fanny-by-the-gas-light..."
17:13.46carl0s-:o
17:14.17[TK]D-Fender</eyesclosed>
17:14.55carl0s-So I've finally given up on the Motorola based X100P and have a TDM01B arriving on Monday.
17:15.10Strom_Ccarl0s-: YAY
17:15.15carl0s-:D
17:15.31[TK]D-Fendercarl0s- : I give you till wednesday TOPS....
17:15.54carl0s-[TK]D-Fender: what, before I come on here asking "How do I..?" or until I've got everything working properly?
17:16.19*** join/#asterisk jhiver (n=jhiver@LReunion-151-20-4.w193-253.abo.wanadoo.fr)
17:16.21*** join/#asterisk dacleric (n=dacleric@p54821F8C.dip0.t-ipconnect.de)
17:18.09[TK]D-Fendercarl0s- : No, that will be monday NIGHT.  Wednesday you will be bald, and retuning it SCREAMING :)
17:18.21carl0s-:(
17:18.22carl0s-;)
17:18.34Strom_C[TK]D-Fender: why dont you bag on digium some more, please
17:18.45Strom_C:)
17:19.01carl0s-The place I ordered it from actually took my call through one, and it sounded great. Actually, it was an openvox board they were using but much the same I should think
17:19.26[TK]D-FenderStrom_C : I never placed BLAME, just his mental state at the end!
17:19.31Strom_C:)
17:20.02[TK]D-FenderStrom_C : Don't put words in my mouth... its full enough with foot as it is ;)
17:20.08Strom_Chaha
17:20.25carl0s-I've already been up 'til 4am the last two nights, and in front of the computer all through the daytime. I look pale and feel a bit zombified.
17:20.39Strom_Cok, ill double check to see if chan_mouth is busy before executing a dial next time
17:21.03jhiveranybody knows when these new transcoding boards will be out, and at what price?
17:21.14mutwell lets hope * can do gr-303 stuff well
17:21.16Strom_Cjhiver: they will be out soon
17:21.26*** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net)
17:21.56jhiverand also, how many concurrent channels they will support? I supposed they will be sized to a 4 E1 / T1 board
17:22.11Strom_Cjhiver: 120 channels
17:22.22jhiverthat's it then :)
17:22.35jhiverand what about the price? any info on this?
17:22.35Strom_Cdisclaimer: specifications may change before final release
17:22.41Strom_Ci haven'
17:22.49Strom_Ci haven't yet heard prices being thrown around
17:23.20Strom_Chowever, they'll include transcoding licenses for 120 channels for g729 and g723 IIRC, so figure that it wont be cheap
17:23.23jhiverwith onboard echo cancel and hardware transcoding, we might finally have some pretty good quality
17:23.49Strom_Cwait, did you just say transcoding and good sound quality in the same sentence?
17:24.12jhiverwell it's better than CPU transcoding and no echo cancel :)
17:24.25Strom_Cbest sound quality == all ulaw, no transcoding :)
17:24.38jhiverI transcode g729 to ulaw on my audiocodes gw and it sounds great
17:25.05jhiverI prefer to do g729 all the way usually
17:25.06Strom_Cg729 still sounds like cellphone calls though
17:25.18jhiverno, cell is clearly worse
17:25.19Strom_Cyou dont get the same clarity that you do with ulaw
17:25.23jhiverof course
17:25.44Strom_Chell, I'd do 48khz 16-bit telephony if I could
17:25.54jhiverbut then you pay the price in terms of bandwith
17:26.00Strom_Cwell sure
17:26.16Strom_Cbut bandwidth is worth paying for if it means better-quality phone calls
17:26.40jhiverwell, it depends what you are doing really
17:27.02jhiverwhen you work in the termination business, it's all g729 and g723.1
17:27.10jhiverso having this board should be very good
17:27.51Strom_Ci would run from a carrier that transcoded my wireline calls to g.729 :)
17:28.19jhiverthen you can run from a _lot_ of telcos
17:28.33jhivereven the 'non voip' ones
17:28.40Strom_Ctrust me, I do :)
17:28.51[TK]D-FenderRun Forrest, RUN!
17:29.03jhiverbecause a lot of them buy from other carriers that interconnect with TDM with them, but then go all VoIP behind the scenes
17:29.38Strom_Cyeah, but you can do voip without touching g.729
17:29.54Strom_Cthe transport and the coding are not intrinsically linked
17:30.02jhiverwell yes and no
17:30.22jhiversometimes you have a route that is only H323 / G723.1 and you need to sell as SIP/g729
17:30.34jhiverand so it's not like you have a choice
17:31.09jhiverto sell a route better, you need to accept as many codecs and protocols as possible
17:31.16Strom_Cwell, yes
17:31.20jhiverand hopefully this new card will let me do just that
17:31.23Strom_C*nod*
17:31.43jhiverBTW, I'm having quite a lot of headaches with H323
17:31.54*** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com)
17:31.56Strom_Cthe H stands for Headache
17:31.59Corydon-wNote that Digium itself isn't licensing the G.723.1 codec.  It's the chip manufacturer that has done that.
17:32.14jhiverit's a complete bitch to compile, and when it works it only sort of half works
17:32.31jhiverI wonder if * business edition has proper support for H323
17:32.38jhiverprobably not
17:32.39*** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com)
17:32.48muti don;t think you can get single 723 licenses
17:33.00mutit's bulk, very large bulk
17:33.14jhiverapparently the way to do H323 is to use Yate but the docs are pretty harsh
17:33.23jhiverand there is no howto either :(
17:36.11StromCommI AM TEH NEXT VONAGE
17:36.27*** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com)
17:40.43drrayI am the next AOL
17:42.17*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
17:42.40NivexI am the next Tastee Freeze! </random>
17:42.52Strom_Cpfft, tastee freeze
17:43.04drraytestes freeze!
17:43.08jhiveractually I have a little trick question for you guys
17:43.09shmaltza clients system that is running externhost in sip.conf went down today because of no internet and therefore no dns. What program can I run on the Asterisk box that will act as a DNS proxy so that it doesn't happen again?
17:43.25jhiverI *think* my ISP is discriminating / limiting VoIP traffic
17:43.48drrayshmaltz - just run a dns server?
17:43.53jhiverI wonder if there are tools I could use to prove this (sending different types of traffic to a given box and measuring the results)
17:44.07jhiverany ideas?
17:44.08drrayjhiver - why not just tunnel the traffic
17:44.09shmaltzdrray, I don't want a full dns server, I just want a silly dns proxy
17:44.17mutmy isp doesn't give a crap what i do cause i pay them enormous amount of monies
17:44.22drraya dns server is a proxy
17:44.25shmaltzjhiver, whos your provider?
17:44.39jhivermut, I _do_ pay them outrageaous amounts of money
17:44.49shmaltzdrray, you are right but I want one that all it can do is proxy
17:44.49jhiverorange business services (i.e. france telecom)
17:45.18jhiverI am supposed to have a 4 Megs guaranteed connection
17:45.28jhiverI use only 1.3 Megs peak
17:45.43muti have 2 ds3's
17:45.50jhiverand I still have packet loss at times of the day when pinging one of their routers
17:46.24jhiverand that kind of pisses me off since the connection costs something like 2k€ / month
17:46.29fholmes~wiki
17:46.43fholmes~wiki extensions.conf
17:46.47drrayfreshmeat returns DNRD a proxy dns server, along with 200 other hits
17:47.20shmaltzjhiver, coplain
17:47.20jhiverso I was wondering what tool you would use to prove that there is an issue...
17:47.24shmaltzcomplain*
17:47.26jhiverI do!
17:47.43jhiverI have loggued all ping times and packet loss and produced pretty graphs
17:47.52jhiverthey still say there's no problem
17:48.05Strom_Cjhiver: complain to the telecom regulation agencies
17:48.08jhiverwhen it's clear that there is something going very wrong
17:48.21*** join/#asterisk iax (i=mbrooks@hijacked.us)
17:48.32jhiverstill, I was wondering if there is another tool than "ping" I could use
17:48.47Strom_Cmtr?
17:48.57jhivermtr? what's that?
17:49.13Strom_Creally really fancy traceroute
17:49.31jhivercool
17:49.52jhivercheers
17:51.07*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
17:55.56*** join/#asterisk praet (n=praet@wsip-68-15-32-50.ri.ri.cox.net)
17:56.54jhiverthanks lad, nice tool
17:57.04jhiverI'll be able to spot where the problem is now
17:58.56*** join/#asterisk Un1x (i=Un1x@CPE0040ca94518b-CM00137116f37a.cpe.net.cable.rogers.com)
17:59.18*** join/#asterisk fndude (i=sobeit@63-191.126-70.tampabay.res.rr.com)
18:00.26fndudeHi , I'm trying to get call forwarding working with asterisk and my grandstream phone. It says in the logs its passing the call to local@default, but then my provider is giving me back a 603 error, any suggestions?
18:00.54*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
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18:02.10*** join/#asterisk oej (n=oej@38.115.133.12)
18:03.11trelane_this is madly offtopic but I figure someone probably knows, does anyone have an idea what Nextel's current FCC licenses are (operating frequencies)?
18:09.05SkarmethWaverly360, hi, are you in toubles with Polycom SIP 1.6.6 ? I'm too
18:09.51SkarmethI just updated some units here to test this new software and I can't make or receive calls with this new sip
18:12.21Strom_Cexit
18:12.22Strom_Cer
18:12.26Strom_Cgah
18:15.38*** join/#asterisk JakBeatZ (n=JakBeatZ@trek.tor1.ebit.ca)
18:18.25*** join/#asterisk xbmodder_newlapp (i=nobody@atarack/staff/xbmodder)
18:18.32xbmodder_newlappHow long until chan_skype?
18:19.31denonnever, hopefully
18:19.39xbmodder_newlappwhy do you say that?
18:19.50*** join/#asterisk evisu (i=hIRC@bzq-88-153-134-199.red.bezeqint.net)
18:20.08denondunno, skype is kinda the OSX of the computer world
18:20.22gandhijeeare the TDM400's 3.3V or 5V?
18:20.31xbmodder_newlappIt is?
18:20.37xbmodder_newlappWell:
18:20.42xbmodder_newlappATA with skype
18:21.01JakBeatZFolks, having an IAX2 registration problem.  I keep getting registration refused.  Configs and debugs at http://pastebin.ca/88258  Anyone have any ideas?
18:21.02xbmodder_newlappintegrate skype into asterisk would mean I can talk to all my non-techy friends
18:21.23denonor you could just dial out pstn at like 1c/min :)
18:22.28*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)
18:23.11xbmodder_newlappJakBeatZ, slight erorr in the configs
18:23.50xbmodder_newlappJakBeatZ, next time post the config files/errors seperately?
18:23.51xbmodder_newlappok?
18:24.22JakBeatZsure, sorry.
18:25.30xbmodder_newlappwell my reccomendation is:
18:25.35xbmodder_newlappset one peer
18:25.39xbmodder_newlappset one user?
18:25.43xbmodder_newlappand vice-versa
18:25.46xbmodder_newlappfriends are complex
18:26.00xbmodder_newlappand I have too bad of a headache to remember how to set that up
18:26.29JakBeatZI see.. ok..  So instead of friend - friend do peer - user..
18:26.43xbmodder_newlappJakBeatZ, er
18:26.47xbmodder_newlappWanna talk in PM?
18:26.52JakBeatZSure, yes.
18:31.11Waverly360Skarmeth: I'm having some issues, but I don't think they're related to your problems...
18:31.15*** join/#asterisk jarrod (i=jarrod@juniperyour.net)
18:31.37jarrodhey is there a way to group individual sip accounts into some type of group in order to specify an call-limit
18:31.55jarrodand asterisk checks the group limit rather than individual
18:31.56*** join/#asterisk pigpen (n=mark@fw.seamans.cc)
18:32.11*** join/#asterisk tzanger (n=tzanger@mixdown.ca)
18:32.44Dr-Linux|workJul 14 11:11:46 NOTICE[18865]: rtp.c:564 ast_rtp_read: Unknown RTP codec 104 received
18:32.51Dr-Linux|workanybody know what's wrong?
18:33.14stoffell_hDr-Linux|work: did you try setting your asterisk to force the sip client to ulaw/alaw ?
18:33.32Waverly360MstlyHrmls: I need to figure out how to make asterisk subscribe to the status options in the phone..any ideas where I should start?
18:33.58Dr-Linux|workstoffell_h: well, my codecs prority is something like:
18:34.01Dr-Linux|workg729
18:34.07Dr-Linux|workg723
18:34.12Dr-Linux|workilbc
18:34.15Dr-Linux|workulaw
18:34.17Dr-Linux|workalaw
18:34.20Dr-Linux|workgsm
18:34.38stoffell_hDr-Linux|work: to see if it changes, try setting only 1 codec (example: the well known u/alaw)... to see if it changes..
18:34.56MstlyHrmlsWaverly360: unfortunately, no. I know Polycoms, but I'm not great with asterisk yet... :-7
18:35.03Dr-Linux|workstoffell_h: i can't that
18:35.34stoffell_hfor testing purposes, anything goes ;)
18:35.53Dr-Linux|workstoffell_h: i'm having this problem with only one node like outside caller >> Mulititech VOIP GW (g729) >> asterisk >> softphone
18:36.14Dr-Linux|workso if i remove g723 nothing will work
18:36.39stoffell_hhm, try "sip debug" and prepare for lots of debug messages then.. then post to mailing list ?
18:37.01Waverly360MstlyHrmls: Hrm...  There has to be a way I can trick my phone into thinking someone subscribed to that feature...
18:37.36*** join/#asterisk rva (n=rafa@200.210.51.130)
18:37.41*** join/#asterisk d-tech (n=dtc@72.245.233.107)
18:37.46MstlyHrmlsWaverly360: if you could find something that would send it the right SUBSCRIBE message, but I'm not sure what event you have to subscribe to
18:37.55rvahas anyone ever managed to make sip can reinvite work with asterisk?
18:39.04*** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca)
18:39.08Waverly360MstlyHrmls: I guess I could go through the Admin manual some more...not sure if I'll find what I'm looking for there.
18:44.07*** part/#asterisk mtaht4 (n=m@71.198.23.124)
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18:44.36Corydon-wrva:  I wasn't aware that it was ever broken
18:44.43*** part/#asterisk rva (n=rafa@200.210.51.130)
18:47.06jarrodhey is there a way to group individual sip accounts into some type of group in order to specify an call-limit so asterisk checks the group and not the individual
18:48.42*** join/#asterisk matheusbh (n=pankz@200150009221.corp.wayinternet.com.br)
18:48.46Mercestesjarrod  We do something like that with a mysql database using an AGI script with .php.
18:49.01matheusbhHi All... Somebody know how to use one Pap2na ATA like a FXO port?
18:49.07*** join/#asterisk veepster_ (n=veepster@67.130.38.2)
18:49.10jarrodim looking at checkgroup() or something and app_groupcount
18:49.12jarrodor something
18:49.45matheusbhit´s possible to do that?
18:50.16Mercestespossible.  That's just the working model I have.
18:51.06matheusbhbut using like a fxo?
18:51.48matheusbhMercestes: Are u using a pap2na like a fxo port?
18:52.16Mercestesmatheusbh:  We use it to convert to analog.  If by FXO port you mean "auto answer" or "not provide dial tone" then no.
18:52.27sandra78http://www.mail-archive.com/asterisk-users@lists.digium.com/msg143374.html this is my issue
18:52.38*** join/#asterisk alexns (n=alex@static-71-240-121-39.pitt.east.verizon.net)
18:53.04alexnsyo
18:53.41*** join/#asterisk Waverly360 (n=mirc@209.12.249.243)
18:54.30*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
18:55.02alexnsso what are you guys using to remotly monitor your clients asterisk systems ?
18:55.13Waverly360SSH :P
18:55.17*** join/#asterisk fulgas (n=fulgas@a81-84-116-58.cpe.netcabo.pt)
18:55.24alexnshehe
18:55.30Tall-guyI use the phone, when it doesn't answer, my asterisk box is down :)
18:55.35*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)
18:55.45alexnsi mean graphically through the managers interface... and im talking about your clients
18:56.24alexnsso the average idiot can know whats up
18:56.35Tall-guyoh, well you're assuming we're average....HUGE mistake :)
18:56.47alexnsabove average in here
18:57.01alexnsbut can't pay an expert to monitor these systems
18:57.26alexnsjust need the average tech to be able to see if providers are down and such
18:57.41alexnsi used to have a  program but cant remember what it was called
18:57.49*** join/#asterisk enjay- (n=enjay@71.216.165.97)
18:58.09jarrodare there any presence utilities that are multiple client aware as in an isp platform?
18:58.27enjay-If I wanted to use a FAX machine on my asterisk server to sound out fax would I use a "HandyTone" type solution for analog->voip conversion?
18:58.56enjay-sound/send
19:01.27Waverly360Ok, I have a request for anyone using a Polycom 501/601 with a NON asterisk PBX...
19:02.46*** join/#asterisk kindor (n=roy@office.open-ict.nl)
19:03.24jarrodhow do i disable the music on hold all together so it doesnt even try to use it
19:03.37Tall-guyum, comment it out of the .conf file?
19:03.49jarrodit still tries to use it
19:03.59enjay-reload your config after commenting out
19:04.22jarrodi did a reload res_musiconhold.so
19:05.31enjay-just do a reload of all config files (i.e. reload) see if that resolves it..
19:06.49Waverly360MstlyHrmls: fyi, I just grabbed a packet capture of my polycom 501 after it's rebooted.
19:07.58*** part/#asterisk fulgas (n=fulgas@a81-84-116-58.cpe.netcabo.pt)
19:09.11*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
19:10.40[TK]D-FenderWaverly360 : What is this question just offhand...
19:10.46*** join/#asterisk alexns (n=alex@static-71-240-121-39.pitt.east.verizon.net)
19:11.52Waverly360[TK]D-Fender: I'm still trying to figure out how to make the statuses work with asterisk :P
19:13.10[TK]D-FenderWaverly360 : Did a sip debug on it?
19:13.57[TK]D-FenderWaverly360 : it may depend on * ASKING for "102 OPTIONS" which it doesn't IIRC.  I believe * just tracks its OWN activity and reports it to phones that poll it.
19:14.07daysmen3which do you prefer using ael or extensions
19:14.17daysmen3and why??
19:14.52Waverly360[TK]D-Fender: I haven't done a sip debug on it...still learning the intricacies of using the CRI.  (If that's even where it's done)
19:15.36Waverly360[TK]D-Fender: hmm...I'd really like to see how other PBXs ask for those options...
19:17.44Waverly360[TK]D-Fender: Crap...how do you turn SIP debugging off once it's on?
19:17.52enjay-sip no debug
19:17.54[TK]D-FenderWaverly360 "sip no debug"
19:18.11Waverly360[TK]D-Fender: Ahh... thanks :)
19:18.19[TK]D-FenderWaverly360 : Use "sip debug peer [peerentry]" to only debug 1 phone to see how it communicates
19:19.23Waverly360[TK]D-Fender: oh good..that makes things much easier.. :)
19:19.30*** join/#asterisk sharp (n=sharp@c-68-45-160-72.hsd1.pa.comcast.net)
19:19.41sharpanybody know any good wireless hardphones?
19:20.02[TK]D-Fendersharp : They all suck ATM
19:20.30sharpthen any good wired hardphones?
19:20.42sharppreferrably cheap?
19:21.13Waverly360[TK]D-Fender: Is it possible to log the stuff from the CRI to a file temporarily?  Or am I stuck copying and pasting?
19:23.04[TK]D-Fendersharp : Polycom > ALL
19:23.05CunningPikealexns: We're using Intermapper with Nagios plugins to monitor our Asterisk servers
19:23.21[TK]D-FenderWaverly360 : Don't know the details, but I think its in logger.conf
19:23.39sharp[TK]D-Fender, thanks for the suggestion
19:23.40Waverly360[TK]D-Fender: K... Appreciated :)
19:28.34Waverly360[TK]D-Fender: I need to get my hands on some polycom source code :P
19:31.17[TK]D-FenderWaverly360 : No, time to start reverse engineering :)
19:31.52Waverly360[TK]D-Fender: heh... I suppose you're right...just start sending random things to the phone and see what happens...lol
19:33.55*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
19:34.02[TK]D-FenderWaverly360 : exactly!
19:34.06*** join/#asterisk MikeJ[Laptop] (n=vircuser@38.115.133.12)
19:34.31[TK]D-FenderWaverly360 : start debug, set a stat, change to another, then back.  Try to create messages in a pattern that will tell you if its a random packet or a reaction.
19:36.45xbmodder_newlappAnyone here used voxee?
19:37.22fholmesSo if I have an extension 1200 and in my dialplan I want someone who presses 1 from the menu and I have exten => 1,1,Dial(1200) is that going to work properly?
19:37.41Qwell[laptop]fholmes: Local/1200 might work better
19:38.00fholmesCool thanks Qwell
19:38.38fholmesShould I put an answer first for extension 1?
19:38.45Qwell[laptop]Don't need o
19:38.46Qwell[laptop]to
19:38.59[TK]D-Fenderfholmes : strike that.  Avoice creating additional Local channels and do something like "Goto(contextwithmyexteninit,theexten,1)"
19:39.10[TK]D-Fenderavoid*
19:39.50fholmesSo I should have a second context for the call queue for my tax sales team.
19:40.22fholmesOr if it is in the same contex then I just do Goto(,1200,1)?
19:40.30fholmescontext..
19:42.13[TK]D-Fenderfholmes : it SHOULDN'T be in the same context.
19:43.54fholmesI don't want to include that in the main context either.  It should only be called with the goto command.
19:45.04*** join/#asterisk SanketMedhi (n=sanket@221.135.150.187)
19:45.21fholmesSo what about the agent extensions?  What context should they go into?  The default, their own context or the Tax-Queue context?
19:48.17SanketMedhihello, I am getting this error when I run asterisk -vvvvvvvvddddddddd .....
19:48.20SanketMedhiJul 15 01:16:55 WARNING[7947]: cdr_addon_mysql.c:295 my_load_module: Unable to load config for mysql CDR's: cdr_mysql.conf
19:48.49drrayrealtime messed up?
19:48.53SanketMedhiI am trying to set up mysql Realtime, but I am not using cdr
19:48.57SanketMedhisort of
19:49.22SanketMedhiany idea?
19:50.44SanketMedhidrray: does a warning make a different? I am anyways not using CDR, and it says Mysql has regd successfully
19:51.35drrayI don't know
19:51.52SanketMedhiok
19:52.37SanketMedhianybody else?
19:54.13JonR80015:49 < SanketMedhi> any idea?
19:54.17JonR800sorry
19:54.19JonR800it means nothing
19:54.31JonR800it means it couldn't load the config so it won't load the module, no big deal.
19:54.37SanketMedhiok cool
20:01.01*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
20:01.01*** mode/#asterisk [+o anthm] by ChanServ
20:02.15SanketMedhihow do I check if Realtime has actually parsed the DB?
20:03.21carl0s-groovy.
20:03.29Alricrealtime load ?
20:03.47SanketMedhiYou must supply a family name, a column to match on, and a value to match to.
20:03.51SanketMedhiI get that
20:03.54tzafrir_laptopcheck the actual config?
20:04.02AlricYeah...
20:04.04SanketMedhiextconfig?
20:04.08AlricSo then you supply that information?
20:04.15SanketMedhiin extconfig?
20:04.17carl0s-We don't get free same-network mobile calls over here (UK), but I just found out Orange will let me rent another handset on my business tarrif, and have free calls between the two mobiles on my contract. So for an extra £17.50/month I could route all my inbound SIP calls to my mobile..
20:04.18SanketMedhiI have
20:04.20Alriclike realtime load sip_users username 100
20:04.32SanketMedhiwhat should that do?
20:04.35AlricThats probably sipusers
20:04.43AlricShow whatever username 100 has
20:04.54Alricin the sip config.  What are you trying to load from realtime?
20:05.02*** join/#asterisk Primer (n=vi@sh.nu)
20:05.05*** join/#asterisk chksolutions (n=damian@201.232.77.205)
20:05.23PrimerSo now that the skype protocol's been cracked, can we expect to see a chan_skype soon? :)
20:05.36SanketMedhiAlric: I have only created one table called sip_buddies and I am loading users from that DB
20:05.47chksolutionshello, I really need help in a topic
20:06.10AlricOkay.
20:06.16AlricSo try to pull some data from it?
20:06.19carl0s-Hey even better.. I could route all my own mobile calls though my free mobile-to-mobile plan, via my Asterisk box and pay landline/SIP rates! That £17.50 could pay for itself!
20:06.25AlricThink of it like a select statement from SQL.
20:06.39SanketMedhiAlric: should I be having a username field in the table for that command to work?  I have a name column and other
20:06.50*** join/#asterisk s0lid (n=s0lid@61.28.161.132)
20:06.52chksolutionswe've installed about 4 tdm400 cards in diferent computers, and all of them have noise, like a tick
20:07.02AlricTry matching by name then.
20:07.04chksolutionshow can I resolve this issue
20:07.16AlricIt just wants a column to match by.  The code doesn't care what column it happens to be.
20:07.45*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)
20:08.13SanketMedhiAlric: I have mapped sip_users and sip_peers to the same table , when I tried that  command with these names it worked! :) Thanks
20:08.25AlricCool
20:08.31AlricWelcome to Realtime.
20:09.04SanketMedhiThanx :)
20:09.36SanketMedhiAlric: is there any other resource than voip-info to help me with Realtime?
20:09.44AlricProbably.
20:09.59AlricGoogle might be of assistance.  I've only ever used voip-info though.
20:10.16SanketMedhiok
20:13.07chksolutionsI need help
20:13.10chksolutionsplease
20:14.00*** join/#asterisk N3WWN (n=N3WWN@ns1.futuretek.cx)
20:14.40N3WWNHi Guys!  Anyone know why asterisk would hang up immediately when a call comes in (or out) of my PRI with a TE110 card?
20:15.39AlricDoes it throw an error, or give any kind of information that would help point in a direction?
20:15.51*** join/#asterisk MikeJ__ (n=vircuser@38.115.133.12)
20:16.01chksolutionsThis issue used to happend to me until my provider change the modem
20:16.04N3WWNNot that I can see....
20:16.19*** join/#asterisk bkw__ (n=bkw_@asterisk/friend-and-developer/bkw)
20:16.20N3WWNI see the Accepting call msg, the Answer("Zap/1-1" etc msg
20:16.23AlricWhat verbosity are you running at?
20:16.26drrayhey bkw
20:16.27*** join/#asterisk hfb (n=hfb@pool-71-106-220-165.lsanca.dsl-w.verizon.net)
20:16.32AlricAnd what logging is enabled?
20:16.34N3WWNverbosity 17 ;)
20:17.04Alric... more paranoid than I am...
20:17.12tzafrir_laptopN3WWN, to what context do the call go?
20:17.21N3WWNI then see "Channel 1/1, span 1 got hangup" and a busy
20:17.27tzafrir_laptopyou can see that in 'zap show channels'
20:17.28N3WWNdefault context
20:17.42*** join/#asterisk bkw__ (n=bkw_@asterisk/friend-and-developer/bkw)
20:17.51N3WWNall default
20:18.21tzafrir_laptopWhat do you have in the context 'default'? 'show dialplan deault' . pastebin it if it is long
20:18.23N3WWNI usually use verb 4, but I"m trying to find out why the calls hangup immediately
20:19.13N3WWNhttp://pastebin.ca/88334
20:21.50N3WWNI'm just using MusicOnHold to test cuz I didn't know if jumping to the Winbeam context was causing the hangup
20:22.21*** join/#asterisk Eggplant (i=No@dsl-216-155-214-162.cascadeaccess.com)
20:25.28*** part/#asterisk chksolutions (n=damian@201.232.77.205)
20:25.35N3WWNany thoughts?
20:26.47*** part/#asterisk Primer (n=vi@sh.nu)
20:28.47N3WWNhttp://pastebin.ca/88343 now includes the output from /var/log/asterisk/debug
20:31.56wrmemN3WWN: What are you expecing it to do?  You don't put the call on hold, so it's not going to play music, it will fall through to the next priority which doesn't exist (hence hangup)
20:32.09N3WWNeven changing the exten so it calls a SIP phone causes the hangup
20:32.19N3WWN<PROTECTED>
20:32.19N3WWN<PROTECTED>
20:32.19N3WWN<PROTECTED>
20:32.20N3WWN<PROTECTED>
20:32.20N3WWN<PROTECTED>
20:32.20N3WWN<PROTECTED>
20:32.22N3WWN<PROTECTED>
20:32.35N3WWNThe SIP phones work fine... they can call each other, call voicemail, etc
20:32.36AlricYou can send an incoming call straight to MusicOnHold()...
20:32.59wrmemTry "Answer(), then SayDigits(12345) and see what happens
20:33.23N3WWNAlric, you're right, I use exten 8601 (legacy from the old PBX) to play music through the speaker
20:33.27N3WWNwrmem, I"ll try
20:34.08N3WWN<PROTECTED>
20:34.08N3WWN<PROTECTED>
20:34.08N3WWN<PROTECTED>
20:34.08N3WWN<PROTECTED>
20:34.08N3WWNJul 14 16:35:09 WARNING[10100]: file.c:587 ast_readaudio_callback: Failed to write frame
20:34.09N3WWN<PROTECTED>
20:34.11N3WWN<PROTECTED>
20:34.13N3WWN<PROTECTED>
20:34.35AlricOh yum.  Good luck with that one :)
20:35.04N3WWNtks ;)
20:35.45wrmemIncoming on a PRI?  Hmm.   Has the PRI been turned up?
20:36.07SanketMedhiN3WWN: http://pastebin.ca
20:36.08N3WWNYes, that's how the calls are being sent to the * server
20:36.23N3WWNSanketMedhi, what do you want me to post there?
20:36.37N3WWNI have some info at http://pastebin.ca/88343
20:36.40SanketMedhiN3WWN: next time, post code there
20:36.52*** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net)
20:37.10wrmemI've had PRI's not provisioned completely that wouldn't complete the call, but would have signalling (D up'ed, all B's down)
20:38.14Corydon-wIt's not possible for the B's to be down... they are clear channel signalled
20:38.41wrmemCorydon-w:  Ok.  I'm not using the term exactly correct...
20:38.59wrmem(marked as Out-of-service)
20:39.29Corydon-wSome telcos implement something known as B-channel maintenance protocol, which is not supported by libpri.  The telco needs to turn that off and restart the PRI before it will work with Asterisk
20:39.40N3WWNHow about this... I know zttool doesn't show the sync source right, but should it show total/conf/act channels right?
20:39.46xbmodder_newlappWhen, in an AGI script I "EXEC DIAL SIP/19252029415@plainvoip-out|120r
20:39.46xbmodder_newlapp" it no longer reads from the AGI script until I hang up, is there a way to still work with the channel?
20:40.29N3WWNhttp://pastebin.ca/88353
20:40.49N3WWNOh, that got ugly :(
20:40.56Corydon-wxbmodder_newlapp: not after it's bridged, no
20:41.03xbmodder_newlappdamn
20:41.42xbmodder_newlappWhat about EAGI?
20:41.56xbmodder_newlappHow do I write directly to the audio stream, with PCM data?
20:42.30syzygybsdxbmodder_newlapp: create a local channel
20:42.46xbmodder_newlappsyzygybsd, eh?
20:42.47N3WWNessentially, all the TxA/B/C/D and RxA/B/C/D lines are filled with dashes (-) and Total/Conf/Act is 24/24/0
20:43.14xbmodder_newlappsyzygybsd, how?
20:43.45JakBeatZFolks, box A is * 1.2.0, box B is * 1.2.9.1.  I have a SIP peer configured between the two boxes.  On box A, I get Jul 14 16:43:08 WARNING[5275]: chan_sip.c:1064 __sip_xmit: sip_xmit of 0xb397b0f8 (len 483) to boxBip:-1 returned 0: Invalid argument  Anyone seen anything like this before?!
20:44.11syzygybsdheh, sorry, little information, but the way I do it is spawn a local extension, (dial(local/something@othercontext)) then bridge them together or put them in the same meetme
20:44.48syzygybsdagi won't hang because the local channel is spawned immediatly
20:44.59xbmodder_newlappso have a dynamically created meetme?
20:45.03xbmodder_newlappand do it that way
20:45.09carl0s-Has anybody seen a miniPCI GPRS/GSM card? And.. next question.. would one work as a zap channel in Asterisk? I'm planning on using a WRAP embedded board, which has dual ethernet and dual miniPCI, but no way to hookup POTS which is what most of the GSM gateways output.
20:45.19syzygybsdsure
20:45.40syzygybsdyou might just be able to bridge the calls without a meetme, but I have always done it with one
20:46.11xbmodder_newlappwhat about EAGI?
20:46.20Corydon-wcarl0s-: theoretically, yes.  However, when was the last time you wrote a kernel driver?
20:46.28syzygybsdwhat about it?
20:46.56carl0s-Corydon-w: never. So it hasn't been done already then? I guess I'm going to be limited to a SIP GSM gateway then. There aren'y many/any of those on eBay..
20:47.14syzygybsdthe dial command doesn't return until it is hungup.  Doesn't matter where it is called from
20:47.43carl0s-am I correct in thinking there is a bluetooth channel driver which will utilise a cellphone as an FXO?
20:48.22syzygybsdcarl0s-: http://www.voip-info.org/wiki-Asterisk+Bluetooth+channels
20:48.27syzygybsdfirst result in google...
20:48.48carl0s-syzygybsd: bugger. no call functionality.
20:49.48syzygybsdwell, this could just be me, but bluetooth doesn't support calls from another device over it
20:50.21carl0s-syzygybsd: I was thinking the bluetooth code would utilise the cellphone in the same way a car handsfree kit does.
20:50.29syzygybsddata service yes.. but when I look at the services provided by my phone connected by bluetooth I dont' see call...
20:50.52*** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com)
20:51.16syzygybsdcarl0s-: there is a huge differnce between sending mic/speaker information and sending the whole call
20:51.55syzygybsdyour car handsfree doesn't have a keypad on it does it?
20:51.56cytrakhi I have Asterisk 1.2.7.1 and I'm now trying to setup the manager.conf file like I did before but for some reason it won't work
20:52.13syzygybsdlol.. k, can you tell us what doesn't work?
20:52.18carl0s-syzygybsd: No it doesn't. I see where you're coming from. It can only tell the cellphone to initialise a call from its existing phonebook.
20:52.25cytrakI try to telnet to iP:5038 I get into it but then it just hangs
20:52.25Nuggettelnet is eeeeeeevil!
20:52.42cytrakI get no prompt for action, login, secret
20:52.43syzygybsdI hate that auto response
20:53.02syzygybsdcytrak: you never get a prompt for those
20:53.04carl0s-I though it might be an auto response.. a bit quick really. telnet.
20:53.21carl0s-bah
20:53.29syzygybsdall you should get is something like "Asterisk call manager version 1.0"
20:53.43syzygybsdI don't know what the exact message is..
20:54.21syzygybsdya, there is a minimum time before he will respond to "telnet" again
20:54.38carl0s-syzygybsd: I figured that :D
20:54.47syzygybsdI want to know what it is
20:54.57syzygybsdmore then 2 minutes...
20:55.05carl0s-three minutes I reckon
20:55.06cytraksyzygybsd: ok I also tried to type them in
20:55.12syzygybsdI would guess 5
20:55.22syzygybsdcytrak what did you type in
20:55.53syzygybsdcuz I am guessing you havn't copied from voip-info and tried that
20:56.47*** join/#asterisk ivanfm (n=ivanfm@201.52.162.52)
20:57.17cytraksyzygybsd: Action:login Username:asterisk Secret:geo
20:57.23cytrakit just hangs
20:57.35tzafrir_laptopcytrak, is a connection established? Is this a linux telnet?
20:57.46*** join/#asterisk test34 (n=test34@unaffiliated/test34)
20:57.49cytrakI do remeber getting the promts on an older version
20:57.53cytrakyes
20:58.02cytrakit connects
20:58.03syzygybsdk, did you get the call manager line, and did you do 2 returns after it?
20:58.14carl0s-syzygybsd: http://cgi.ebay.co.uk/VoIP-GSM-Gateway_W0QQitemZ330005452870QQihZ014QQcategoryZ61840QQrdZ1QQcmdZViewItem
20:58.21cytrakthat's what I forgot
20:58.45cytrakthanks
20:59.21syzygybsdnever tried hitting enter again?  Whenever it looks like a console is hanging I hit enter
20:59.23cytrakthat worked now I got find out why my Asterisk-IM plugin for wildfire doesn't work
21:01.04syzygybsdcarl0s-: that isn't what you think it is
21:01.07russellbfile ... you're literally 2 feet from me
21:01.25filetoo lazy
21:01.29carl0s-syzygybsd: How so?
21:01.55*** join/#asterisk alexns (n=alex@static-71-240-121-39.pitt.east.verizon.net)
21:02.09alexnsanyone have any luck with 1.2.9 and g729 ?
21:02.14syzygybsdI am thinking you want that to dial up to your phone provider (Cingular, T-mobile, whatever)  that will allow an unlocked phone to connect to it.. so your cell phone can call yoru asterisk box...
21:02.49hads|homewow, I went out, got drunk, came home and slept and you guys are still poking :)
21:03.01syzygybsdstill poking?
21:03.07filedifferent location though I suppose
21:03.10syzygybsdmmm.. drunk.. good idea
21:03.49syzygybsdboss bringss the beers in 1 hour
21:03.59carl0s-syzygybsd: I don't understand 100%, but basically I have free calls between my two phones on my Orange contract. So I want to place my calls by calling up my second number (which will be that box - it'll have my SIM in it), and getting through to my Asterisk box where I dial a 9 or whatever for an outside line and place my calls at non-cell-rates. Also I would have my incoming sipgate.co.uk calls coming out to me via that also (follow-me).
21:04.40hads|homeIt's 0904 here, I can't get drunk again till I'm finished being hungover.
21:05.07fileyou better run, you better take cover
21:05.09alexnsYO anyone have any luck with 1.2.9 and g729 ?
21:05.12carl0s-hads|home: I'm just on my first Reassuringly Expensive drink.
21:05.37syzygybsdcarl0s-: doesn't orange have it so you can add landlines or other lines you can call for free too?
21:05.51*** join/#asterisk topping (n=topping@adsl-68-122-71-30.dsl.pltn13.pacbell.net)
21:06.29syzygybsdhads|home: if you never stop being drunk you won't ever get hungover
21:06.55carl0s-syzygybsd: I think they have a deal where you can have them take over your office landlines and have cheap calls between all three - every cell and landline in the office. So anyway, I presume that box does what I was hoping? Basically runs my cellular SIM card like an FXO, except using SIP instead of analogue?
21:06.58syzygybsdalexns: I have used that, but call quality goes to shit after 5-20 calls
21:07.28syzygybsdcarl0s-: no, that box doesnt' do what you want
21:07.44hads|homeTrue, but my liver hurts.
21:08.18carl0s-syzygybsd: grr. What does that box do then that's different? I thought being a 'SIP gateway' it would be able to pass all incoming GSM calls over the Asterisk, and also place outgoing calls originating from Asterisk? What is that box meant for then?
21:08.20syzygybsdthat box can make you your own service provide/phone company.
21:08.39syzygybsdk, let me explain it this way...
21:08.44*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
21:09.02syzygybsdwhat you want: Cingular -> phone/box whatever -> asterisk
21:09.20syzygybsdwhat that box does: phone -> that box -> asterisk
21:09.27syzygybsdno cingular anywhere
21:09.31carl0s-cingular?
21:09.52syzygybsddoesnt' matter, just an example of a phone comany, do3esn't matter who
21:09.54carl0s-oh
21:09.59*** join/#asterisk oej (n=oej@38.115.133.12)
21:10.27syzygybsdhad them for a while, they are gsm too.. mentioned orange.. I was guessing
21:10.38alexnssyzygybsd: hmm i use it quite a bit but now since upgrade to latest version 729 codec doesn't work
21:10.45carl0s-I don't understand. You're saying that box doesn't do Cellphone? What's the point of the Antenna and the fact that it's called a 'gsm' gateway?
21:10.57carl0s-I don't want to use a handset with it. I just want to take the SIM card and place in into that box.
21:11.19*** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com)
21:11.40syzygybsdcarl0s-: think of it this way.. that is a base station not a wifi card
21:11.59syzygybsdhmm.. bad example.. base stations can connect to eachother...
21:12.02kpettitin ParkAndAnnount can you use a goto type statement rather than a dial in the notify section?
21:12.15carl0s-syzygybsd: hmm. Are you saying that in effect, that box can't connect to a GSM provider, but instead it makes you "your own GSM company" type thing?
21:12.22syzygybsdyes
21:12.25carl0s-crikey
21:12.27carl0s-that's useless.
21:12.27kpettitI love the ParkandAnnounce feature but it's killing me that I can only announce in a "DIAL" type fashion.  I'm trying to page polycom phones with the annouce
21:12.31carl0s-Are you sure of that?
21:12.33syzygybsdthat is what I was telling you
21:13.12syzygybsdgo read the description, i looked into one of those to save me minutes while I am in the office, but then I realize why not just use a normal sip phone that doesn't cost $5,000
21:13.35carl0s-dude, that's $5,000 taiwan whatsits. It's about $150
21:13.55syzygybsdoh, didn't look at that
21:13.58carl0s-:D
21:14.12*** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net)
21:14.44*** join/#asterisk zotz (n=zotz@24.244.133.115)
21:14.59syzygybsdcarl0s-: look at the pretty pictures they have on that page
21:15.00FuriousGeorge?
21:15.17*** join/#asterisk aster22 (n=aster@202.177.165.236)
21:15.20carl0s-I'll have to message the seller. I usually ignore taiwanese auctions but this looks genuine from the feedback and the company seems real. The description is typically poor and difficult to understand though.
21:15.25aster22hello guys
21:15.35aster22i need some help
21:15.38aster22:)
21:15.40syzygybsdthe pictures explain what it does well
21:15.54FuriousGeorgei think i found a bug in asterisk whereby if i call my own analog lines via my other analog lines, eventually the line that made the call stays "off the hook" and that fxo becomes permanently "in use"
21:16.02FuriousGeorgeuntil i restart asterisk
21:16.07aster22<PROTECTED>
21:16.12aster22on 100 mbps connection
21:16.22carl0s-syzygybsd: I see what you're saying with the pictures and the phones using it as a base station, but they might just have missed out the GSM-provider cloud from that picture. hmm. A GSM base station of that size would be useless. They need to be everywhere to work and they'd be a hell of a lot more expensive than that.
21:16.38FuriousGeorgeeventually all the fxo are in use and we start getting warnings as to that when we make local calls
21:16.45syzygybsdlol.... no... that isn't what that does
21:16.45FuriousGeorgeand incomming calls stop working
21:16.53syzygybsdbut go ahead and message if you want
21:16.53aster22ntil i restart asterisk
21:16.53aster22<aster22>  is a 3200+ athlon 64 with 1 gb ram enough to handle around 50 calls and 100 extensions??
21:16.53aster22<aster22> on 100 mbps connection
21:17.20aster22??
21:17.25[TK]D-Fenderaster22 : internal LAN w/ SIP on G.711 and 2x PRI?
21:17.28carl0s-syzygybsd: I'll see what they say. The price is even cheaper than the FXO->GSM gateways, which certainly do acheive what I want. But I want to avoid needing analog ports on the Asterisk side.
21:17.30syzygybsdaster22: are you connecting via sip anywhere else?
21:17.40syzygybsdor just internal usage...
21:17.43aster22no hadware all sip softphones
21:17.53aster22both side g711
21:17.55*** join/#asterisk RoyK[uk] (n=roy@83.105.70.179)
21:18.09syzygybsdk... are you connecting to the pstn at all?
21:18.17carl0s-Nikki is out of the big brother house, just in case anybody cares :D
21:18.18aster22like VOIP provider--> my dedicated server-->my office far away
21:18.26aster22yes through a voip provider
21:18.33syzygybsdk, what is the bandwidth between offices?
21:18.34[TK]D-Fenderaster22 : no issue then
21:18.52syzygybsdand to the voip provider?
21:18.52syzygybsdand how many calls to them at the same time
21:18.55aster22dedicated server will have 100 mbps uplink
21:19.07syzygybsdwow.. ya.. no issues with anything then
21:19.19aster22and offices will be kept like each pc will get 128 kbps up/down
21:19.24syzygybsdI was thinking that was just internal network...
21:19.46aster22oki thx i just wanted to make sure b4r i buy server :)
21:19.59aster22besides can i disable call transfer on certain extensions ??
21:20.01*** join/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net)
21:20.16syzygybsdis it bad that i find the ad on this page so funny? http://www.google.com/search?num=50&hs=wZU&hl=en&lr=&safe=off&client=firefox-a&rls=org.mozilla%3Aen-US%3Aofficial&q=dead+babies&btnG=Search
21:20.24sevardCan anyone take a look at this please and tell me what I'm missing? It's a simple while look in a dialplan.  Results after the jump: http://pastebin.ca/88390
21:20.43syzygybsdsevard: can you describe what you want to happen?
21:20.47sevardsyzygybsd: url reconstruction blows, tinyurl.com :)
21:20.57syzygybsdoops.. sorry
21:21.03fileoperator operator!
21:21.04carl0s-syzygybsd: no adverts for me at all.
21:21.06sevardsyzygybsd: yes, i constructed a while loop to play an audio file 3 times
21:21.16syzygybsdcarl0s-: no sponsored links?
21:21.16sevardsyzygybsd: at the pastebin i have thee diaplan and the results.
21:21.21carl0s-syzygybsd: no :(
21:21.29syzygybsdhmm...
21:21.52carl0s-Must be an EU censoring thing
21:22.02syzygybsdhopefully...
21:22.11aster22i am pretty n00b to asterisk and use frepbx for config and somewhat changing files myself ...
21:22.25syzygybsdbasically the ad is
21:22.25aster22i want to know abt then settings given while maing extension
21:22.27aster22like dtmf =
21:22.28aster22nat =
21:22.30aster22and all
21:22.36syzygybsd"Dead Babies at Amazon.com"
21:22.38aster22any good coumentation on that
21:22.45aster22:O whats that
21:22.46aster22lol
21:22.49sevardaster22: http://www.voip-info.org
21:23.30aster22k thx
21:23.45sevardsyzygybsd: can you take a look at the pb?
21:24.55*** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com)
21:25.02syzygybsdsevard: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+While
21:25.09syzygybsd$[]
21:25.29mitchelocasterisk just went fatal on me heh, -- http://pastebin.ca/88393 any ideas anyone? i've never seen that before...
21:26.33carl0s-syzygybsd: The MT-350 (slightly different model to that on eBay) seems to do what I want from the text on the site. What do you think? Look at the second paragraph "When you dial out from your MT-350..." http://www.portech.com.tw/eweb/index1.htm
21:26.53carl0s-syzygybsd: sorry, damn frames. http://www.portech.com.tw/eweb/mt/mt350.htm
21:27.38*** part/#asterisk SanketMedhi (n=sanket@221.135.150.187)
21:27.39sevardsyzygybsd: i'm apparently missing some silly asterisk syntax?
21:27.51*** join/#asterisk De_Mon (n=de_mon@fl-69-69-145-173.dyn.embarqhsd.net)
21:28.03carl0s-syzygybsd: bah. it's not SIP. It's FXO/FXS.
21:28.19fileshows should be more... what's the word?
21:28.39syzygybsdit isn't really silly once you understand it, basically it is saying to evaluate whether what is inside that is true or not
21:30.02sevardsyzygybsd: that makes sense.
21:30.18carl0s-syzygybsd: hmm. the MT-370 block diagram shows a SIM card. I think it might just do what I want. I'm downloading the user-manual now. http://www.portech.com.tw/eweb/MV370/mv370.htm
21:30.37syzygybsdya, sim cards are a good indication..
21:30.37enjay-Can flash operator panel integrate with do not disturb?
21:30.48syzygybsdenjay-: yes
21:30.49carl0s-syzygybsd: yup.
21:31.03enjay-syzygybsd; where can I find data on that?
21:31.07syzygybsdcarl0s-: http://www.voip-info.org/wiki/view/GSM
21:31.17*** part/#asterisk m4rkl4r (n=markp@66.129.95.30)
21:31.44syzygybsdlook under pci adapters
21:32.48carl0s-syzygybsd: I looked at the Junghanns stuff but I want to be able to use this with an embedded WRAP ethernet/mini-pci only solution. The 2N thing will do also but I suspect it's pricey.
21:32.57syzygybsdenjay-: voip-info
21:33.10syzygybsdI don't really know what you want so that is as specific as I can get
21:33.40nortexenjay-, Check asternic.org and in the current release readup on the astdb conifg.
21:33.46syzygybsdenjay-: I wouldn't guess any more then $200 but I could be wrong
21:34.34*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
21:34.34*** mode/#asterisk [+o anthm] by ChanServ
21:34.41carl0s-oh yes. The manual for MT-370 shows : "Network registration: The telecom carrier with which the SIM card is registered". I think we're onto a winner!
21:34.59enjay-thanks.
21:34.59syzygybsdya, but how much does it cost?
21:35.11aster22iax2 better or sip if pingtime is more than 300 ms ?? between server and extension
21:35.31carl0s-syzygybsd: £92 + delivery
21:35.50syzygybsdwow, ping time > 300ms... wouldn't really suggest voip
21:35.54syzygybsdI*
21:36.05nortexaster22, Neither :)
21:36.12aster22lol
21:36.14aster22why??
21:36.43syzygybsdI don't really like the delay in my conversations, hearing what they said .3 or .4 seconds after they said it
21:36.43aster22actually i do call in US from my softphone through internet voip providers
21:36.47nortexthe latency would make it about as good as a cell phone in a bunker.
21:36.52aster22and quality is pretty good there
21:36.52CunningPikeaster22: Bcse i wnt wrk vy wl, if at all
21:36.59nortexlol
21:37.16syzygybsdthank you CunningPike
21:37.27carl0s-I get 37ms to sipgate.co.uk. That's adequate I suppose.
21:38.02syzygybsdi get 1ms to my voip server
21:38.16aster22wow
21:38.50syzygybsdthat is going through 3 switches, before it hits the gateway router
21:38.54*** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)
21:39.07nortexCorrect me if I'm wrong, but you want something sub 100ms
21:39.19CunningPikeWe have about 70ms on average in 'sip show peers'
21:39.21syzygybsdthen it goes to the ISP... then their ISP.. then back in through a couple..
21:39.57syzygybsdI am suprised it is 1 ms personally.. with that many networking interfaces I would think at least 10 ms..
21:40.00FuriousGeorgemy snom 360 users are telling me that if they talk fast the other party tells them they break up.  when they call me i hear it, but im not sure it has to do with the rate of speach
21:40.19FuriousGeorgenow that i think about it, maybe it has to do with the switch built into the phone that i'm using
21:40.38syzygybsdhmm.. all of my sip show peers is unmonitored
21:40.38aster22but i get more than 300 ms to my voip provider
21:40.45aster22and quality is still very bearable
21:40.48aster22n gsm
21:40.50aster22on*
21:41.00carl0s-syzygybsd: mine show unmonitored too. I was just wondering how to change that.
21:41.08syzygybsdFuriousGeorge: what codec are you using?
21:41.17aster22gsm or g711
21:41.24aster22on 256 k conn
21:41.30FuriousGeorgesyzygybsd: ulaw
21:41.33TrixVoxAnyone here using VoicePulse?
21:41.42jbroomei am
21:41.59aster22damn no good voip providers in india :(:(
21:42.32aster22which app do u use normally for post paid billing ??
21:42.37aster22on asterisk servers
21:42.38syzygybsdwell, international calls will always have a bit higher of a delay.. so if you think it is acceptable then that is your choice
21:42.41aster22for small scale
21:42.51TrixVoxTheir "Rates" page now shows 0.019 as their highest rate instead of 0.024... When did that happen!?  Some of my calls are still showing as 0.024, even though most are less than 0.01...
21:43.00syzygybsdI dont' use post paid billing... make my company handle all the cost
21:43.10aster22yes thats  ok
21:43.19aster22suppose if i am reselling from higher providers
21:43.29*** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu)
21:43.32aster22ofcourse i cna get billing done via sql records and excel
21:43.35jbroomeTrixVox: they had us change our outgoing server the first week of june, when we did that we dropped down to the cheaper outging rate
21:43.37carl0s-I wonder what "Payment terms:EXW" means.
21:43.39aster22but anything lil automated
21:43.42syzygybsdTrixVox: is that for "new subscribers only"
21:43.52*** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net)
21:44.23De_MonI've got a main-menu that plays some options. The caller can choose one of those options, which goes to submenus, or dial an employees extension directly...
21:45.19TrixVoxEven the wiki still shows "0.5¢ to 2.4¢/min outgoing US long distance"... I think they just dropped their rates from 2.4 to 1.9c
21:45.26De_Monthe timeout extension is used to repeat the menu options, and I need a way to disconnect people that don't dial any extensions
21:45.31*** mode/#asterisk [+o Corydon76-home] by Corydon-w
21:45.57TrixVoxjbroome: Are most of your calls less than 1¢ with VoicePulse?
21:46.37De_MonIf I set the abs timeout, I'd have to turn it off if they called any of the local extensions, and turning off in that many places is not a plesent thought
21:47.03*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
21:48.27jbroomeTrixVox: i believe so, i don't handle the billing, just the * upkeep
21:48.35syzygybsdDe_Mon: that is what timeout is for... you can also have a counter set.. so if counter > 5 hangup
21:49.23TrixVoxAhh, I see... I don't know why people say they're "expensive"
21:49.59jbroomeme either.  call quality is great, haven't had any kind of problems either
21:50.02De_Monsyzygybsd I'd *like* to use the timeout, but turning it off if they leave that menu leaves a *lot* of placs to turn it off
21:51.36TrixVoxYeah, I didn't realize the audio quality difference... I think Asterisk users have gotten used to 'cell phone' quality due to their own bad hardware and/or these smaller providers with echo/dtmf issues.
21:52.11tzafrir_laptopOT: anybody knows if there is any provider except google of voip->pstn service that supports Jingle clients?
21:54.15syzygybsdtzafrir_laptop: how is that OT?
21:54.31tzafrir_laptopIt's not about Asterisk
21:54.41*** join/#asterisk feld_ (n=feld@12.148.212.157)
21:55.21syzygybsd3/4 of the stuff in here is about general VOIP
21:57.28feld_I have 5 incoming analog phone lines
21:58.01feld_I want it to ring multiple phones. By using the DIAL command, if the first person picks up a line and another call comes in it gets sent straight to voicemail.
21:58.18feld_Is the Queue with members the correct solution?
21:58.50syzygybsdya, that sounds like it would work for what you want
21:59.49feld_okay, so am I right in saying that I have to configure these sip phones as agents, set it to "ringall", and give all these members the same priority?
22:01.30*** join/#asterisk TheBlack (n=sirius@p54BB63BC.dip.t-dialin.net)
22:01.38TheBlackhi^^
22:01.57syzygybsdbeers are out, i am afk
22:02.17TheBlacki'm searching a howto to install asterisk with visdn as trunk ...
22:02.21TheBlackcan anyone help me ?^^
22:04.35*** mode/#asterisk [+o file] by ChanServ
22:04.54Qwell[laptop]still a nub
22:06.52RoyK[uk]fucking israelis.....
22:08.07De_Mon^_o
22:08.20De_Monare they interupting your VOIP service?
22:08.25jbroomethey not doing voip right?
22:09.04De_Monshow function GOTOIF doesn't give me anything on the CLI, isn't it a function?
22:09.45Qwell[laptop]De_Mon: I don't think so
22:09.58*** part/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell)
22:09.59hads|homeDe_Mon: show application
22:10.00*** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell)
22:10.05*** mode/#asterisk [+o Qwell[laptop]] by ChanServ
22:10.09Qwell[laptop]:D
22:10.29De_Monah, thats better
22:14.11*** join/#asterisk zotz (n=zotz@24.244.133.115)
22:16.50*** join/#asterisk topping (n=topping@adsl-68-122-71-30.dsl.pltn13.pacbell.net)
22:24.08docelmoHAY!   Anyone in here ever load balanced asterisk w/o SER?  Like by using a F5 or Cisco L4 Switch?
22:24.25benjkanybody here using AstLinux?
22:24.38docelmohehe..  What do you wanna know?
22:25.06benjkI want to know how to modify configuration files on the disk (not on the ram disk)
22:25.09carl0s-that's it. Daddy's bought a Portech MV-370. YAY.
22:25.15docelmoI personally know the developer..  He puked all over my car when he drank WAY TOO much
22:25.21benjkbecause everytime I reboot all the changes are gone
22:25.28Qwell[laptop]docelmo: That boy likes to drink...
22:25.33Qwell[laptop]heavily...
22:25.45fileQwell[laptop]: more then you?
22:25.49Qwell[laptop]file: much more
22:25.51docelmoyes..  I know..  I have a drunken pic of him to prove it
22:25.57filescary
22:26.00Qwell[laptop]file: very
22:26.03benjkWell, that's another issue, the AstLinux bugtracker doesn't seem to work, created an account but didn't get any email with the password
22:26.27benjkand there appears to be no documentation whatsoever
22:26.46docelmokick kris in the balls
22:27.32benjkI already had to set up a fake DNS so the system would boot because it went into an endless loop trying to find its NTP server
22:27.35docelmowhen he was at my house a couple months ago he passed out just before the sprinklers went off he got soaked..
22:27.51DaminI've got Pictures of Kristian swigging vodka in our Hotel room at the Luxor.. :)
22:27.58DaminThat was a WILD night.. :)
22:27.58docelmohaha
22:28.09docelmoI sent you the pic of him passed out at my house didnt I?
22:28.14RoyK[uk]~lart benjk for fun
22:28.14DaminYeah.. that's great..
22:29.09benjkRoyK, so you know how to make AstLinux work then?
22:29.42*** join/#asterisk gcarrillog (n=gcarrill@dsl-201-128-97-89.prod-infinitum.com.mx)
22:29.45gcarrilloghi
22:30.02gcarrillogsorry my english isnt good
22:30.04RoyK[uk]benjk: i just use asterisk. sorry.
22:30.17gcarrillogi have some questions about asterisk
22:30.31aster22what?
22:30.34benjkyeah, I would like to do that, but I can't install any system on that damn CF card
22:30.57benjkgot a 2GB card, cant get any Linux installed
22:31.07carl0s-benjk: AstLinux doesn't work right so far then? :) You know I'll be pestering you for the solutions to these problems you're encountering once my WRAP board arrives.
22:31.17docelmoHay G whats your experience with load balancing a SIP device with a F5 or L4 switch?
22:31.30gcarrillogi was had my server asterisk behind of a router
22:31.41gcarrillogwith DMZ enabled
22:31.57benjkno probs with the WRAP board because you don't have to do anything other than copying the WRAP image onto the CF
22:32.02gcarrillogbut i need know which ports i need for SIP authentication
22:32.11gcarrillogplease help me
22:32.21benjkbut if you want to use  PC and isntall from ISO CD, that's where things get tricky
22:32.28carl0s-benjk: ah I see.
22:32.32enjay-5060
22:32.49gcarrillogenjay- 5060 dont works
22:32.57enjay-5060 is the port
22:33.04gcarrillogmy softclient says request timeouth
22:33.11enjay-got a firewall?
22:33.18*** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx
22:33.27enjay-got NAT behind the firewall? do you have the port forwarded to the server etc etc etc?
22:33.41hads|homeooo
22:34.00gcarrillogenjay- yea but the port 5060 its routing to asterisk server
22:34.27gcarrillogenjay- only works if i enabled DMZ to server
22:35.23*** part/#asterisk mog (n=mogorman@gateway.digium.com)
22:35.26*** join/#asterisk mog (n=mogorman@gateway.digium.com)
22:35.26*** mode/#asterisk [+o mog] by ChanServ
22:35.51enjay-uh
22:35.55*** join/#asterisk fndude (i=sobeit@63-191.126-70.tampabay.res.rr.com)
22:36.00enjay-whats your border device?
22:36.34gcarrillogborder?
22:36.36gcarrillogrouter?
22:36.37enjay-Like a PIX firewall, a Netgear Wireless AP/router?
22:36.40Luke-Jrgcarrillog: it's UDP, not TCP
22:36.50gcarrillogLuke-Jr :O ok
22:36.52enjay-yup
22:36.52fndudemy call forwarding from my grandstream phone keeps getting '603' from my provider, can somebody tell me what would cause this?
22:36.54gcarrillogi will try
22:36.58enjay-sorry thought that was obvious :D
22:37.00gcarrillogmy router its a 2wire
22:37.08Luke-Jrgcarrillog: unfortunately, Asterisk doesn't support TCP SIP
22:37.29rob02wire ... ugh.
22:37.47Luke-Jrenjay-: SIP can be either
22:38.01enjay-Luke-Jr, as you said not with asterisk..
22:38.20Luke-Jrenjay-: he might not have known ;)
22:38.22enjay-and Im assuming he's working with an asterisk system..
22:38.26enjay-yea absolutely, my bad..
22:38.40Luke-Jrand it's not really a bad assumption to think Asterisk can handle TCP SIP
22:38.45Luke-Jrsince TCP makes sense for SIP
22:38.50gcarrillog:D
22:38.54gcarrillogthat works
22:38.54gcarrillog:D
22:39.01gcarrillogenabling UDP
22:39.02gcarrillog:D
22:39.04gcarrillogthanks
22:39.53*** join/#asterisk MACscr (n=MACscr@66.73.154.70)
22:40.03enjay-kinda makes sense.. I dont see it as entirely necessary to have an established connection.. I would think a datastream would be more efficient, more problematic yet more efficient..
22:40.17E-bolaDo any1 know of a click-to-call browser solution for windows? For none firefox users
22:40.30Luke-Jrenjay-: UDP is always more efficient =p
22:40.37MACscrAny phone techs in here that work a lot with avaya equipment?
22:40.39Luke-Jrenjay-: it's not reliable tho
22:40.49enjay-when dealing with text :D
22:41.08hads|homeE-bola: http://www.snapanumber.com/ possibly.
22:41.17E-bolathanks hads
22:41.21enjay-Luke-Jr familiar with FOP?
22:41.34Luke-JrFOP? never heard of it
22:41.40enjay-flash operator panel?
22:41.46Luke-Jr...
22:41.47mitchelocwhoo hoo!
22:41.57Luke-Jrif you mean Macromedia
22:41.59Luke-JrFlash can die and burn
22:42.09mitchelocE-bola, we are working on IE right now, maybe less then 2 weeks away
22:42.10Luke-Jralong with companies who abuse it
22:42.16E-bolaohhh sweet
22:42.21enjay-haha
22:42.36Luke-Jrwhere 'abuse' is defined as anothing other than animation
22:42.52enjay-Im talking more specifically teh "Operators Panel, Receptionist Panel, etc" called Flash Operators Panel "http://www.asternic.org"
22:42.57E-bolawhats difference between pro and free version micheloc?
22:43.06enjay-but I'll take that as a no..
22:43.15E-bolaahh nm
22:43.16E-bolafound it
22:43.45mitchelocE-bola, one helps feed my 6 children, the other doesnt ;)
22:44.08E-bolahehe k
22:44.16drray6 children
22:44.24feld_ok guys I have an agents/queue setup and working, only I have one gripe.... It will ring in from the analog phone lines and then get picked up by extension 5000 which rings, waits 2 seconds, and puts them in the Queue.
22:44.48feld_The problem is that the instant they're in the call group they're on music on hold
22:45.09feld_I would prefer there to ONLY be music on hold if all agents are busy; I'd rather have them hear it ringing.
22:45.37E-bolamitcheloc: i dont see any license info
22:45.43E-bolais the free version freeware?
22:45.53feld_*instant they're in the call queue, not call group
22:46.32*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
22:47.08groogsGoto(context,s/5551234,1) fails -- is there a way to make that work? I have   exten=>s/5551234,1,Answer.....  for doing CID-matching
22:49.02*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
22:49.21fndudewhen asterisk trys to forward calls from my grandstream, my service provider (telasip) denies the request, any ideas on what could be causing this?
22:50.38mitchelocE-Bola: it runs as pro thens switches to basic, if you have more questions you can pm me ;)
22:51.29E-bolahehe alright
22:51.39E-bolai guess i gotta fix my asterisk installation first though :)
22:57.22*** join/#asterisk adorah (n=Administ@87.69.72.228)
23:01.24*** join/#asterisk Chotaire (i=chotaire@chotaire.net)
23:10.03xbmodder_newlappHas anyone here messed with Asterisk EAGI?
23:15.51*** join/#asterisk adker (n=adker@70-100-230-148.br1.glv.ny.frontiernet.net)
23:16.47*** join/#asterisk n9urk (n=leonard@user-0ce2dhc.cable.mindspring.com)
23:17.24n9urkhi all, Is it possible to have music on hold playing while an extension is dialed?
23:17.52n9urkinstead of playing the ringing
23:18.53xbmodder_newlappprobably
23:19.30n9urkwhich function do you think might do it?  (or which func. would you look at first)
23:20.00CunningPiken9urk: Look at the options for the Dial() command - I think one of them does that
23:20.20n9urkThanks
23:20.36xbmodder_newlappm: Provide Music on Hold to the calling party until the called channel answers. This is mutually exclusive with option 'r', obviously. Use m(class) to specify a class for the music on hold.
23:20.59xbmodder_newlappthat'll be 19.95, do you want frys with that?
23:21.18n9urkYes, SuperSize me as well please
23:21.30CunningPikeHey - where's my cut?
23:21.39xbmodder_newlappWould you like to with credit card, check, or paypal?
23:21.42xbmodder_newlappCunningPike, ...
23:21.42n9urkI would like a milkshake
23:21.45xbmodder_newlappyour a cook
23:21.51n9urkIOU in blood
23:22.06rob0mmmmmm milkshake
23:22.11CunningPike:)
23:23.13n9urkmmmmmmmmmmmmm milkshake with sprinkles!
23:23.20xbmodder_newlappn9urk, which vein would you prefer?
23:23.27xbmodder_newlappjugular is my favorite
23:23.30xbmodder_newlappquick & dirt
23:23.31xbmodder_newlappy
23:23.37n9urkleft ventrical
23:23.42xbmodder_newlapphm
23:23.45xbmodder_newlappthats a little hard
23:24.52rob0Vein? No way, arterial blood is better tasting and better for you. (And much faster for the victim.)
23:25.15rob0Especially in a milkshake. :)
23:25.16xbmodder_newlappstill, its a pain
23:27.09xbmodder_newlappI mean peircing it
23:31.25rob0Not far past that jugular, to get to the carotid artery.
23:31.45rob0oh ... you mean the wall resistance
23:32.13rob0Anything worth doing is worth extra effort. :)
23:32.49xbmodder_newlappwould you prefer mouth to organ, or using some sort of siphon.
23:33.13n9urkI always like it mouth to organ
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23:56.23PakiPenguinwhat is the meaning of this Forcing Marker bit, because SSRC has changed
23:59.59aster22anyone know a good wholesale voip provider for corporate purposes

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