| 00:00.05 | JT | oh, so it's not releasable? |
| 00:00.19 | unpaidbill | i did it that way so i didnt have to replicate my number formatting rules |
| 00:00.22 | unpaidbill | oh it's releasable |
| 00:00.33 | unpaidbill | if you want i'll send you the perl cgi and the extension |
| 00:00.37 | unpaidbill | i mean, the code is fucking horrible |
| 00:00.40 | JT | is the code tainted by this cisco stuff though? |
| 00:00.45 | unpaidbill | but as long as you dont mind |
| 00:00.51 | unpaidbill | no, there is no cisco code |
| 00:00.52 | JT | i've never done any firefox extensions |
| 00:01.17 | unpaidbill | i ripped out all the cisco shit |
| 00:01.26 | unpaidbill | i mainly took his number regex |
| 00:01.33 | unpaidbill | and his structure |
| 00:01.57 | JT | yeah i wouldn't mind a copy if you're offering :) |
| 00:02.27 | unpaidbill | i'd release it officially but im kind of embarrased by the code |
| 00:02.38 | unpaidbill | i'm not a programmer.. im a tinkerer and duct taper |
| 00:02.40 | unpaidbill | hehe |
| 00:03.12 | unpaidbill | i shoudl have it finished in few hours unless someone bothers me |
| 00:03.25 | unpaidbill | (coworker) |
| 00:03.30 | JT | i'm not a hardcore developer either |
| 00:04.25 | unpaidbill | anyway, the perl script requires Net::Telnet and CGI |
| 00:04.36 | unpaidbill | you may need to isntall them |
| 00:04.52 | JT | ok |
| 00:06.20 | doolph | JT any other trick ? |
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| 00:50.01 | Nivex | has anyone in here managed to get a call out with this: http://gizmo5.com/pc/calling-list/ ? |
| 00:50.14 | Nivex | I get connected, hear 0-1 seconds of audio, then the call drops |
| 01:04.23 | drmessano | Nivex, 0-1 seconds of real audio or white noise? |
| 01:10.03 | Nivex | drmessano: I've encountered both. |
| 01:10.19 | Nivex | I've heard a "Hello", and also just noise :( |
| 01:12.31 | Nivex | calls to other sipphone/gizmo users go through fine |
| 01:12.38 | Nivex | it's only when I try to transit to PSTN |
| 01:12.59 | Nivex | It reeks of being on their end, but I thought I'd check here for any troubleshooting suggestions before I wrote it off |
| 01:14.28 | *** join/#asterisk macros73_ (n=cs@c-71-236-81-28.hsd1.pa.comcast.net) |
| 01:15.11 | mchou | drmessano: I don't know if you remember from a couple weeks ago, I was the guy with 2 PAP2 behind a openwrt NAT |
| 01:15.53 | mchou | drmessano: in any case, it did turn out to be a NAT issue |
| 01:16.03 | mchou | drmessano: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=464357 |
| 01:16.13 | *** join/#asterisk smace (n=smace@189.84.240.251) |
| 01:16.21 | mchou | drmessano: just wanted to let you know you were correct |
| 01:16.39 | smace | hi [TK]D-Fender ... have you been away? |
| 01:24.01 | andrewn | someone is using my DID as their caller ID and harassing people with solicitation phone calls. people don't understand that you can fake caller ID so they call me and complain |
| 01:24.29 | andrewn | anyone know of any sources of (604) DIDs? |
| 01:26.30 | *** join/#asterisk PepOSX (n=angeldav@190.72.154.134) |
| 01:31.14 | hardwire | dangit.. so queues kill channel variables set along the dialplan eh. |
| 01:31.24 | hardwire | I was trying to leave "breadcrumbs" as a userfield for cdr |
| 01:31.47 | hardwire | and I'm calling "Local" instead of "Sip".. but I think I know how to fix this.. hmm |
| 01:31.50 | *** join/#asterisk implicit (n=bayan@ip68-4-97-211.oc.oc.cox.net) |
| 01:32.35 | hardwire | <PROTECTED> |
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| 01:54.10 | hsv-al | . |
| 01:54.24 | lanning | .. |
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| 01:54.52 | hsv-al | http://wallpaperstock.com/audi-r8-mtm_wallpapers_8550_1600x1200.jpg |
| 01:55.01 | lanning | -.-. --.- |
| 01:56.02 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:58.04 | hsv-al | http://192.87.65.61:8200 - in a .pls streamer |
| 01:59.16 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 02:00.20 | hsv-al | WTF?? |
| 02:00.21 | hsv-al | http://www.blizzard.com/diablo3/ |
| 02:00.23 | hsv-al | WTF??? |
| 02:00.23 | hsv-al | lol |
| 02:04.19 | *** join/#asterisk Juggie (i=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 02:04.45 | WilliamK | interesting, that's actually the 1st time I've seen the cinematic trailer |
| 02:04.53 | hsv-al | dude |
| 02:04.58 | hsv-al | that game looks more addicting then D2 was |
| 02:05.02 | hsv-al | thats the last video game i played 8 year ago Diablo2 |
| 02:06.26 | hsv-al | http://www.blizzard.com/diablo3/media/index.xml#gameplaytrailer |
| 02:11.33 | Juggie | so how the hell do the drop down boxes work |
| 02:11.36 | Juggie | i cant get them to scroll |
| 02:12.18 | jaytee | drop down boxes? scrolling? |
| 02:13.47 | hsv-al | skull scroller |
| 02:13.53 | JT | surely addicting is not a word |
| 02:14.48 | jaytee | Addict Ad*dict", v. t. [imp. & p. p. Addicted; p. pr. & vb. |
| 02:14.48 | jaytee | <PROTECTED> |
| 02:14.48 | jaytee | <PROTECTED> |
| 02:15.13 | JT | it's usually addictive |
| 02:15.26 | JT | i think they added addicting just because people started abusing english so much |
| 02:15.35 | jaytee | probably |
| 02:16.36 | jaytee | someday roflcopter will be in the Oxford Concise English Dictionary. |
| 02:18.15 | JT | i bet |
| 02:22.08 | *** join/#asterisk colin2007 (n=info@82-171-111-153.ip.telfort.nl) |
| 02:24.41 | jaytee | that'll probably be in the revised edition published the week before Armageddon. |
| 02:25.24 | colin2007 | hi |
| 02:27.23 | colin2007 | i was wondering if i could use my existing sip line (provided by my dsl provider) with asterisk |
| 02:27.26 | Robba | hi guys |
| 02:27.42 | frogonwheels | hey - I'v ebeen told that there's some problems with SIP: I quote - "everything since 1.4.18.1 is broken" |
| 02:28.00 | frogonwheels | does anybody know what exactly is wrong with SIP on these builds? |
| 02:28.15 | Strom_M | colin2007: (a) there's no such thing as a "sip line" and (b) you need to get account credentials from your ITSP |
| 02:28.17 | Robba | any ideas as to when using ztmonitor if the calling party hangs up first tx goes full? |
| 02:28.34 | Strom_M | Robba: huh? |
| 02:28.37 | frogonwheels | colin2007: yes you can. |
| 02:28.39 | Strom_M | question sense makes doesn't |
| 02:28.55 | frogonwheels | Strom_M: made sense to me |
| 02:28.59 | Robba | have you used ztmonitor? |
| 02:29.24 | frogonwheels | oh sorry second question. |
| 02:29.33 | frogonwheels | I was still reading the first question :) |
| 02:29.39 | frogonwheels | sorry Strom_M - you're right |
| 02:29.50 | colin2007 | and can i use a standard pci telephone modem card to connect my existing phone? |
| 02:29.50 | Robba | ./ztmonitor 1 -v |
| 02:30.06 | Strom_M | colin2007: no |
| 02:30.26 | frogonwheels | colin2007: I've used a PAP2T to connect to my existing phone - |
| 02:30.38 | colin2007 | PAP2T? |
| 02:30.51 | frogonwheels | colin2007: linksys |
| 02:31.22 | frogonwheels | has 2 "sip line" ports |
| 02:31.28 | jaytee | a PAP2T is an ATA from Linksys. it has FXS ports to connect to standard telephones and an ethernet port to connect via SIP to * |
| 02:31.43 | colin2007 | i already have a Zyxel dsl modem+router that has 2 sip ports |
| 02:31.50 | jaytee | about 47 bucks from Telephonydepot.com |
| 02:32.04 | jaytee | no such thing as a SIP port |
| 02:32.21 | frogonwheels | oh picky picky |
| 02:32.31 | colin2007 | you know what i mean don't you? |
| 02:32.42 | frogonwheels | colin2007: yep. |
| 02:32.42 | jaytee | being precise is nice, being sloppy is for slobs |
| 02:32.55 | colin2007 | could i use those? |
| 02:33.15 | frogonwheels | don't know - depends how much leniancy you have with the configuration of the SIP provider |
| 02:33.18 | jaytee | if they're FXS ports you can plug any analog phone into them |
| 02:33.33 | *** part/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 02:34.06 | frogonwheels | colin2007: assuming jaytee's guess is right ('cause he actually knows what those telphone port thingumijigs are called) |
| 02:34.39 | frogonwheels | colin2007: it's then a matter of how easy the SIP bit is to configure - and whether it will let you connect to the sip server inside the network rather than outside the network. |
| 02:34.40 | colin2007 | i have a analog phone plugged in to it as we speak which works fine |
| 02:34.56 | frogonwheels | colin2007: SIAS |
| 02:35.01 | frogonwheels | (suck it and see) |
| 02:35.11 | colin2007 | hehe |
| 02:35.13 | jaytee | colin2007, what model #Zyxel is it? |
| 02:35.18 | colin2007 | P-2602HW-D1A |
| 02:37.17 | colin2007 | all SIP settings are customizable including SIP Server Address, Port, protocol, etc... |
| 02:37.57 | colin2007 | so is that all the hardware i need then? |
| 02:38.14 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 02:40.21 | jaytee | colin2007, you could probably get that to work with *. Just set the SIP server address to your asterisk box. port would normally be 5060 and a SIP number to register to * but routing calls to your ITSP via * through the Zyxel could be a challenge. |
| 02:41.36 | colin2007 | why is that? |
| 02:45.03 | jaytee | because it's a DSL router with SIP to FXS built in designed to be an endpoint for ITSP providers like Comcast, etc. not for internal networks. It may have the ability to register to Asterisk with one sip address and to your ITSP with another. |
| 02:45.33 | jaytee | but I'm not seeing a very in depth description of it on their website. |
| 02:47.06 | *** join/#asterisk colin_ (n=info@82-171-111-153.ip.telfort.nl) |
| 02:47.22 | colin_ | back irc client freaked out |
| 02:48.40 | colin2007 | it has different fields for: SIP Server Address and REGISTER Server Address |
| 02:49.10 | colin_ | brb |
| 02:49.18 | colin_ | going to reboot my irc client |
| 02:50.21 | hsv-al | i still cant believe their making diablo3 |
| 02:50.25 | hsv-al | late 2009 it comes out heh |
| 02:50.53 | *** join/#asterisk colin2007 (n=info@82-171-111-153.ip.telfort.nl) |
| 02:51.00 | colin2007 | back |
| 02:51.29 | jaytee | hsv-al, so you really like Diablo? I've got a great game for ya then! It's called "trixbox" |
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| 02:53.26 | colin2007 | jaytee: you got that last message? |
| 02:53.55 | jaytee | colin2007> why is that? <- that one? |
| 02:54.17 | colin2007 | no: "it has different fields for: SIP Server Address and REGISTER Server Address" |
| 02:54.27 | colin2007 | my irc client freaked out |
| 02:54.33 | jaytee | no, I didn't get that one |
| 02:54.54 | jaytee | ok, that's confusing |
| 02:55.44 | [TK]D-Fender | colin2007: Depending on how your router works it could be that SIP only works on the WAN port, and not the LAN port. |
| 02:56.27 | jaytee | oh, thank god! finally someone with a brain expressed it in exactly the right words that I couldn't quite figure out how to. |
| 02:56.28 | [TK]D-Fender | colin2007: And that it might block SIP from coming in and reaching * by being blocked at the border since that device probably assumes it should be the only rightful target of SIP calls. |
| 02:56.54 | jaytee | [TK]D-Fender, it kinda looks that way from the specs. |
| 02:56.57 | colin2007 | well i guess there's only 1 way to find out ;) |
| 02:57.07 | jaytee | http://us.zyxel.com/web/product_family_detail.php?PC1indexflag=20040520161246&CategoryGroupNo=PDCA2007019 |
| 02:57.14 | [TK]D-Fender | colin2007: Sometimes those boxes are locked down so hard you're fubar'd. Anyways, if you have a prayer of using it with *, I highly recommend you get another router, and use this one behind it |
| 02:57.34 | jaytee | it sure has alot of bells and whistles.....no frikken kazoo though |
| 02:57.54 | [TK]D-Fender | jaytee: just like an SPA-2102 more or less. |
| 02:57.56 | colin2007 | mine has wifi also |
| 02:58.42 | [TK]D-Fender | colin2007: \o/ |
| 02:59.05 | jaytee | [TK]D-Fender, yeah but since my only experience with outbound calling is over PRI and not through an ITSP using SIP I couldn't wrap my head around how he'd set that up. |
| 02:59.06 | [TK]D-Fender | colin2007: Anyways, proxy / registration / whateverelseserver should all point to * |
| 02:59.29 | colin2007 | and then * would point to my ITSP ? |
| 02:59.52 | *** join/#asterisk adr3nalin3 (n=plasma@64.196.12.68) |
| 03:03.02 | [TK]D-Fender | colin2007: thats the plan. If I we you I would attempt to disable the SIP part of your router first and see if you can get * working by itself before messing up your router too much |
| 03:03.28 | colin2007 | i have my whole config backed up |
| 03:03.36 | [TK]D-Fender | colin2007: Go for it then. |
| 03:03.46 | colin2007 | have to install * first ;) |
| 03:03.53 | jaytee | experimenting is half the fun of this stuff |
| 03:04.23 | colin2007 | true |
| 03:05.08 | colin2007 | wow installing asterisk is even easier then i thought: apt-get install asterisk ;) |
| 03:05.13 | jaytee | until you go to install CentOS on a test box with an Intel DG31PR mobo and find out the Intel mobo has a Realtek 8110 gigabit nic that needs a compiled driver. |
| 03:05.57 | jaytee | I always thought it was pretty easy typing ./configure, makemenuselect and then make and make install but that's just how I roll |
| 03:09.04 | colin2007 | is there a GUI available? or is that too much to ask? |
| 03:10.04 | jaytee | the Enchirto from Taco Bell is kinda gooey. I highly recommend it. |
| 03:11.35 | colin2007 | lol |
| 03:12.57 | jaytee | but no, * doesn't come with a gui. there are "spinoffs" that provide a gui through remote web connections but they kinda lock down the dialplan logic and back you into a corner. Not at all flexible. |
| 03:13.31 | colin2007 | ok i'll stick with the cli then |
| 03:13.36 | jaytee | like AsteriskNOW or trixbox and freepbx |
| 03:14.47 | jaytee | there is a GUI for * itself but I've never used it but I think it behaves the same way the GUI in trixbox does with the same limitations. [TK]D-Fender could explain it better. |
| 03:15.22 | jaytee | colin2007, do you have a copy of the book? |
| 03:15.25 | jaytee | ~book |
| 03:15.26 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 03:15.28 | colin2007 | nope |
| 03:15.35 | jaytee | free pdf download |
| 03:15.51 | colin2007 | thnx |
| 03:16.55 | jaytee | the printed version is much better but it costs money. I felt it was worth the price though. hate flipping back and forth through a PDF. |
| 03:17.30 | colin2007 | it's only 604 pages, i could print that ;) |
| 03:18.06 | jaytee | that might violate copyright but sure, go ahead and kill a few trees. makes me happy cuz I hate them and it really pisses of the druids. |
| 03:18.16 | colin2007 | lol |
| 03:18.59 | drmessano | TREES WRECKED MY ROCKETCYCLE |
| 03:19.09 | colin2007 | lol |
| 03:19.32 | jaytee | drmessano!!!!! how are ya? |
| 03:19.32 | jbot | jaytee: peachy |
| 03:19.46 | jaytee | wtf? |
| 03:20.03 | jaytee | how the hell did that happen? |
| 03:20.24 | [TK]D-Fender | jaytee: No violation of copyright. |
| 03:21.05 | jaytee | [TK]D-Fender, well I was kinda hoping it would boost sales so Leif could afford a tune-up on his Chevy Metro |
| 03:21.38 | colin2007 | [TK]D-Fender: is the 'offical' gui limited? |
| 03:23.47 | [TK]D-Fender | colin2007: they're ALL limited. |
| 03:24.59 | colin2007 | cause i'm kinda lost here |
| 03:25.02 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 03:25.13 | colin2007 | all i've got is a directory full of .conf files |
| 03:25.49 | jaytee | which is all you really need. |
| 03:25.59 | colin2007 | where do i start? |
| 03:26.07 | jaytee | with the book |
| 03:26.16 | jaytee | and especially Chapter 5 |
| 03:26.34 | jaytee | but the whole thing is a great read. Part of it even covers a GUI. |
| 03:26.56 | jaytee | but Chapter 5 and 6 are vital to understanding *. |
| 03:27.11 | colin2007 | k thnx |
| 03:28.11 | jaytee | plus the .conf files are heavily commented with examples. |
| 03:29.09 | jaytee | and you'd probably only need to deal with extensions.conf (heart of your dialplan) and sip.conf |
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| 03:31.16 | colin2007 | k |
| 03:34.18 | jaytee | nite all |
| 03:34.22 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
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| 03:37.25 | colin2007 | i'm of to bed also |
| 03:37.28 | colin2007 | gnite all |
| 03:37.31 | colin2007 | out |
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| 04:13.52 | x86 | colin2007: out? of the closet? |
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| 04:16.36 | [TK]D-Fender | *b00m* |
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| 04:18.40 | x86 | [TK]D-Fender: heya man |
| 04:18.44 | x86 | [TK]D-Fender: what's new? |
| 04:19.28 | *** join/#asterisk SanityIO__ (n=SanityIO@77.242.106.77) |
| 04:20.29 | [TK]D-Fender | x86: not much. Cleaning up house (so to speak). Selling off od things I don't need any more. Trying to get back to the essentials, and get my ass outside and enjoying summer |
| 04:21.11 | x86 | hehe |
| 04:21.38 | x86 | had the day off, so the wife and I took the dog to the park in the next city over |
| 04:22.02 | x86 | i was all about the trails, but she pooped out rather quickly |
| 04:22.23 | x86 | (as did the dog) |
| 04:23.14 | *** join/#asterisk doolph (n=doolph@190.141.69.38) |
| 04:23.38 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.139) |
| 04:23.41 | doolph | anyone know what's the trick to don't let analog fxo lines get stuck |
| 04:23.54 | x86 | "stuck"? |
| 04:24.04 | doolph | http://pastebin.ca/1060902 |
| 04:24.15 | *** join/#asterisk eharris (n=eharris@75-43-20-21.lightspeed.austtx.sbcglobal.net) [NETSPLIT VICTIM] |
| 04:24.24 | doolph | yes :( they are like on hold and don't disconnect |
| 04:24.50 | x86 | probably want disconnect supervision |
| 04:25.02 | x86 | voip-info.org has more information on that |
| 04:25.14 | x86 | search for "disconnect supervision" |
| 04:26.28 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 04:27.05 | doolph | there's always the same thing |
| 04:27.52 | x86 | ? |
| 04:28.56 | doolph | what you mean with disconnect supervision |
| 04:28.57 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) [NETSPLIT VICTIM] |
| 04:28.57 | *** join/#asterisk hsv-al (n=sdfsdf@user-24-214-126-81.knology.net) [NETSPLIT VICTIM] |
| 04:28.57 | *** join/#asterisk redax (i=redax@r6.hu) [NETSPLIT VICTIM] |
| 04:28.57 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) [NETSPLIT VICTIM] |
| 04:28.57 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) [NETSPLIT VICTIM] |
| 04:28.57 | *** join/#asterisk Madkiss (i=madkiss@freenode/staff-emeritus/madkiss) [NETSPLIT VICTIM] |
| 04:28.57 | *** join/#asterisk ltd (n=z@pat.transact.net.au) [NETSPLIT VICTIM] |
| 04:28.57 | *** join/#asterisk lotho (n=lotho@static.69.46.46.78.clients.your-server.de) [NETSPLIT VICTIM] |
| 04:29.20 | doolph | theres any command to test ? |
| 04:29.20 | doolph | or i have to ask the telco |
| 04:29.20 | doolph | I asked them already and they are totally lost too |
| 04:29.29 | doolph | so they are useless |
| 04:29.49 | [TK]D-Fender | ~cds |
| 04:29.50 | jbot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 04:30.07 | [TK]D-Fender | doolph: Introduce it like that and they should be able to help you |
| 04:30.32 | doolph | but what can I do |
| 04:30.48 | x86 | [TK]D-Fender: my local LEC had no idea either :( |
| 04:31.06 | x86 | nor did my vendor that supports the ancient Toshiba PBX |
| 04:31.22 | [TK]D-Fender | doolph: maybe its not sinking in... CALL YOUR TELCO, and use the description I just gave you. |
| 04:31.50 | x86 | [TK]D-Fender: be nice now ;) |
| 04:32.33 | doolph | is it somehing to do with indication.conf? |
| 04:32.38 | [TK]D-Fender | doolph: No. |
| 04:32.47 | [TK]D-Fender | doolph: this is ELECTRICAL signalling, not tone. |
| 04:32.47 | doolph | then |
| 04:33.43 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 04:33.57 | doolph | I think the problem is my fxs transfers to others |
| 04:34.26 | doolph | because it never happen if I use SIP as phone |
| 04:35.20 | doolph | can you check http://pastebin.ca/1060902 |
| 04:35.35 | [TK]D-Fender | doolph: only difference would be is if you're saying * can't tell when you've hung up an FXS channel (assuming Zaptel FXS here) |
| 04:35.37 | *** join/#asterisk SanityIO (n=SanityIO@77.242.106.77) |
| 04:35.53 | doolph | it is something weird there |
| 04:36.17 | doolph | let me try to explain that maybe I am wrong |
| 04:36.38 | doolph | I am on FXS, I make a call... |
| 04:36.41 | [TK]D-Fender | doolph: is that 2 FXS channels talking to 2 FXO channels? |
| 04:36.48 | doolph | all fxo |
| 04:36.51 | doolph | let me explain |
| 04:37.02 | doolph | I think its happening this: |
| 04:37.42 | doolph | I am on fxs, i make a call, then i finish, i want to make another one, but i am just too stupid and click on flash trying to close the line |
| 04:37.48 | [TK]D-Fender | doolph: SIP works because someone hangs up the phone, and * KNOWS that the SIP device has hung up. Your FXO/FXO bridged channels don't stand a change because you have no CDS |
| 04:38.00 | doolph | then i dial another one, * things i am putting the first line on hold |
| 04:38.33 | doolph | after I close the 2nd line, it try to bridge them 2 lines |
| 04:38.49 | [TK]D-Fender | doolph: How the hell can you be on FXS when you told me thats all FXO? |
| 04:39.35 | doolph | ok here I go again, I am on FXS, making 1st call (fxo), I try to close (not enough time) then it things that i am putting the line on hold |
| 04:39.52 | doolph | then I make another call (2nd fxo port), after I close the system try to bridge them |
| 04:40.30 | doolph | I can fix the problem if I dont allow threewaycalling |
| 04:40.34 | [TK]D-Fender | doolph: sounds like you aren't staying hung up long enough and * thinks of it like a "flash" and starts a 3-way call / transfer. |
| 04:40.42 | doolph | yes |
| 04:40.43 | [TK]D-Fender | doolph: stay hung up longer |
| 04:40.59 | drmessano | Viagra? |
| 04:41.02 | drmessano | Oh, hung up |
| 04:41.09 | doolph | there should be another fix |
| 04:41.27 | *** join/#asterisk hsv-al (n=sdfsdf@user-24-214-126-81.knology.net) |
| 04:41.47 | *** join/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 04:42.45 | [TK]D-Fender | doolph: that is the timing you have to respect for "flash". |
| 04:43.16 | doolph | I am not the "user" I can understand |
| 04:43.19 | doolph | but... |
| 04:43.26 | *** join/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 04:43.40 | doolph | can I make something to tell them dial * if they want to make another call |
| 04:43.54 | doolph | instead of of hold 1 sec for hungup |
| 04:44.07 | *** part/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 04:44.52 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.139) |
| 04:47.23 | doolph | ah |
| 04:48.37 | [TK]D-Fender | doolph: Yes... and then when they press * in an IVR they will get cut off... |
| 04:50.53 | doolph | how can I do that? |
| 04:50.58 | doolph | or is it by default |
| 04:51.34 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584664.dsl.bell.ca) |
| 04:51.38 | hsv-al | wirts leg + town of tomb book + horadric cube |
| 04:55.14 | [TK]D-Fender | hsv-al: MOOOOOO!!!!!!!! |
| 04:55.23 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 04:57.55 | doolph | tk I cannot tranfer call if i have threewaycalling off? |
| 04:59.52 | [TK]D-Fender | doolph: "core show application dial" |
| 05:00.59 | *** join/#asterisk xand (n=xand@82-71-12-170.dsl.in-addr.zen.co.uk) |
| 05:04.08 | *** join/#asterisk xenonex (n=xenonex@89.218.235.89) |
| 05:19.47 | TrentCreek | help help |
| 05:20.14 | TrentCreek | anyone got an idea after I created a new user, why when I enter the password in it reports "Permission Denied" |
| 05:21.09 | drmessano | a user in what? |
| 05:21.41 | TrentCreek | oops |
| 05:21.43 | TrentCreek | linux |
| 05:22.11 | x86 | login to the console? |
| 05:22.16 | x86 | or ssh? |
| 05:22.25 | TrentCreek | that is what I am trying to do via SSH |
| 05:22.31 | TrentCreek | and I tried FTP |
| 05:22.42 | TrentCreek | FTP indicated "incorrect login" |
| 05:22.50 | x86 | don't have permission to write to the destination directory |
| 05:24.58 | TrentCreek | drxwx------ |
| 05:25.07 | x86 | who owns it? |
| 05:25.13 | TrentCreek | that user |
| 05:25.20 | x86 | ls -la /path |
| 05:25.29 | x86 | copy full output here |
| 05:25.49 | TrentCreek | total 12 |
| 05:25.49 | TrentCreek | drwxr-xr-x 3 root root 4096 Jul 1 15:27 . |
| 05:25.49 | TrentCreek | drwxr-xr-x 22 root root 4096 Jul 1 15:32 .. |
| 05:25.49 | TrentCreek | drwx------ 2 trent trent 4096 Jul 1 15:27 trent |
| 05:26.07 | drmessano | Dude, it's the grey server |
| 05:26.12 | drmessano | "they're all grey" |
| 05:26.20 | drmessano | "The one with the grey bottom" |
| 05:27.06 | x86 | TrentCreek: and you're usin user "trent"? |
| 05:27.18 | x86 | via SFTP over SSH? |
| 05:27.42 | TrentCreek | not using ..trying to login via SSH |
| 05:27.46 | drmessano | http://www.youtube.com/watch?v=BcQ7RkyBoBc |
| 05:27.46 | TrentCreek | to that account |
| 05:28.02 | x86 | you can't login via ssh with that user? |
| 05:28.18 | x86 | check allowed users in /etc/ssh/sshd_config |
| 05:28.30 | x86 | I'd set it to allow all |
| 05:28.34 | TrentCreek | no |
| 05:30.03 | TrentCreek | looking now |
| 05:31.48 | TrentCreek | i dont see anything that would prevent a user from logging in |
| 05:32.13 | TrentCreek | and FTP is not allowing connect either |
| 05:32.43 | x86 | user has a password set? |
| 05:32.57 | x86 | what shell does the user have |
| 05:33.05 | TrentCreek | yes... i set it 3 times to be sure |
| 05:33.29 | TrentCreek | i did not set the shell |
| 05:33.35 | TrentCreek | let me try that now |
| 05:33.52 | x86 | lol yeah you need a valid shell |
| 05:34.03 | x86 | chsh will allow you to change a user's shell |
| 05:34.18 | TrentCreek | well...some do it by default.. |
| 05:35.52 | TrentCreek | okay..set, but server is running super slow for some reason |
| 05:36.45 | TrentCreek | same result |
| 05:37.04 | TrentCreek | "Permission denied, please try again." |
| 05:37.05 | *** join/#asterisk [hC] (n=hardcore@190.10.9.126) |
| 05:38.37 | x86 | what do the logs say? |
| 05:38.48 | x86 | tail -f /var/log/syslog |
| 05:38.54 | x86 | do that as you try to ssh in |
| 05:42.18 | TrentCreek | oh |
| 05:42.57 | TrentCreek | no syslog file |
| 05:43.30 | x86 | <PROTECTED> |
| 05:43.38 | x86 | what do you have in /var/log? |
| 05:43.47 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:44.30 | TrentCreek | http://pastebin.com/d160c539b |
| 05:46.14 | TrentCreek | i got it...secure |
| 05:46.49 | TrentCreek | User trent not allowed because shell bash does not exist |
| 05:47.46 | TrentCreek | now how can that be? |
| 05:47.46 | *** join/#asterisk ltd-- (n=z@patwk.transact.net.au) |
| 05:52.06 | TrentCreek | now it's saying FAILED PASSWORS |
| 05:55.31 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 05:59.07 | TrentCreek | okay...finally go it |
| 06:07.50 | TrentCreek | i dont think that call is real |
| 06:11.08 | *** join/#asterisk rsc (n=robert@fedora/rsc) |
| 06:11.14 | rsc | MOin folks. |
| 06:12.08 | rsc | I've asterisk 1.4 and since a while (I don't know how long now) the problem, that the DTMF tones are somehow broken. The 1 has a completely different DTMF tone as 2-9. |
| 06:12.38 | rsc | Unfortunately, no end recognizes my 1 as DTMF tone in voice menus any longer. I did no asterisk update etc. |
| 06:12.52 | rsc | Urgs? |
| 06:12.58 | rsc | CentOS 5.2 Zaptel troubles:type ~centos52bug? |
| 06:13.03 | rsc | ~centos52bug |
| 06:13.04 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 06:14.21 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:15.04 | TrentCreek | how much crap does that guy have on his workstation? |
| 06:15.29 | *** part/#asterisk pikachu2000 (n=pikachu2@196-209-198-201-rrdg-esr-2.dynamic.isadsl.co.za) |
| 06:15.36 | drmessano | who/ |
| 06:15.42 | TrentCreek | your web guy |
| 06:15.46 | drmessano | ha |
| 06:16.06 | drmessano | Yeah, you just shut down the exchange server |
| 06:16.16 | TrentCreek | he must be running Vmware on XP, or on Linux |
| 06:16.40 | drmessano | DO NOT REBOOT THE WEBSERVER! |
| 06:16.46 | TrentCreek | gotta be 4 gigs of ram |
| 06:16.50 | TrentCreek | no |
| 06:17.00 | TrentCreek | it was DO NOT REBOOT THE WEBSERVER!!!!!!!!!!!!!!!!!!!!! |
| 06:17.16 | drmessano | He totally |
| 06:17.19 | drmessano | Totally |
| 06:17.21 | drmessano | totally |
| 06:17.32 | drmessano | pwned the dude that called him and bitched about rebooting |
| 06:17.36 | drmessano | That was some shit I would do |
| 06:17.45 | drmessano | Fucking open his exchange mailbox, delete the sent item |
| 06:17.50 | drmessano | Fucking GENIUS |
| 06:18.25 | drmessano | FUK U! |
| 06:18.51 | drmessano | We all watched that at work today |
| 06:20.19 | TrentCreek | yeah |
| 06:20.20 | TrentCreek | lol |
| 06:20.43 | drmessano | Why are you using AOL |
| 06:20.45 | *** join/#asterisk rcy` (n=rcy@shop.freegeekvancouver.org) |
| 06:20.47 | drmessano | To get on the internet |
| 06:21.07 | drmessano | We have highspeed.. like an OC30 we pay $1000 a month for |
| 06:21.11 | drmessano | Don't use fucking AOL |
| 06:21.18 | drmessano | "BUT I HAVE 4000 FREE HOURS" |
| 06:21.23 | drmessano | "Can I roll them over?" |
| 06:22.26 | TrentCreek | I wish OC30 was $1000 a month |
| 06:24.10 | drmessano | I loved how the sales guy was ripping into the tech about how everytime they call for support, they fuck more things up and don;t actually fix the problem |
| 06:24.25 | drmessano | Sounds like someone who had his Weatherbug removed under protest |
| 06:25.13 | TrentCreek | yeah the guy.."web site is down".."okay I will reboot the server"..."oh I can't get to the homepage now" |
| 06:28.10 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:28.10 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 06:29.54 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582384.dsl.bell.ca) |
| 06:31.53 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-36b3fda861485b8b) |
| 06:32.03 | *** join/#asterisk lordmortis (n=lordmort@203-206-116-137.dyn.iinet.net.au) |
| 06:34.46 | *** join/#asterisk L-info (n=L-info@g0962184.demon.co.uk) |
| 06:38.27 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 06:43.22 | dominic1 | hello guys I need something which shows my the state of specific extensions on my windowsscreen |
| 06:45.10 | drmessano | AstAssistant |
| 06:47.17 | dominic1 | is there something more usable for a secretary which only will have four or five people in her list? |
| 06:49.39 | drmessano | Not really |
| 07:06.05 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:08.17 | *** join/#asterisk Mike8861 (n=IE@059148244254.ctinets.com) |
| 07:08.25 | Mike8861 | hello all. |
| 07:11.53 | *** join/#asterisk coreyf52 (i=coreyf52@ftc164.gw.coreyfarrell.com) |
| 07:12.27 | coreyf52 | ~centos52bug |
| 07:12.28 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 07:14.16 | *** part/#asterisk coreyf52 (i=coreyf52@ftc164.gw.coreyfarrell.com) |
| 07:17.02 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:19.46 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 07:21.33 | Mike8861 | hello gr0mit |
| 07:21.52 | gr0mit | morning Mike8861 |
| 07:22.11 | Mike8861 | mornging!!! it has been so late now |
| 07:22.20 | Mike8861 | just have pizza for lunch! |
| 07:22.21 | Mike8861 | yay |
| 07:22.41 | Mike8861 | well, gr0mit, i got question about dialplan, can u help ? |
| 07:23.41 | gr0mit | ok |
| 07:26.08 | Mike8861 | i am trying to create a extening 9000, when user dial ext9000, it will dial out to a PSTN number, and send some DTMF to the external IVR (located on the PSTN side) |
| 07:26.37 | gr0mit | ok |
| 07:26.49 | Mike8861 | i will post the config to pastebin site. whats the web address ? |
| 07:27.04 | gr0mit | what type of channel are you calling out via? analogue, ISDN, VOIP? |
| 07:27.25 | gr0mit | pastebin.ca |
| 07:27.31 | Mike8861 | SIP channel |
| 07:27.34 | gr0mit | ok |
| 07:27.45 | Mike8861 | i am calling out to PSTN via a SIP provider |
| 07:28.11 | gr0mit | understood. paste your config at pastebin.ca, removing all passwords etc please! |
| 07:28.12 | Mike8861 | all channels have been setup without issue, we have test dialing out with number way |
| 07:28.33 | gr0mit | ok |
| 07:28.58 | gr0mit | and also which codec are you using to the PSTN? |
| 07:29.03 | Mike8861 | [tk]D-fender has help me on this lastnight, but he is too pro, i cannot understand what he speak |
| 07:29.19 | Mike8861 | how to verify for codes to PSTN ? |
| 07:30.03 | gr0mit | well, dont worry about the codec for the moment |
| 07:30.09 | gr0mit | paste me your config |
| 07:30.43 | Mike8861 | yup, i guess i got some syntax problem on dialplan |
| 07:30.52 | Mike8861 | here it is: http://pastebin.ca/1061119 |
| 07:31.40 | Mike8861 | the PSTN number will be 1878200 (however dialing out with trunks require a 9 prefix) |
| 07:33.04 | gr0mit | ok well there are a few issues! |
| 07:37.29 | gr0mit | http://pastebin.ca/1061121 |
| 07:37.39 | gr0mit | I think this is more like what you wanr |
| 07:37.41 | gr0mit | want |
| 07:37.57 | gr0mit | but i smell freepbx here somewhere |
| 07:39.02 | creativx | it has that scent.. |
| 07:39.54 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 07:39.56 | Mike8861 | gr0mit: thanks, well, I do this manual without help of any GUI! |
| 07:40.40 | gr0mit | yes, but who knows what goes on in the depths of the freepbx gui. I never got my head around it |
| 07:41.51 | gr0mit | Mike8861, as we say here - we do not support freepbx - it causes waaaay too much pain. |
| 07:41.52 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:42.57 | Mike8861 | gr0mit: i am not that pro, so I do not understand what pain on freepbx and trixbox, but i do understand that rules on this channel do not allow question on those |
| 07:44.11 | gr0mit | the issue is, Mike8861 that a lot of problems people post here would not be problems if they would just use asterisk instead! |
| 07:46.12 | Mike8861 | gr0mit, uhmmm, I can see trixbox and freepbx is a twoside knife, it gives convience, and make things easy to complete, but also create trouble and problem ? |
| 07:47.25 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:49.06 | unpaidbill | damn |
| 07:49.13 | unpaidbill | you lucky dr pepper hoarding dude |
| 07:49.14 | unpaidbill | s |
| 07:49.57 | Mike8861 | hello unpaidbill ! |
| 07:50.05 | unpaidbill | hi |
| 07:51.41 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-4e68eb8f73d2f90b) |
| 07:51.48 | unpaidbill | i picture trixbox and the likes as an oil painting. it's nice, you pay a bunch of money for it and you can tell your friends 'hey look what i bought'. |
| 07:52.11 | unpaidbill | asterisk on the other hand, is a blank canvas that you make into something beautiful. something you can tell your friends 'hey look what i created' |
| 07:52.17 | unpaidbill | personally, i'd rather paint my own picture |
| 07:52.37 | creativx | quite the analogy |
| 07:52.48 | Mike8861 | feels sad |
| 07:52.53 | unpaidbill | what can i say ,i'm 5 beers in and i dig asterisk |
| 07:53.00 | Mike8861 | dump trixbox to the trash area |
| 07:53.25 | creativx | haha |
| 07:53.40 | creativx | the problem arises when you try to make trixbox do what asterisk can do |
| 07:53.49 | creativx | and you headbang into the bounds of its gui/config engine whatever |
| 07:55.13 | unpaidbill | oh baby my gyoza is ready |
| 07:56.46 | rsc | Somebody an idea what's wrong if only the key "one" on the telephone causes a strange sounding DTMF tone? Keys 2-9 are sounding normal and as expected. |
| 07:59.58 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:03.54 | Mike8861 | rsc: can u provide teails about the PHONE ? |
| 08:04.17 | Mike8861 | rsc: what model are u using excatly, how do u connect it with asterisk ? |
| 08:04.27 | Mike8861 | rsc: do u use POE ? |
| 08:05.12 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 08:05.17 | rsc | Mike8861: this is independent of the phone, happens via SIP and ISDN. |
| 08:05.52 | rsc | Mike8861: if I listen to the DTMF tones internally, all sounds as expected, only from outside it's strange for key 1. |
| 08:06.07 | rsc | unfortunately, it already worked and nobody knows, when the behaviour switched. But there was no asterisk upgrade etc. |
| 08:07.41 | mosty | i'm looking at these sangoma BRI cards which appear to use chan_woomera, for which there is not a whole lot of info on. is this a dead software project? |
| 08:08.13 | JT | not completely dead |
| 08:08.33 | mosty | mosty, have you used the sangoma a500d? |
| 08:08.43 | JT | i believe there's a bit of work with chan_woomera development going on, but not in asterisk circles atm |
| 08:08.53 | mosty | JT, rather |
| 08:09.11 | JT | no i haven't |
| 08:09.26 | JT | i'm trying to avoid BRI |
| 08:09.30 | JT | it's too much hassle |
| 08:09.38 | JT | PRI is so cheap in australia now |
| 08:09.43 | Mike8861 | rsc: did DTMF-1 used to be working ? or it never works nomally |
| 08:10.19 | mosty | JT, this box will only have 6 channels, makes PRI a bit much |
| 08:10.36 | JT | i can't remember, are you in nz or au? |
| 08:11.59 | rsc | Mike8861: DTMF-1? |
| 08:12.23 | Mike8861 | rsc: dtmf tone for button 1 |
| 08:12.28 | mosty | JT, au |
| 08:12.50 | JT | is the system in a metro area? |
| 08:14.18 | jblack | LOL |
| 08:14.43 | mosty | JT, yeah, st kilda road in melbourne |
| 08:14.54 | JT | mosty: PRI will be cheaper than 6 BRI channels. |
| 08:14.59 | JT | the calls will be cheaper too |
| 08:15.09 | JT | who cares if you don't use all the channels |
| 08:15.20 | JT | and i don't need to mentione 1000 times less headache with asterisk |
| 08:15.34 | JT | s/mentione/mention/ |
| 08:16.24 | rsc | Mike8861: ah. DTMF-1 worked in the past somewhen, but since unspecified time (sorry) it doesn't work any longer - on all phones. |
| 08:16.46 | *** join/#asterisk defswork (n=andy@mx1.3gcomms.co.uk) |
| 08:16.50 | rsc | Mike8861: colleage said, DTMF-1 shounds non-German, while DTMF-[2-9] sounds correct |
| 08:17.03 | mosty | JT, i tend to agree. i'm trying to decide if i should offer to help this company who i just discovered have already signed a contract for BRI service |
| 08:17.24 | JT | the only bri solution i can recommend for asterisk with any degree of confidence is to buy an external BRI to SIP gateway |
| 08:17.51 | JT | i've heard rumours that bri will be phased out at the end of next year, but that's probably rubbish |
| 08:18.23 | mosty | i hope it is phased out |
| 08:18.38 | kamanashisroy | hi, does anyone use unlimitel iax ? I cannot send outgoing call . I was trying iax2/number:pass@iax3.unlimitel.ca/destination .. .... It is making me crazy ! |
| 08:18.52 | JT | it still has its uses, but yeah, it's getting more and more useless |
| 08:19.14 | JT | with current prices from telstra competitiors |
| 08:21.09 | mosty | JT: what does a 10 channel PRI from optus cost approximately these days? |
| 08:21.21 | JT | last i checked $200/mo |
| 08:21.26 | dominic1 | is there any alternative to hudlite? |
| 08:21.26 | JT | it's $165/mo from primus |
| 08:21.39 | JT | +$33 for a 100 number range |
| 08:21.43 | dominic1 | I need a desktop notification system for my users |
| 08:21.47 | JT | inc gst |
| 08:22.14 | JT | primus call rates are a fair bit better than optus |
| 08:24.29 | JT | mosty: free install on a 24 month contract too |
| 08:25.13 | mosty | JT: their website seems almost devoid of isdn info, i will try calling them tomorrow |
| 08:25.28 | JT | all good prices are hidden |
| 08:25.39 | JT | verizon call rates are even better, and so is their rental |
| 08:25.47 | JT | but their install fees are prohibitive |
| 08:25.50 | *** join/#asterisk Vortice (n=skjhf@81-208-60-197.ip.fastwebnet.it) |
| 08:25.54 | JT | and their sales is singapore based |
| 08:25.57 | Vortice | hi |
| 08:26.11 | JT | powertel only sell through resellers now, and their call rates aren't that grand |
| 08:26.22 | kamanashisroy | JT: yes .. most of them take a huge installation charge .. |
| 08:26.32 | mosty | yeah we have dealings with verizon, but not at the customers end |
| 08:26.41 | kamanashisroy | JT: that is why I am doing some configuration .. |
| 08:27.06 | JT | mosty: a *lot* of voip providers/managed pabx providers do at least inbound termination through primus |
| 08:27.17 | JT | some other providers use verizon |
| 08:27.23 | JT | kamanashisroy: in australia? |
| 08:27.34 | kamanashisroy | JT: now please let me know if I can call like IAX2/number:pass@iax3.unlimitel.ca/destination_number .. :D |
| 08:27.45 | kamanashisroy | JT: nooop .. |
| 08:28.03 | JT | also if you need circuits in global switch, primus have muxes there now |
| 08:28.04 | kamanashisroy | JT: I think it is us provider .. unlimitel.ca |
| 08:28.25 | JT | kamanashisroy: why don't you just call normally and setup an entry in iax.conf? |
| 08:28.49 | JT | and Dial(IAX2/<entry in iax2.conf>/number) |
| 08:28.49 | kamanashisroy | JT: normally means ? |
| 08:28.58 | kamanashisroy | JT: OK |
| 08:29.20 | kamanashisroy | JT: thanks I am trying .. I added a register using username:password@proxy .. OK trying .. |
| 08:29.43 | JT | kamanashisroy: there is no need to highlight me with every line of progress |
| 08:29.57 | kamanashisroy | JT: sure |
| 08:31.59 | *** join/#asterisk buliwyf (n=buliwyf@email.stwhas.de) |
| 08:32.05 | buliwyf | re |
| 08:33.41 | *** join/#asterisk MrNaz (n=naz@ppp59-167-72-221.lns1.mel6.internode.on.net) |
| 08:35.27 | *** part/#asterisk Vortice (n=skjhf@81-208-60-197.ip.fastwebnet.it) |
| 08:39.44 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 08:40.23 | *** join/#asterisk darviria (n=darviria@87-194-177-180.bethere.co.uk) |
| 08:46.17 | buliwyf | which free asterik bases pbx solution woul you recommend? switchbox, trixbox or something different, there so many different |
| 08:49.59 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:53.22 | mosty | go with whatever you can get good, local support for |
| 08:58.36 | buliwyf | whatabout cti e. HUD from trixbox |
| 09:02.16 | *** join/#asterisk cabbiepete (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
| 09:09.40 | buliwyf | if i want to use a sip trunk, do i have to add an sip account for each phone ? i dont, know wether our Voip Provider support real trunk or just a lot of numbers |
| 09:14.04 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:19.25 | JT | there is not really any such thing as a sip trunk |
| 09:19.43 | JT | but you should be able to share one account to an ITSP amongst multiple phones |
| 09:19.48 | JT | unless your provider sucks |
| 09:37.27 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:39.38 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582384.dsl.bell.ca) |
| 09:47.03 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 09:48.49 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 10:04.10 | mandh | when user hangup phone line (zapat channel ) , asterisk some times not notified that call hanguped , is that related to Signal Busy? |
| 10:05.05 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 10:06.45 | *** join/#asterisk Fuzix (n=fuzix@250-118.citynet.ftth.internl.net) |
| 10:08.19 | *** join/#asterisk MagInfo (n=Miranda@pandora.mgn.ru) |
| 10:08.33 | MagInfo | Hi there |
| 10:09.10 | MagInfo | Does anybody know what happened with MOH in Dial()? |
| 10:09.39 | *** join/#asterisk dimas (n=ds@84.53.210.46) |
| 10:10.00 | MagInfo | Before upgrade it worked correctly but now in console I see that MOH are playing but hear only silence in my phone |
| 10:12.48 | *** join/#asterisk yang (i=yang@CAcert/Assurer/yang) |
| 10:16.23 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 10:19.20 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 10:20.16 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 10:24.27 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 10:25.33 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 10:26.04 | shtoom | does asterisk support VAD? |
| 10:27.45 | MagInfo | shtoom, Yes. But you must enable vad on client side also |
| 10:28.41 | shtoom | MagInfo: is there a way to disable VAD in asterisk ? |
| 10:28.53 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:29.30 | MagInfo | shtoom: You can switch off vad on client side |
| 10:30.20 | shtoom | MagInfo:in case of Zap channels ? |
| 10:31.29 | MagInfo | shtoom: I didn't work with Zap channels and don't know how to disable VAD on Zap |
| 10:35.51 | MagInfo | shtoom: I think that you need to change vad setting in Zap device's settings but not in asterisk |
| 10:42.19 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-321c54251bdef947) |
| 10:44.13 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:44.24 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 10:51.33 | *** join/#asterisk masus (i=masus@88.248.14.186) |
| 10:51.39 | masus | hi all |
| 10:54.41 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 10:58.54 | dominic1 | I need a desktopbased BLF - monitoring application |
| 10:59.14 | dominic1 | I found hudlite, but it's only for trixbox |
| 10:59.27 | dominic1 | and astassist is not very good.. |
| 11:04.23 | Fuzix | http://www.hudlite.org/linuxserver.htm |
| 11:05.34 | *** join/#asterisk ccesario (n=ccesario@189.19.9.100) |
| 11:25.44 | *** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132) |
| 11:26.07 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
| 11:26.12 | dominic1 | helolo |
| 11:26.40 | dominic1 | I was not able to run hudlite on my debain based system |
| 11:27.21 | Mike8861 | dominic1: hudlite is not part of asterisk |
| 11:27.36 | Mike8861 | dominic1: please visit #trixbox to get help on hudlite |
| 11:27.57 | dominic1 | I asked for a alternative twenty minutes before |
| 11:29.11 | Mike8861 | dominic1: products like trixbox are not supported on this channel. |
| 11:29.48 | dominic1 | I know!!!!!!!!!!!!!!!!!!!!! |
| 11:30.16 | dominic1 | I asked for a BLF system which I can install on my desktop |
| 11:30.22 | dominic1 | in combination with asterisk |
| 11:30.40 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:31.05 | *** join/#asterisk trafim (n=reallyma@212.200.84.70) |
| 11:31.58 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:32.31 | trafim | hi guys. quick question: what module should i load on asterisk 1.2, so that STRFTIME func becomes available? cause no DATETIME nor TIMESTAMP is working for me (just return nothing). or maybe i'm missing something? |
| 11:33.46 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:38.24 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 11:39.34 | EmleyMoor | I have 3 iaxmodems running, one as a friend and two as peers. For a reason I can't fathom, the peers appear to be attempting to dial without a context - fax-out is speficied in iax.conf though - do they actually need to be friends for it to work? |
| 11:44.36 | EmleyMoor | Ah, it's users they need to be - silly me |
| 11:48.14 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 11:49.39 | EmleyMoor | How can I set it up so I don't keep getting "No registration for peer..."? |
| 11:49.49 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582384.dsl.bell.ca) |
| 11:50.07 | *** join/#asterisk bsaxon (n=bsaxon@49.sub-70-223-36.myvzw.com) |
| 12:02.20 | *** join/#asterisk merlinn (n=merlin@bramble.vostron.net) |
| 12:02.49 | merlinn | hi, does anyone have any recommendations for TDM terminations |
| 12:03.39 | *** join/#asterisk coppice (n=chatzill@179.202.17.210.dyn.pacific.net.hk) |
| 12:04.03 | gr0mit | merlinn, more info pls. |
| 12:04.19 | merlinn | I need to terminate about 10,000 calls simultaneously |
| 12:04.39 | merlinn | and am ideally looking for something that can terminate channelized STM-1 circuits |
| 12:05.00 | merlinn | and convert to sip to talk to my call routing boxes |
| 12:05.11 | *** join/#asterisk suma (n=suma@c-71-196-147-116.hsd1.co.comcast.net) |
| 12:05.35 | merlinn | thanks. |
| 12:07.05 | merlinn | ss7 termination is also possible |
| 12:07.16 | merlinn | but I really donn't know much about that |
| 12:08.06 | gr0mit | 10 000 calls at a time? that is one massive system |
| 12:08.23 | merlinn | yeah it's service provider rather than enterprise |
| 12:08.45 | merlinn | it equates to about 4M minutes a day of local rate calls |
| 12:08.52 | *** join/#asterisk simprix (n=simprix@cosmas.supportdept.com) |
| 12:08.55 | merlinn | with bursty periods |
| 12:09.14 | gr0mit | which country? |
| 12:09.17 | merlinn | UK |
| 12:09.33 | merlinn | are you in the UK yourself? |
| 12:09.36 | gr0mit | yup |
| 12:09.40 | merlinn | ah whereabouts? |
| 12:09.44 | gr0mit | Basingstoke |
| 12:09.46 | merlinn | oh yeah? |
| 12:09.49 | merlinn | I'm in southampton |
| 12:09.55 | gr0mit | cool! |
| 12:10.08 | gr0mit | you an ISP then? |
| 12:10.11 | merlinn | sort of |
| 12:10.17 | gr0mit | voip provider? |
| 12:10.20 | merlinn | sort of |
| 12:10.22 | merlinn | we're a startup |
| 12:10.29 | merlinn | I'm a consultant by trade |
| 12:10.33 | merlinn | network engineer |
| 12:10.44 | merlinn | we started the business doing odds and ends about 2 years ago |
| 12:10.45 | gr0mit | me too |
| 12:10.56 | merlinn | but it all went wrong and we somehow woke up and were a service provider |
| 12:11.19 | merlinn | we bought a datacenter down here a little while ago |
| 12:11.24 | merlinn | and I've just moved the family down from london |
| 12:11.30 | gr0mit | aah nice. |
| 12:11.38 | gr0mit | used to live in Surbiton |
| 12:11.40 | coppice | very few people provide service. you usually need to drag it out of them :-) |
| 12:11.46 | gr0mit | Hants so much nicer |
| 12:11.51 | merlinn | I like it |
| 12:11.55 | merlinn | but it's a shock to the system |
| 12:12.02 | merlinn | moving from chelsea to a suburb of southampton |
| 12:12.08 | merlinn | I think my other half is getting a nose bleed |
| 12:12.30 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 12:12.48 | coppice | you stopped too soon. another few thousand km and the move would really have been worthwhile |
| 12:17.16 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 12:22.08 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583156.dsl.bell.ca) |
| 12:22.40 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 12:24.06 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 12:27.15 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 12:30.57 | *** join/#asterisk ahven (n=kala@194.126.113.140) |
| 12:31.33 | ahven | hi, how to hide callerid just for one number? |
| 12:32.14 | ahven | there was an option in sip.conf, but is deprecated now |
| 12:32.42 | ahven | sip.config* |
| 12:33.20 | kamanashisroy | ahven: setCallerId() in the dialplan ?? |
| 12:33.37 | ahven | dialplan? |
| 12:34.31 | phpboy | :( |
| 12:34.46 | kamanashisroy | ahven: why not you use the SetCallerID() application ? |
| 12:35.14 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 12:35.25 | *** join/#asterisk aksyn (n=aksyn@nat76.mia.three.co.uk) |
| 12:35.33 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 12:35.35 | ahven | because I wasn't aware of it before :) |
| 12:36.01 | rwaite | isnt setcallerid deprecated |
| 12:36.15 | ahven | http://www.voip-info.org/wiki/view/Setting+Callerid |
| 12:36.36 | ahven | this is currently working, http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID is deprecated |
| 12:37.23 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 12:41.02 | dominic1 | hello guys, I have one question if I set hints for the extension 1- 999, will that be a problem? |
| 12:41.10 | kamanashisroy | ahven: I see .. then there is function to set caller id :) |
| 12:41.15 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 12:41.22 | dominic1 | cause I know asterisk can't add hints dynamically |
| 12:42.15 | ahven | kamanashisroy, but does this hide callerid for the analogue networks outside? |
| 12:42.33 | ahven | I wan't to disable CID completely for that number |
| 12:42.35 | ahven | want* |
| 12:42.41 | kamanashisroy | ahven: I think that is out of your control .. |
| 12:43.15 | kamanashisroy | ahven: I *think*, does not mean it is true :) |
| 12:45.56 | ahven | hmm, running asterisk 1.2.7 |
| 12:46.05 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:50.46 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:54.11 | EmleyMoor | Is there anything I can do if I have a couple of data modems sharing the telco line with my FXO port, to detect when it is already in use? |
| 12:54.27 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 12:55.07 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:56.02 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 12:56.03 | ahven | seems that I got the idea how the calls are put together with macros |
| 12:57.06 | [TK]D-Fender | EmleyMoor: What needs to do the detecting, and what decision will be made based on it? |
| 12:57.40 | EmleyMoor | asterisk, and whether to use it, if it would otherwise be appropriate |
| 12:58.12 | *** join/#asterisk delparnel (n=delparne@KTNRON06-1168103470.sdsl.bell.ca) |
| 12:58.31 | kamanashisroy | EmleyMoor: does it support multiplexing ? |
| 12:58.55 | kamanashisroy | EmleyMoor: if not then you will be able to connect to it if it is already in use ! |
| 12:58.55 | EmleyMoor | kamanashisroy: As far as I know, no |
| 12:59.14 | Mavvie | bye |
| 12:59.14 | *** part/#asterisk Mavvie (n=edwin@ppp121-44-55-174.lns10.syd7.internode.on.net) |
| 12:59.23 | *** join/#asterisk DSpair (n=D-Spare@163.muaa.syrc.chcgil24.dsl.att.net) |
| 12:59.27 | EmleyMoor | Yes, that's what can happen - and it's a pain when it does |
| 13:00.04 | kamanashisroy | EmleyMoor: does not it connect to the modem when asterisk starts up ? |
| 13:00.06 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 13:00.26 | kamanashisroy | EmleyMoor: in that case you cannot use the modem as long as asterisk is running .. right ? |
| 13:00.27 | EmleyMoor | I gather that trying to somehow route the data calls through asterisk in order to get round the contention is a non-starter |
| 13:00.50 | [TK]D-Fender | EmleyMoor: I've seen small modules that will block the line if it is in use elsewhere, but thats it |
| 13:00.55 | EmleyMoor | No, not correct. No modem is connected to the asterisk machine |
| 13:01.33 | [TK]D-Fender | EmleyMoor: At which point you'd need to enable call progress to have * become potentially aware of the problem, but I'm not 100% that would even do |
| 13:01.33 | EmleyMoor | [TK]D-Fender: Yes - seen that - but it would just leave asterisk floating in silence |
| 13:01.48 | EmleyMoor | Ah... |
| 13:02.01 | EmleyMoor | Well, it's rarely likely to happen anyway |
| 13:02.21 | DSpair | Hey gang. Got a question: No, wiat, I figured it. |
| 13:05.22 | creativx | sweet |
| 13:05.26 | creativx | how the hell is this possible! |
| 13:05.32 | creativx | both our itsps boxes are down |
| 13:05.57 | *** join/#asterisk pcrack (n=pcrack@122.53.143.105) |
| 13:06.41 | *** join/#asterisk ManxPower (n=manxpowe@121.sub-75-248-102.myvzw.com) |
| 13:07.40 | *** join/#asterisk dieno (i=771e61a2@gateway/web/ajax/mibbit.com/x-2a3e24a712942444) |
| 13:08.06 | pcrack | hi guys...this is not a spam or any scam... |
| 13:08.27 | ahven | spit? :) |
| 13:08.58 | EmleyMoor | pcrack: What isn't? |
| 13:09.08 | pcrack | is there anyone here from US, that can lend me a SIP account, i need to call my mom in california..very urgent...hope you can help me out..even 5-10 mins of call time.. |
| 13:09.49 | pcrack | i know this sound rediculous...but a help will be appriciated.. |
| 13:09.51 | dieno | and where are you from ? |
| 13:10.02 | [TK]D-Fender | pcrack: call her collect |
| 13:10.16 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:10.37 | pcrack | well my local phone doesnt have IDD... |
| 13:10.59 | [TK]D-Fender | pcrack: borrow one that does. |
| 13:11.27 | [TK]D-Fender | pcrack: because I doubt anyone's giving you account details. |
| 13:11.49 | pcrack | yeah i know...but its worth a try... |
| 13:11.58 | [TK]D-Fender | pcrack: the most you could dream of is to have someone bridge a SINGLE call for you. |
| 13:12.30 | *** join/#asterisk coppice (n=chatzill@179.202.17.210.dyn.pacific.net.hk) |
| 13:13.40 | [TK]D-Fender | pcrack: And where are you from? |
| 13:13.47 | creativx | "crack" |
| 13:14.32 | [TK]D-Fender | pcrack: just call collect. |
| 13:14.32 | gr0mit | Philipines IP address.... |
| 13:14.36 | macros73 | Getting multiple NAT retrainsmitting errors in Asterisk now, followed by 'all circuits are busy now.' |
| 13:14.37 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 13:14.55 | [TK]D-Fender | macros73: then go fix them. Read the guide : |
| 13:14.57 | [TK]D-Fender | ~sipnat |
| 13:14.58 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:15.01 | [TK]D-Fender | ^^^^^ |
| 13:15.19 | macros73 | [TK]D-Fender: Thanks for the links, I'll run through those. |
| 13:17.11 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 13:19.54 | dieno | pcrack :D use voipbuster for making free calls |
| 13:20.26 | gr0mit | considers an opportunity..... |
| 13:20.42 | gr0mit | what is the going rate from Philipines to the USA? $2 per min?!! |
| 13:20.50 | gr0mit | soo, 10 mins, $20 |
| 13:21.26 | ahven | use skype even |
| 13:21.32 | dieno | whoa SKYPe offerrs $3 a month for 45K mins t o US & CA dont you thinsk tats betta |
| 13:22.02 | gr0mit | thinks that someone on #asterisk probably knows what they are doing! |
| 13:22.04 | coppice | gr0mit: its best to learn to spell their country if you want to do business with them :-) |
| 13:22.50 | gr0mit | as long as i get cash in my paypal account in advance they can spell it however they want! |
| 13:23.14 | JT | wow what a great way to do business :/ |
| 13:24.38 | ahven | on what commands does the macro break? only on Hangup? |
| 13:24.50 | pcrack | guys im not rich.im just asking who can lend me, if none well thats ok. i have no choice. |
| 13:25.00 | gr0mit | is this the telcom equivilent of the person at the train station begging for £15 to buy a ticket to get to their sick mum in hospital? |
| 13:25.41 | creativx | it looks like it |
| 13:25.44 | creativx | get rich quick scams next! |
| 13:26.13 | gr0mit | hey, please can you wire some funds to this account in Nigeria? |
| 13:26.42 | macros73 | Okay. As a test, I set a static NAT on our Cisco ASA for the test Asterisk server, and opened all UDP ports. Works fine now. Will now tighten down the UDP port range. |
| 13:27.23 | pcrack | ok sorry guys..i know its wrong to ask...but it worth a try..after all we all use asterisk. 1 community |
| 13:27.26 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
| 13:27.51 | Mike8861 | hello all. |
| 13:28.45 | Mike8861 | i head that tomorrow will be independece day~~happy holiday for you all! |
| 13:28.55 | coppice | "can anyone spare an OC-192 for an old soldier, with a sick mother?" |
| 13:29.03 | *** join/#asterisk eXistenZ (n=network1@unaffiliated/existenz) |
| 13:29.21 | gr0mit | yup. We rejoice over here too. |
| 13:29.33 | Mike8861 | gr0mit, are u a US citizen ? |
| 13:29.42 | gr0mit | er nope. |
| 13:29.58 | gr0mit | British. |
| 13:30.01 | dieno | Britian |
| 13:30.06 | kamanashisroy | searching the defender ! |
| 13:30.21 | Mike8861 | oh. cool |
| 13:30.58 | *** part/#asterisk dieno (i=771e61a2@gateway/web/ajax/mibbit.com/x-2a3e24a712942444) |
| 13:31.21 | coppice | tomorrow is no good. we don't get a holiday. two days ago was better |
| 13:31.59 | Mike8861 | i have a question, some user are using ZAP trunk to dial long distance call. what is a good idea of notifying them know, that they are calling via IDD |
| 13:32.13 | ManxPower | IDD? |
| 13:32.20 | Mike8861 | maybe some annoncement ? is there any pre-record sound on this ? |
| 13:32.39 | Mike8861 | ManxPower: IUD = long distance PSTN call |
| 13:32.41 | ManxPower | Mike8861: You do this easily by just doing a playback of a sound file before the Dial |
| 13:32.56 | ManxPower | Mike8861: I thought it was INWARD Direct Dial, not outgoing. |
| 13:33.14 | Mike8861 | ManxPower: sorry for the confusion |
| 13:33.44 | Mike8861 | ManxPower: i wonder if DID and DDI and IDD means the samething = = |
| 13:34.01 | gr0mit | DID = Direct Inward Dialling (US term) |
| 13:34.21 | gr0mit | DDI = Direct Dial In - ( British term for the same thing) |
| 13:34.26 | coppice | DIDN'T == kid's term |
| 13:34.30 | *** join/#asterisk albertoandrade (n=alberto@200.195.161.164) |
| 13:34.32 | gr0mit | IDD = International Direct Dial |
| 13:34.47 | Mike8861 | gr0mit: cool!!! |
| 13:35.28 | macros73 | pcrack: I have a SIP account you can use |
| 13:35.30 | Mike8861 | ManxPower: i understand it can be playback of sound file, is there any pre-recorde sound does that prompt ?? |
| 13:35.41 | macros73 | oh...wait, no, I don't |
| 13:36.01 | Mike8861 | ManxPower: i didnt see something similar in the sound directory |
| 13:36.04 | ManxPower | Mike8861: sounds.txt lists all the included sound files. You can find it in the asterisk source fir |
| 13:36.20 | ManxPower | there is, of course, sounds-extra that you can also download |
| 13:36.37 | gr0mit | Mike8861, just record it yourself! |
| 13:36.43 | gr0mit | 'This call may be expensive!' |
| 13:36.45 | Mike8861 | ManxPower: cool. if i have to record a sound file, what will be good propmt ? |
| 13:36.56 | ManxPower | Mike8861: that is up to you |
| 13:36.58 | Mike8861 | "you are dialing via IDD trunk" something like that ?? |
| 13:37.19 | Mike8861 | ManxPower: ok, thats easy for me to do. |
| 13:37.38 | *** join/#asterisk tobias (n=tobias@cpe-069-134-204-037.nc.res.rr.com) |
| 13:37.47 | Mike8861 | ManxPower: one more question, can we add a BEEP reminder every minute to notify the user ? |
| 13:37.59 | [TK]D-Fender | Mike8861: "core show application dial" |
| 13:39.21 | Mike8861 | [TK]D-Fender: arghhh. thats the command we used yesterday.... |
| 13:40.01 | *** join/#asterisk s0lid (n=s0lid@58.69.91.82) |
| 13:40.03 | eXistenZ | How can I get spa3102 and asterisknow to work together, if spa3102 is on 192.168.0.3 and asterisknow on 192.168.0.4? |
| 13:40.03 | [TK]D-Fender | Mike8861: You'll end up using it al lot. Funny how often you use * to not only accept calls, but to place them. |
| 13:40.16 | [TK]D-Fender | eXistenZ: ... |
| 13:40.17 | [TK]D-Fender | ~book |
| 13:40.18 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 13:40.20 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 13:40.21 | eXistenZ | it is not deteced in the "setup hardware" |
| 13:40.22 | [TK]D-Fender | ~jerjerguide |
| 13:40.23 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 13:40.36 | [TK]D-Fender | eXistenZ: there is no detection. |
| 13:41.04 | Mike8861 | [TK]D-Fender: THANKS, I need to take sometime to learn this |
| 13:48.41 | creativx | god damnit how difficult can it be to buy a damn icecube maker |
| 13:48.41 | JT | a freezer? |
| 13:48.41 | [TK]D-Fender | plastic tray? |
| 13:48.57 | Mike8861 | professional icecube making ? |
| 13:49.17 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-981cac1034c0c151) |
| 13:50.43 | *** join/#asterisk dieno (i=771e61a2@gateway/web/ajax/mibbit.com/x-2a3e24a712942444) |
| 13:51.59 | *** join/#asterisk mbwjr12 (n=kvirc@pal-179-077.itap.purdue.edu) |
| 13:52.26 | mbwjr12 | hey all |
| 13:52.47 | Mike8861 | Hello mbwjr12 |
| 13:52.56 | coppice | people still make ice cubes for a living |
| 13:53.17 | *** join/#asterisk DIABLO3 (n=hsval@66.0.46.210) |
| 13:53.39 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:53.39 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:53.57 | DIABLO3 | hello |
| 13:54.02 | Mike8861 | i guess we can connect ice cube maker with X10 , and connect it with asterisk IVR ??? |
| 13:54.13 | Mike8861 | dial in IVR to make ice cube!!! yay |
| 13:54.39 | mbwjr12 | is anyone here familiar with chan_mobile? I'm having SCO problems... |
| 13:55.20 | eXistenZ | [TK]D-Fender, Am I supposed to route PSTN calls to asterisk? |
| 13:55.24 | kamanashisroy | chan_mobile chan_gsm chan_celliax .. I get confused :) |
| 13:55.58 | Rico29 | does anybody have the SIPDefault.cnf file (for Cisco 7960 provisioning) with all aptions available please ? |
| 13:56.54 | Mike8861 | Rico29: cisco do not likely support feature key on SIP. use MGCP instead |
| 13:56.58 | mbwjr12 | kamanashisroy: this is the bluetooth mobile connection module, will be released in asterisk addons for 1.6 |
| 13:57.04 | [TK]D-Fender | eXistenZ: you're not "supposed" to do anything. * does what you set it up to. |
| 13:57.24 | Rico29 | Mike8861> I can't |
| 13:57.25 | [TK]D-Fender | Mike8861: I set up my coffee maker via X-10 (heyu2) on * |
| 13:57.57 | creativx | JT: an automated ice cube machinae |
| 13:58.07 | JT | commercial? |
| 13:58.15 | Mike8861 | [TK]D-Fender: wow , so cool! i wanna try it out |
| 13:58.16 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:58.48 | Mike8861 | [TK]D-Fender: how to make this happen without x10, x10 is not popular in hongkong, also, theres voltage issue |
| 13:58.55 | creativx | JT: big enough for the office! |
| 13:59.26 | creativx | not a single retailer has one in stock |
| 13:59.30 | creativx | aaaafrikkinmazing |
| 13:59.44 | JT | so consumer then |
| 13:59.45 | Mike8861 | creativx: http://www.google.com/products?hl=en&resnum=0&q=ice%20cube%20maker&um=1&ie=UTF-8&ei=0dlsSMHVI4au6wPihoniAQ&gbv=2&sa=N&tab=if |
| 14:00.56 | [TK]D-Fender | Mike8861: I'm sure there are plenty of other viable home-automation interfaces out there that would work for you. Nothing nearly as generic & inexpensive I'd bet however |
| 14:01.08 | jaytee | lol, I clicked on that link expecting to see some kind of voip hardware. Ice cube makers!!! rofl |
| 14:01.21 | coppice | Mike8861: get a filipino to do it :-) |
| 14:02.43 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 14:02.52 | Mike8861 | coppice, thanks, i will teain them to understand DTMF or voicexml language |
| 14:03.24 | coppice | well, its the standard form of home automation in HK |
| 14:03.28 | kamanashisroy | why asterisk directly need to communicate that .. ? |
| 14:03.41 | Mike8861 | *ROFL* |
| 14:03.42 | eXistenZ | [TK]D-Fender, if I need to block incoming calls, then I need to route pstn (from fxo) calls to asterisk then reroute them to the fxs of spsa3102? |
| 14:03.43 | kamanashisroy | why not AGI ? |
| 14:04.29 | [TK]D-Fender | eXistenZ: Yes |
| 14:04.41 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:07.52 | creativx | Mike8861: thanks for the google search, but i need a local retailer :-) |
| 14:08.05 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.228.111) |
| 14:08.34 | creativx | anyways.. im out! 30 c degrees outside.. summer! wah ahh |
| 14:08.36 | merlinn | does anyone have any recommendations for decent offload devices for TDM termination |
| 14:08.57 | merlinn | tdm & SDH |
| 14:09.01 | *** join/#asterisk r0land (n=r0land@193.227.191.91) |
| 14:09.04 | r0land | hello all |
| 14:09.07 | r0land | hi [TK]D-Fender |
| 14:09.49 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 14:10.01 | [TK]D-Fender | merlinn: SDH? |
| 14:10.18 | merlinn | I'm hoping to terminate using channelized STM-1s |
| 14:10.31 | r0land | im trying to dial through my sip. at the moment i managed to setup a 2 stage dialing! i dial 01, i get a dial tone, and then i dial out.. this is my dialplan ( exten =>_01,1,Dial(SIP/$(EXTEN)@200) ) wht hsould i change to b able to dial a number and call out immediatly |
| 14:11.25 | [TK]D-Fender | merlinn: OUCH |
| 14:11.34 | merlinn | if you have any other suggestions |
| 14:11.35 | merlinn | I'm happy to hear |
| 14:11.39 | r0land | would this work? exten =>_01.,1,Dial(SIP/200) |
| 14:11.41 | merlinn | I need to terminate about 10,000 calls simultaneously |
| 14:12.06 | [TK]D-Fender | r0land: that code does not even function. Pastebin the real dialplan. |
| 14:12.18 | merlinn | and I need to be able to scale significantly higher |
| 14:12.19 | kamanashisroy | r0land: what happend to you ? |
| 14:12.31 | kamanashisroy | r0land: is 200 the trunk ? |
| 14:12.49 | kamanashisroy | merlinn: is that possible ? |
| 14:13.01 | r0land | kamanashisroy yes |
| 14:13.04 | [TK]D-Fender | merlinn: I'm trying to think of the * setup that would be needed to survive anything like that, and you're in a HUG league there... |
| 14:13.13 | merlinn | HUG ? |
| 14:13.22 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:13.22 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:13.23 | [TK]D-Fender | HUGE* |
| 14:13.24 | merlinn | the * setup is very very simple |
| 14:13.37 | r0land | at the moment with this: exten =>_01,1,Dial(SIP/$(EXTEN)@200) , if i dial 01 from my softphone, i get a dial tone from my PSTN which is connection to "200" on my sipura 3102 |
| 14:13.38 | kamanashisroy | r0land: then what is the problem ? |
| 14:13.45 | merlinn | it's heavily distributed |
| 14:13.59 | merlinn | so we assume a box routes about 400 calls simultaneously |
| 14:14.13 | merlinn | it's not doing anything complicated just routing |
| 14:14.18 | [TK]D-Fender | r0land: because you are not passing it a number to dial. |
| 14:14.27 | merlinn | so I need a big box, maybe a cisco or perhaps an SS7 switch |
| 14:14.32 | merlinn | to route stuff to using SIP |
| 14:14.35 | kamanashisroy | r0land: I see .. do you need automated dtmf ? |
| 14:14.50 | [TK]D-Fender | r0land: go learn how to use variables again. |
| 14:14.55 | kamanashisroy | r0land: you mean your trunk needs the destination to be set as dtmf ? |
| 14:15.09 | [TK]D-Fender | kamanashisroy: No. His dialplan is bad. |
| 14:15.27 | r0land | [TK]D-Fender i dont see how my dialplan is bad, since its working! i just need to have a 1 stage dialing instead of 2 |
| 14:15.47 | [TK]D-Fender | r0land: it is bad, because you aren't telling your SPA what # to dial. |
| 14:16.16 | kamanashisroy | r0land: I think there is some syntactic mistake there .. |
| 14:16.19 | eXistenZ | [TK]D-Fender, is it possible to edit the sip.conf through the interface of asterisknow? |
| 14:16.36 | [TK]D-Fender | eXistenZ: this is NOT the support channel for that distro or its GUI. |
| 14:16.44 | r0land | exten =>_01XXXXXX,1,Dial(SIP/$(EXTEN)@200) <<== shouldnt this work! |
| 14:16.59 | [TK]D-Fender | eXistenZ: your attempt to look at * as an easy quick-fix are going to flounder. |
| 14:17.06 | [TK]D-Fender | r0land: No, it shuoldn't |
| 14:17.08 | *** join/#asterisk lotho (n=lotho@static.69.46.46.78.clients.your-server.de) |
| 14:17.15 | [TK]D-Fender | r0land: go learn how to use VARIABLES again. |
| 14:18.04 | tzafrir_laptop | eXistenZ, the asterisk-gui of asterisk-now includes a "file manager", which is a glorified file editor |
| 14:18.11 | tzafrir_laptop | vim is still the best |
| 14:18.18 | tzafrir_laptop | ;-) |
| 14:18.29 | lmadsen | vim++ |
| 14:18.38 | *** join/#asterisk L-info (n=L-info@g0962184.demon.co.uk) |
| 14:18.49 | tzafrir_laptop | Also note that it will probalbly put configurations in users.conf rather than sip.conf |
| 14:18.51 | kamanashisroy | r0land: as the rule says [TK]D-Fender will not be able to tell you the exact issue in the code .. but you know you still have problem using variable :D .. |
| 14:19.14 | [TK]D-Fender | kamanashisroy: I hide things in the big print ;) |
| 14:19.20 | tzafrir_laptop | but if you add your own entries you can use sip.conf normally |
| 14:20.34 | *** join/#asterisk qdk (n=qdk@87.48.132.114) |
| 14:22.30 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:23.48 | *** part/#asterisk ahven (n=kala@194.126.113.140) |
| 14:24.58 | dieno | can any one tell me how to pass a specific value to mysql before making every call |
| 14:24.59 | *** join/#asterisk qdk (n=qdk@87.48.132.114) |
| 14:25.35 | [TK]D-Fender | dieno: there is an entire chapter in the book for func_odbc... |
| 14:25.56 | coppice | vim is OK, unless you have a plastic bathtub |
| 14:26.34 | dieno | ok |
| 14:26.43 | hsv-al | d-fender, it seems that d3 doesnt come out |
| 14:26.46 | hsv-al | till late 2009, argh ! |
| 14:27.49 | [TK]D-Fender | hsv-al: Thats ok... I'll have DN3D to keep me busy until then ;) |
| 14:27.57 | [TK]D-Fender | Sorry... DNF ;) |
| 14:28.18 | kamanashisroy | dieno: write AGI script .. |
| 14:29.17 | [TK]D-Fender | dieno: Or "System", or, or, or, or... |
| 14:30.01 | kamanashisroy | mysql application .. |
| 14:32.06 | kamanashisroy | asterisk could have a jsr .. so that I could write application in java !!! |
| 14:32.27 | *** join/#asterisk ManxPower (n=manxpowe@35.sub-75-250-157.myvzw.com) |
| 14:32.45 | [TK]D-Fender | kamanashisroy: res_java |
| 14:38.55 | *** join/#asterisk scampbell (n=scampbel@35.8.206.106) |
| 14:42.02 | *** join/#asterisk L-info (n=L-info@g0962184.demon.co.uk) |
| 14:47.38 | *** join/#asterisk bmg505 (n=leon@196-209-78-27-tbnb-esr-2.dynamic.isadsl.co.za) |
| 14:47.41 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:48.48 | *** join/#asterisk xpot (n=xpot@144.35.254.4) |
| 14:49.37 | *** join/#asterisk huey23 (n=yea@65.111.241.212) |
| 14:50.55 | huey23 | what's happening? |
| 14:51.04 | *** join/#asterisk spokra (n=spokra@host093-179-132.sea0.speakeasy.net) |
| 14:55.03 | eXistenZ | I am trying to route pstn calls from the fxo of my spa3102 to asterisk and then reroute them to the fxs (line1) of the spa3102. Here are my sip.conf and extensions.conf http://pastebin.com/m1ae6395b and http://pastebin.com/m668454f4 |
| 14:55.07 | eXistenZ | what might be the problemn |
| 14:55.19 | *** part/#asterisk Edder_ (n=edder@201.192.8.198) |
| 14:56.24 | tzafrir_laptop | eXistenZ, please also provide a trace from the CLI (or maybe /var/log/asterisk/full or whatever) |
| 14:56.59 | eXistenZ | tzafrir_laptop, how? |
| 14:57.17 | eXistenZ | tzafrir_laptop, how can I follow up the call? |
| 14:58.51 | *** join/#asterisk neurosys0 (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 14:59.12 | [TK]D-Fender | eXistenZ: Nice to know you don't have a sip.conf entry for your FXS either... |
| 14:59.53 | [TK]D-Fender | nvm |
| 15:00.01 | [TK]D-Fender | poor mashed up mess. |
| 15:00.02 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:00.09 | eXistenZ | =) |
| 15:00.31 | [TK]D-Fender | ath might actually be close to something functional. |
| 15:00.35 | [TK]D-Fender | that* |
| 15:02.37 | eXistenZ | [TK]D-Fender, how can I trace the call in the CLI? |
| 15:02.52 | *** join/#asterisk MihiNomenEst (n=argh@65-85-36-130.client.dsl.net) |
| 15:02.54 | [TK]D-Fender | eXistenZ: go LOOK in CLI. |
| 15:03.18 | [TK]D-Fender | eXistenZ: If you aren't seeing anything at verbose 10, then the call isn't being accepted and you'll have to enable "sip debug" |
| 15:03.44 | dieno | can any one tell me what does that mean app_setcdruserfield.c:121 |
| 15:04.09 | [TK]D-Fender | dieno: sets the "userfield" for that calls CDR record. |
| 15:04.15 | EmleyMoor | dieno: What's the rest of the message? |
| 15:04.33 | dieno | setcdruserfield_exec: SetCDRUserField is deprecated. Please use CDR(userfield) instead. |
| 15:04.39 | [TK]D-Fender | dieno: jsut like the INSTRUCTIONS tell you. |
| 15:04.58 | [TK]D-Fender | dieno: yes... and it tells you the new & proper way you should be calling it. |
| 15:05.05 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:05.08 | EmleyMoor | dieno: It means that SetCDRUserField won't work in the next asterisk |
| 15:05.19 | dieno | hmm how do i enable it |
| 15:05.28 | dieno | should i change something in c file |
| 15:05.36 | [TK]D-Fender | dieno: You don't ENABLE it. You CALL IT from your dialplan. |
| 15:05.40 | EmleyMoor | No - in extensions.conf |
| 15:06.35 | dieno | ok |
| 15:07.06 | EmleyMoor | Change all references to SetCDRUserField(whatever) to Set(CDR(userfield)=whatever), I would think |
| 15:08.11 | dieno | ok let me try |
| 15:08.20 | dieno | http://pastebin.ca/1061370 btw here is my extensions.conf |
| 15:08.31 | MihiNomenEst | what's the name of that site that sells "weird" phone stuff? like the "nuclear hotline" phones and all that? |
| 15:08.54 | [TK]D-Fender | dieno: That fine, although as the warning says, it will not work in 1.6 |
| 15:09.22 | [TK]D-Fender | dieno: For which EmleyMoor already told you exactly how you SHOULD be calling it. |
| 15:09.25 | dieno | its 1.4.20 :D |
| 15:09.41 | *** join/#asterisk BrokenNoze (n=root@host86-150-237-169.range86-150.btcentralplus.com) |
| 15:09.51 | EmleyMoor | dieno: It is now - but the new method works in 1.4 and 1.6 - the old one only in 1.4 |
| 15:10.07 | dieno | it worked |
| 15:10.26 | BrokenNoze | Hi, anyone know why i don't get the BT engaged tone from Asterisk 1.4 when i place an aoutbound call and line is engaged? just dies with DIALSTATUS set, want to hear BT engaged tone |
| 15:10.30 | dieno | thnx for your support guys |
| 15:10.57 | EmleyMoor | I'm still on a 1.2 release - there are things in my dialplan that are known to have been made 1.6 safe already |
| 15:11.19 | [TK]D-Fender | BrokenNoze: pastebin the CLI output of your failed call at verbose 10, and include all relevant configs |
| 15:11.22 | [TK]D-Fender | ~pb |
| 15:11.23 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:11.42 | [TK]D-Fender | ^^^^^^ |
| 15:11.47 | EmleyMoor | BrokenNoze: Is the call going over BT? |
| 15:12.04 | eXistenZ | [TK]D-Fender, is there any error in the config files? |
| 15:12.26 | [TK]D-Fender | eXistenZ: Can't tell. You aren't showing us an ERROR. |
| 15:12.35 | ManxPower | MihiNomenEst: sandman.com |
| 15:12.40 | [TK]D-Fender | eXistenZ: I told you what to look for already |
| 15:13.01 | BrokenNoze | EmleyMoor : Yes, over BT, removed the "r" option on dial. I do get BT annoncements ( such as "this number does not allow calls frm witheld numbers etc) but for somereason not the actual BT engaged |
| 15:13.14 | eXistenZ | [TK]D-Fender, I type sip set debug , but nothing happens when I try to make a call |
| 15:13.22 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
| 15:13.31 | EmleyMoor | BrokenNoze: Hmmmm... I get it... hold on |
| 15:13.34 | gr0mit | BrokenNoze, so you want to get busy tone when the BT line is busy? |
| 15:13.51 | BrokenNoze | yep, thats it.. should be real simple right? |
| 15:13.56 | gr0mit | yup |
| 15:14.12 | gr0mit | well what i think you actually want is congestion, not busy |
| 15:14.14 | [TK]D-Fender | eXistenZ: then your device isn't even talking to * at all. |
| 15:14.14 | BrokenNoze | but the handset just dies after 2 secs |
| 15:14.31 | BrokenNoze | congestion doesn't play back any audio to the handset either from some reason |
| 15:14.42 | EmleyMoor | BrokenNoze: Do you mean when the BT line is in use at your side or when the number you are calling is busy? |
| 15:14.43 | gr0mit | what version of asterisk again? |
| 15:14.58 | BrokenNoze | EmleyMoor : no the called party. |
| 15:15.04 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 15:15.14 | BrokenNoze | 1.4.20 |
| 15:15.27 | gr0mit | and what type of interface? ISDN? |
| 15:15.35 | EmleyMoor | How is the BT line connected? |
| 15:15.42 | BrokenNoze | yep. iSDN |
| 15:15.49 | BrokenNoze | through a Sangoma card |
| 15:15.49 | EmleyMoor | Ah... |
| 15:16.00 | gr0mit | ok pls paste me your zapata.conf files in a pastebin |
| 15:16.10 | EmleyMoor | In zapata.conf, you need to change the setting of priindication, I think |
| 15:16.25 | gr0mit | EmleyMoor, this is the problem i reckon |
| 15:17.10 | gr0mit | priindication=outofband |
| 15:17.14 | gr0mit | is what you need |
| 15:17.19 | BrokenNoze | http://pastebin.com/m152d1c48 |
| 15:17.26 | EmleyMoor | gr0mit: Actially, what he desires is inband |
| 15:17.52 | EmleyMoor | ... or is it? Hard to tell from the notes |
| 15:18.34 | gr0mit | well, if he uses outof band then the tones will be generated locally by Asteirsk |
| 15:18.41 | gr0mit | but he will still hear tones |
| 15:18.44 | ManxPower | EmleyMoor: he may desire inband, but he needs out of band |
| 15:19.00 | BrokenNoze | sorry thats the wrong file, |
| 15:19.12 | gr0mit | BrokenNoze, can't see what you are doing in this file! |
| 15:19.17 | gr0mit | this looks lioke the default |
| 15:19.27 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:19.27 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:19.32 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:19.46 | BrokenNoze | it is, i can't get on the right machone to send you the pastebin link for the right one. on KVM |
| 15:19.56 | gr0mit | hehe |
| 15:20.03 | dieno | EmleyMoor hmm asterisk didnt showed any error while transfering value to CDR userfield but i havent got any entry |
| 15:20.06 | [TK]D-Fender | BrokenNoze: SSH is your friend |
| 15:20.30 | [TK]D-Fender | dieno: And like usual you aren't showing us anything of value. |
| 15:21.05 | gr0mit | here is mine, BrokenNoze |
| 15:21.06 | dieno | ohh ok :D |
| 15:21.07 | gr0mit | http://pastebin.com/m60cd8add |
| 15:21.13 | BrokenNoze | Fender : Doh |
| 15:21.15 | gr0mit | for a BT ISDN2e |
| 15:21.27 | dieno | Set("Local/@from-internal-196c,2", "CDR(userfield)=2424") in new stack |
| 15:22.02 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 15:22.44 | EmleyMoor | That looks right - checked the CDR? |
| 15:23.28 | dieno | but in Master.CSV it shows that i have sent the 2424 |
| 15:23.42 | EmleyMoor | ... so? |
| 15:23.50 | dieno | so its not passing to mysql :D |
| 15:24.08 | [TK]D-Fender | dieno: then you didn't set * up to use mysql properly. |
| 15:24.10 | BrokenNoze | Sorry there you go guys http://pastebin.com/m6570e5c5 |
| 15:24.27 | BrokenNoze | ISDN 30 e |
| 15:24.32 | [TK]D-Fender | dieno: Because it shouldn't be writing CDRs to both at the same time. |
| 15:24.38 | dieno | let me check i think mysql.conf enabled the userfield |
| 15:25.08 | [TK]D-Fender | IIRC |
| 15:25.15 | *** join/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 15:25.39 | *** part/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 15:26.18 | BrokenNoze | i've added priindication=outofband. will i need to restart or will a reload do? |
| 15:26.24 | gr0mit | so just insert the priindication=outofband in the [channels] section and you are good to go |
| 15:26.31 | gr0mit | restart will be better |
| 15:26.49 | BrokenNoze | can't really, live system. I'll give it a go. Cheers gr0mit |
| 15:27.02 | dieno | heeh again thnx guys field was disabled :D in mysql |
| 15:27.10 | gr0mit | where are you, BrokenNoze ? |
| 15:27.13 | gr0mit | jooi |
| 15:27.17 | dieno | i mean in cdr_mysql.conf |
| 15:27.19 | EmleyMoor | BrokenNoze: restart when convenient |
| 15:28.19 | BrokenNoze | gr0mit: 24/7 customer call centre for doctors :-S |
| 15:28.42 | gr0mit | aaah, not NEG? |
| 15:28.45 | BrokenNoze | reload didn't work. just silence. |
| 15:28.49 | BrokenNoze | NEG? |
| 15:28.57 | gr0mit | SurgeryLine |
| 15:29.01 | BrokenNoze | nope |
| 15:29.04 | EmleyMoor | If it won't come to pass that there are no calls, you will have to plan |
| 15:29.32 | BrokenNoze | OK, so i can't do without a restart basically. that does make sense to be honest |
| 15:29.42 | *** join/#asterisk gormux (n=julien@ks361262.kimsufi.com) |
| 15:29.45 | gormux | hello all |
| 15:29.49 | gormux | i have a question |
| 15:30.03 | EmleyMoor | BrokenNoze: Is there ever a time when the lines are clear? |
| 15:30.12 | EmleyMoor | gormux: Just ask |
| 15:30.17 | gormux | does it costs a lot in terms of performances to convert g711a to g711u ? |
| 15:30.58 | [TK]D-Fender | gormux: Very little |
| 15:31.07 | xpot | anyone know what is going on with IAXtel? |
| 15:31.14 | BrokenNoze | EmleyMoor, yeah. 5am! |
| 15:31.27 | gormux | the point is : is it interesting to authorize only g711a rather that g711a and u ? |
| 15:31.27 | BrokenNoze | not hanging round until then |
| 15:31.32 | ManxPower | I didn't know anyone used IAXtel anymore. |
| 15:31.40 | gormux | if the server has load problems already |
| 15:31.48 | gr0mit | BrokenNoze, can you do it overnight? |
| 15:31.50 | ManxPower | gormux: allow only u or a, not both |
| 15:31.51 | EmleyMoor | BrokenNoze: You could issue "restart when convenient" now and it will restart once all the lines become clear |
| 15:31.54 | gormux | ok |
| 15:32.08 | BrokenNoze | it'd be easier just to record a BT engaged tone to be honest |
| 15:32.18 | gr0mit | nah! |
| 15:32.20 | eXistenZ | [TK]D-Fender, Am I supposed to reboot asterisk after every change in the config files? |
| 15:32.20 | ManxPower | gormux: usa/canada = g711 ulaw, most of the rest of the world is G711 alaw. |
| 15:32.22 | [TK]D-Fender | gormux: Always good to standardize. |
| 15:32.25 | gr0mit | just restart when convenient |
| 15:32.30 | EmleyMoor | BrokenNoze: Might PlayTones help? |
| 15:32.35 | [TK]D-Fender | eXistenZ: reboot? |
| 15:32.40 | ManxPower | eXistenZ: no. only a few options require a restart if changed |
| 15:32.55 | xpot | ManxPower: the IAXtel site is up and down, promising a new interface and looks like it wants to come back online... hence my question. |
| 15:32.57 | EmleyMoor | Sometimes you just need to reload the appropriate config |
| 15:33.05 | BrokenNoze | did try playtone. didnt play anything, but neither did Congestion and I thought that was meant to play back a US engaged tone |
| 15:33.23 | ManxPower | BrokenNoze: if you don't see SIP debug output there is nothing we can do |
| 15:33.25 | BrokenNoze | I'll just have to restart when the customers not looking. |
| 15:33.56 | BrokenNoze | yeah, it's in the SIP output, i get busy from the ISDN30, just no audio back to the caller |
| 15:34.09 | gr0mit | BrokenNoze, you can set the locale to present UK tones |
| 15:34.20 | BrokenNoze | I did that in indicator.conf |
| 15:34.22 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-13-121.revip2.asianet.co.th) |
| 15:34.24 | BrokenNoze | indicators.conf |
| 15:34.30 | gr0mit | indications.conf even |
| 15:34.37 | BrokenNoze | thats the bunny |
| 15:34.40 | gr0mit | hehe! |
| 15:34.48 | BrokenNoze | but still nothing |
| 15:35.04 | BrokenNoze | have to just leave them with a recording of my own voice for the time being i think |
| 15:35.12 | BrokenNoze | lukcy them |
| 15:35.19 | gr0mit | not sure that will work |
| 15:35.51 | gr0mit | as i think without the priindication set correctly you will not know what is going on |
| 15:36.02 | BrokenNoze | it has been set to that for last month, they're just sick of my voice |
| 15:36.04 | gr0mit | not sure what the default is |
| 15:36.07 | gr0mit | aaah ok |
| 15:36.26 | gr0mit | so BrokenNoze needs fixing ;-) |
| 15:36.40 | BrokenNoze | it all works fine, just the audio part of the call ain't going through |
| 15:36.48 | gr0mit | just do a restart when convenient |
| 15:37.00 | BrokenNoze | lol, yeah. Glastonbury voice.. still sounds like i smoked 1000 fags |
| 15:37.03 | gr0mit | if it is a real emergency they will always call 999 ;-) |
| 15:37.13 | BrokenNoze | we do 999 call centres too |
| 15:37.14 | EmleyMoor | Not "restart" when convenient but "restart when convenient" |
| 15:37.33 | *** join/#asterisk SparFux (n=raoul@e182021086.adsl.alicedsl.de) |
| 15:37.36 | gr0mit | EmleyMoor, lol! |
| 15:37.51 | BrokenNoze | anyway thanks for the help guys |
| 15:37.53 | gr0mit | BrokenNoze, 999 centres running Asterisk?! |
| 15:37.57 | BrokenNoze | yep |
| 15:38.05 | gr0mit | for BT? |
| 15:38.12 | BrokenNoze | BT? no |
| 15:38.25 | BrokenNoze | BT use Meridians still |
| 15:38.31 | gr0mit | aah ok |
| 15:38.35 | gr0mit | not surprised |
| 15:38.35 | EmleyMoor | My 999 routine was interesting to write |
| 15:38.40 | BrokenNoze | no, just use :-D |
| 15:38.45 | EmleyMoor | (or rather, to modernise) |
| 15:38.56 | BrokenNoze | *s |
| 15:39.03 | BrokenNoze | *us |
| 15:39.11 | BrokenNoze | laters anyway people |
| 15:39.20 | BrokenNoze | thanks for help |
| 15:41.15 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-8bfccf6f4a6abe2b) |
| 15:42.38 | *** join/#asterisk raytruz` (n=raytruz_@74-138-103-238.dhcp.insightbb.com) |
| 15:43.01 | EmleyMoor | Where does the default ring cadence for FXS ports come from? |
| 15:43.16 | raytruz` | Does Asterisk stop trying to renew SIP registrations if internet is lost ? |
| 15:43.31 | *** join/#asterisk DarnoQ (n=d@chello089076192243.chello.pl) |
| 15:44.33 | DarnoQ | hi guys! which g729a codec package from digium is the best for 4x athlon 32bit hardware ? |
| 15:45.35 | *** join/#asterisk neonerz (i=18bb0206@gateway/web/ajax/mibbit.com/x-cd7dc39d2050f576) |
| 15:46.10 | neonerz | Would someone be able to help me with a non-asterisk voip question? |
| 15:46.25 | EmleyMoor | neonerz: Maybe... |
| 15:46.27 | [TK]D-Fender | neonerz: Possibly. Ask. |
| 15:46.57 | neonerz | RFC 1889 says that the SSRC has to be unique throughout the whole RTP session |
| 15:47.10 | neonerz | I just noticed my softswitch is changing the SSRC upon reply |
| 15:47.33 | neonerz | beside the face of increasing the probability that two calls will be using the same SSRC at the same time |
| 15:47.35 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 15:47.45 | neonerz | and totally screwing with my analysis applications |
| 15:47.57 | neonerz | is there any other adverse effects with that? |
| 15:48.33 | neonerz | http://rapidshare.com/files/126726489/070308-ycs_sip_capture-1.1.1.1-486BFA07.pcap.html |
| 15:48.40 | *** join/#asterisk eXistenZ (n=network1@unaffiliated/existenz) |
| 15:48.48 | neonerz | pcap of two sip calls to show what I'm talking about |
| 15:48.56 | eXistenZ | [TK]D-Fender, I get this error chan_sip.c: Registration from 'PSTN <sip:pstn@192.168.0.3>' failed for '192.168.0.4' - No matching peer found |
| 15:49.52 | ManxPower | eXistenZ: you don't have a [PSTN] (or maybe [pstn]) section in sip.conf |
| 15:50.10 | eXistenZ | ManxPower, now I have :) |
| 15:51.17 | raytruz` | Does Asterisk give up on SIP registrations after X amount of time if it cannot reach host? |
| 15:51.52 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:51.55 | raytruz` | And specifically, can I make it not give up? I think the net went down and then the remote registration went down, and I did not receive any phone calls until asterisk restarted. |
| 15:52.46 | eXistenZ | ManxPower, Is it possible to trace registrations other than through Asterisk Logs? |
| 15:53.12 | ManxPower | eXistenZ: you mean like "sip debug"? |
| 15:53.34 | eXistenZ | ManxPower, set sip debug? |
| 15:53.50 | ManxPower | raytruz`: Asterisk will stop working if you are using VoIP and it can't do a DNS lookup for it's own IP addresses. Put the IP addresses of the machine in /etc/hosts. |
| 15:54.02 | ManxPower | eXistenZ: Uh, you can't try it? |
| 15:54.19 | ManxPower | on MY version of Asterisk it's "sip debug on", but I use 1.2 |
| 15:54.27 | eXistenZ | ManxPower, are successful registrations logged? |
| 15:54.39 | [TK]D-Fender | eXistenZ: no. |
| 15:54.43 | ManxPower | eXistenZ: WHY DON'T YOU TRY IT |
| 15:54.47 | raytruz` | ManxPower: Any further reading on that ? |
| 15:54.52 | eXistenZ | :/ |
| 15:54.56 | ManxPower | raytruz`: nope. |
| 15:55.07 | raytruz` | ManxPower: And should i put the ip of the external host in there too? |
| 15:55.14 | raytruz` | Or will it keep trying as long as It can look itself up? |
| 15:55.35 | ManxPower | raytruz`: put the IP addresses that are on the asterisk machine |
| 15:55.54 | eXistenZ | [TK]D-Fender, so basically I receive the pstn from the spa3102, but it doesn't reroute it into the fxs |
| 15:56.06 | eXistenZ | [TK]D-Fender, this should be a problem in the extensions.conf? |
| 15:56.08 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 15:56.27 | [TK]D-Fender | eXistenZ: you aren't showing us a failed call. Stop wasting our time |
| 15:57.23 | eXistenZ | [TK]D-Fender, Where would that be logged, in /var/log/messages ? |
| 15:57.39 | [TK]D-Fender | eXistenZ: ....* CLI <--- |
| 15:57.53 | ManxPower | eXistenZ: It would be logged in the CLI and in /var/log/asterisk/whatever |
| 15:57.54 | [TK]D-Fender | eXistenZ: Forget about "log files" that is worthless for debugging real problems. |
| 15:58.26 | eXistenZ | [TK]D-Fender, how can I run the CLI using ssh? |
| 15:58.36 | EmleyMoor | asterisk -r |
| 15:58.38 | [TK]D-Fender | eXistenZ: "asterisk -r" |
| 15:59.08 | ManxPower | eXistenZ: I think you need to read The Good Book, you are asking questions a person using Asterisk on their first day asks. |
| 15:59.21 | ManxPower | ~book |
| 15:59.22 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:59.22 | raytruz` | ManxPower: how do i put the ip's in the /etc/hosts file, I have always put IP<TAB>localhost or myserver. My external IP does not have a name to resolve to |
| 15:59.26 | eXistenZ | ManxPower, quite so |
| 15:59.34 | ManxPower | raytruz`: then make up a name |
| 15:59.42 | raytruz` | rgr |
| 15:59.49 | raytruz` | just making sure it was that simple :-) |
| 15:59.59 | ManxPower | raytruz`: ONLY in this specific situation |
| 16:00.07 | raytruz` | yeah i figured |
| 16:00.15 | raytruz` | I just wish I could recreate the problem to make sure this fixes it |
| 16:00.29 | ManxPower | raytruz`: break your internet connection, that will test it. |
| 16:00.44 | raytruz` | Yeah, that is one way to do it |
| 16:01.08 | ManxPower | raytruz`: it would not hurt to put the ip/name info in /etc/hosts for the sites your are registering to as well. The only issue is that it will break of they change IPs |
| 16:01.18 | raytruz` | yeah that was my thought |
| 16:01.26 | raytruz` | i'm sure it will stay static since its a voip provider |
| 16:01.27 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 16:01.27 | raytruz` | but still |
| 16:01.37 | [TK]D-Fender | raytruz`: You could always run a script to periodically look it up and update your hosts file. |
| 16:01.45 | raytruz` | true |
| 16:01.55 | raytruz` | 4:00 AM type of script |
| 16:02.05 | [TK]D-Fender | raytruz`: or whenever |
| 16:04.36 | eXistenZ | [TK]D-Fender, should I do anything after I save sip.conf? |
| 16:04.42 | eXistenZ | to get settings working |
| 16:04.45 | [TK]D-Fender | eXistenZ: "sip reload" |
| 16:04.53 | eXistenZ | ah :) |
| 16:05.48 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 16:06.09 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 16:06.55 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:07.42 | eXistenZ | [TK]D-Fender, sip show peers, shows registered peers? |
| 16:08.19 | [TK]D-Fender | eXistenZ: should. do "sip show peer [peernamewithoutbraces]" to see thedetails for one |
| 16:09.21 | eXistenZ | [TK]D-Fender, which property shows that it is connected to asterisk |
| 16:09.31 | [TK]D-Fender | eXistenZ: look at the IP's |
| 16:09.34 | EmleyMoor | Where does the default ringing cadence of an FXS port come from? |
| 16:09.51 | [TK]D-Fender | EmleyMoor: look in your zaptel source |
| 16:10.14 | eXistenZ | [TK]D-Fender, it is dynamic |
| 16:10.28 | [TK]D-Fender | eXistenZ: pastebin the peer dump |
| 16:10.55 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 16:11.09 | *** join/#asterisk snowy_owl (n=snowy_ow@201-048-005-105.static.ctbctelecom.com.br) |
| 16:11.18 | EmleyMoor | Oh, I see what might well be it in zoneinfo.c |
| 16:11.44 | snowy_owl | Alou |
| 16:11.48 | EmleyMoor | zonedata.c even |
| 16:12.20 | tzafrir_laptop | EmleyMoor, you have 'dahdi show cadences', but that's for the custom rings (such as Zap/1r1) |
| 16:12.21 | eXistenZ | [TK]D-Fender, http://pastebin.com/mf56f470 |
| 16:12.51 | [TK]D-Fender | eXistenZ: Addr->IP : (Unspecified) Port 0 <- not registered. |
| 16:12.59 | tzafrir_laptop | EmleyMoor, the data in the sample indications.conf are exactly the same as those in zonedata.c (in practice) |
| 16:13.29 | EmleyMoor | tzafrir_laptop: zap show cadences shows those for me.. actually, what I was seeking to do was confirm the default as I am thinking of rationalising and also amending the one we use on ringbacks |
| 16:14.49 | snowy_owl | People!! I know you all have enough wisdom to give a little hint. Sometimes, my sip proxy sends an INVITE to asterisk and this one try to forward the message to server X. But, sometimes, this last server isnt running. So asterisk will send an error to the first server |
| 16:15.30 | snowy_owl | this sip proxy will load the failure route and will send a new INVITE to asterisk (using the same call-id from the first call). |
| 16:15.45 | snowy_owl | But Asterisk will deny it. |
| 16:16.12 | snowy_owl | is there a variable which I can set to avoid this behavior? |
| 16:17.09 | snowy_owl | all this communication is using SIP. Just sip |
| 16:18.18 | *** join/#asterisk mighty-d (i=500@63.58.83.190.static.coldecon.com) |
| 16:18.18 | snowy_owl | the sip proxy changes the INVITE's RURI and try to send to asterisk again, but it does not accept it at all. |
| 16:18.29 | snowy_owl | I think that dont exist something like this |
| 16:18.45 | snowy_owl | something to change this behavior |
| 16:18.49 | snowy_owl | am i right? |
| 16:21.11 | mighty-d | Hi, i need to deploy asterisk on a 100 extension office with a primary trunk line, i am trying to figure out the size of the machine i need, do you think i should go with expensive multiple xenon boards or use a dual/quad-core processor, i was thinking on 4 GB R.A.M but im not sure if it will suffice, |
| 16:21.28 | Qwell | mighty-d: overkill |
| 16:21.31 | eXistenZ | [TK]D-Fender, pstn now works, not line1 though : |
| 16:21.40 | [TK]D-Fender | mighty-d: And * doesn't run well on inert gasses ;) |
| 16:21.54 | Qwell | and a xenon motherboard might be ridiculously expensive ;) |
| 16:22.14 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 16:22.22 | Qwell | seeing as how it's a gas... |
| 16:22.36 | [TK]D-Fender | Qwell: Not taht expensive... |
| 16:22.44 | [TK]D-Fender | Qwell: jsut not a good conductor ;) |
| 16:22.50 | Qwell | [TK]D-Fender: how are you going to make a board out of gas? :P |
| 16:23.12 | Qwell | (the word you want is Xeon) |
| 16:23.16 | [TK]D-Fender | Qwell: High-pressure reverse sublimation ;) |
| 16:23.25 | mighty-d | LOL |
| 16:23.28 | mighty-d | i understand now! |
| 16:23.40 | mighty-d | yeah, i meant Xeon |
| 16:23.48 | Qwell | mighty-d: but, yeah, that would be overkill for only 100 extensions |
| 16:24.00 | mighty-d | Qwell, what would you suggest? |
| 16:24.02 | Qwell | even 100 simultaneous calls with transcoding shouldn't need that much |
| 16:24.24 | [TK]D-Fender | mighty-d: And you might do just fine on a basic 3ghz+ box with a gig of RAM. Of course I'd ramp it up to a "moderate power" desktop PC level which is nearly double that these days. |
| 16:25.19 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:25.52 | mighty-d | so, you think i should go with a dualcore at 2.5 and 2 Gig RAM?, they will be using the service a lot on the day and i want to be sure... |
| 16:26.10 | Qwell | mighty-d: that'd be sufficient |
| 16:27.21 | mighty-d | Qwell, [TK]D-Fender , thanks a lot, :) |
| 16:27.57 | [TK]D-Fender | mighty-d: TBH I'd put in 4 gig of ram just because its so cheap, and would at least max you out for 32bit |
| 16:27.59 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:28.34 | mighty-d | [TK]D-Fender, yeah, i was thinking the same |
| 16:29.49 | *** join/#asterisk ariel_ (n=ariel_@74.8.35.6) |
| 16:29.53 | ariel_ | Hello everyone |
| 16:30.30 | *** join/#asterisk neoalex (n=chatzill@user-1087rj6.cable.mindspring.com) |
| 16:30.36 | ariel_ | quick question, has anyone here have any info on the ICD? The wiki has nothing really on it. And have not seen any doc's on how to configure it. |
| 16:30.57 | [TK]D-Fender | ariel_: the alternaive queue? |
| 16:31.05 | [TK]D-Fender | lol... funny typo :) |
| 16:31.06 | ariel_ | yes |
| 16:31.16 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 16:31.22 | [TK]D-Fender | ariel_: Its ancient, unmaintained... WGLWAT |
| 16:31.25 | neoalex | hi guys I have a problem getting incoming callerid on IAX: I have this in the user context in iax.conf: callerid="Adrian"<2342> |
| 16:31.28 | ariel_ | [TK]D-Fender, how are things going? |
| 16:31.38 | neoalex | however when I get a call the name is passed but not the number |
| 16:31.43 | ManxPower | neoalex: NEVER put quotes in callerid |
| 16:31.51 | [TK]D-Fender | ManxPower: Sure you do |
| 16:31.57 | neoalex | tried without too |
| 16:31.59 | [TK]D-Fender | ManxPower: Zero issues with that |
| 16:32.02 | ManxPower | also dont' put anything but letters, numbers and the <> |
| 16:32.09 | eXistenZ | [TK]D-Fender, how can I make an extension, that plays a sound file when someone calls? I simply want to try it |
| 16:32.17 | ManxPower | [TK]D-Fender: some cisco phones have problems with it as do some carriers, even if they overwrite it |
| 16:32.32 | [TK]D-Fender | ManxPower: * strips them from what I recall.. |
| 16:32.35 | ManxPower | and quotes - , etc are not part of the callerid |
| 16:32.39 | ManxPower | [TK]D-Fender: not in my experience |
| 16:32.40 | eXistenZ | [TK]D-Fender, in my [pstn] context, in extensions.conf I want to play a soundfile from /var/lib/asterisk/sounds |
| 16:32.50 | [TK]D-Fender | eXistenZ: its your dialplan. go make an extension in it |
| 16:32.55 | ManxPower | eXistenZ: what did I tell you about the book? |
| 16:33.29 | neoalex | ok now it's callerid=Adrian<2342> |
| 16:33.36 | neoalex | still the same thing |
| 16:33.43 | neoalex | it only shows the name not the number |
| 16:33.51 | [TK]D-Fender | neoalex: pastebin is your friend... |
| 16:34.07 | eXistenZ | [TK]D-Fender, do I have to reload extensions.conf as well? |
| 16:34.23 | neoalex | [TK]D-Fender: what do you need me to paste? |
| 16:35.24 | ManxPower | neoalex: add a space between the name and the < just to be safe, but I don't think that is the issue. |
| 16:35.26 | ariel_ | neoalex, I just use: callerid=Reception Station <7600> and works fine |
| 16:35.32 | [TK]D-Fender | neoalex: your config, and the failed call at verbose 10, IAX2 debug |
| 16:35.46 | [TK]D-Fender | eXistenZ: if you change it... dug |
| 16:35.50 | [TK]D-Fender | suh* |
| 16:37.13 | [TK]D-Fender | askldkjasdash |
| 16:37.16 | neoalex | aaah, nevermind I had a number validity thing for that extension (some providers send me CID with a 1 in front, others don't) |
| 16:41.02 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:44.00 | colin2007 | hi |
| 16:46.10 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 16:47.15 | *** join/#asterisk cabbiepete (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
| 16:49.33 | *** join/#asterisk Zuchmir (n=dddddd@ool-18bd3bfc.dyn.optonline.net) |
| 16:49.56 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 16:50.39 | Zuchmir | how can i set * to call out to 2 SIP addresses, and bridge them? |
| 16:51.04 | [TK]D-Fender | Zuchmir: Look up "AMI Originate" and "call files" on the WIKI |
| 16:51.06 | [TK]D-Fender | ~wikis |
| 16:51.07 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 16:52.44 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 16:58.19 | spokra | http://www.microsoft.com/responsepoint/ has micro$oft packaged asterisk!! |
| 16:59.00 | seanbright | spokra: no, they haven't. |
| 16:59.04 | Veggen | Hmm. I wonder. Someone should make a place where you can register hobbys, interests etc., and sign up for automated call-bridging (surprise phones) from people with similar interests :) |
| 16:59.07 | Qwell | or, have they? |
| 16:59.14 | spokra | but the invented VOIP :> |
| 16:59.16 | seanbright | Qwell: they haven't. |
| 16:59.19 | Veggen | (that was in relation to Zuchmirs comments) |
| 16:59.22 | Qwell | or, HAVE they? |
| 16:59.46 | Qwell | spokra: Why would you think it's Asterisk? |
| 16:59.48 | seanbright | Qwell: no sir, they have not. |
| 16:59.57 | mvanbaak | who knows |
| 17:00.01 | mvanbaak | maybe they have |
| 17:00.03 | spokra | I don;t really.. |
| 17:00.09 | seanbright | mvanbaak: no, they haven't. |
| 17:00.24 | mvanbaak | gheh that banner is fun |
| 17:00.37 | spokra | kinda pricy for four phone a a switch |
| 17:00.37 | mvanbaak | 'A phone system so easy, you can buy it online.' |
| 17:00.40 | mvanbaak | oh really ! |
| 17:01.06 | mvanbaak | they should replace that page with a redirect to ebay |
| 17:01.09 | outtolunc | yeah, for $7k |
| 17:01.11 | outtolunc | hehe |
| 17:01.16 | phpboy | :/ |
| 17:01.22 | seanbright | and people will buy it because it's from MS |
| 17:01.37 | seanbright | for the same reason people buy ABE |
| 17:01.39 | phpboy | that's the truth |
| 17:01.57 | phpboy | I best get going |
| 17:02.01 | phpboy | I need to eat :T |
| 17:02.06 | phpboy | it's been a long day :( |
| 17:02.25 | mvanbaak | it's always a long day when coding php |
| 17:02.41 | phpboy | I don't code at this company |
| 17:02.49 | Qwell | blah, just redefine the time constant |
| 17:02.53 | phpboy | well, I do small dev things from time to time |
| 17:02.57 | mvanbaak | lol Qwell |
| 17:03.03 | phpboy | geek :P |
| 17:03.06 | mvanbaak | php coding is my daytime job |
| 17:03.16 | phpboy | It used to be my daytime job |
| 17:03.25 | phpboy | but I've decided to do the systems thing for a while again |
| 17:03.32 | phpboy | I believe I enjoy it more |
| 17:06.19 | *** join/#asterisk robf (n=robf@24.214.206.254) |
| 17:06.40 | *** join/#asterisk JenniferAkemi (n=akemi@76-10-172-166.dsl.teksavvy.com) |
| 17:06.56 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 17:11.24 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 17:11.25 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 17:14.46 | *** part/#asterisk neoalex (n=chatzill@user-1087rj6.cable.mindspring.com) |
| 17:16.55 | *** join/#asterisk s0lid (n=s0lid@122.53.110.157) |
| 17:17.03 | *** join/#asterisk kombi_ (n=kombi@port-87-234-216-47.static.qsc.de) |
| 17:17.35 | *** join/#asterisk Superbartt (n=bart@ip503c8e6d.speed.planet.nl) |
| 17:18.25 | kombi_ | trying to bridge two network interfaces, I managed to break my b140p somehow.. Incoming calls sound like they are switched indefinetly but are never visible in CLI. Is there a way to reset the card somehow? |
| 17:19.53 | [TK]D-Fender | kombi_: kill *, unload the kernel module, reload it. |
| 17:20.34 | kombi_ | I'll reboot the whole box now.. |
| 17:21.38 | *** join/#asterisk TheH (n=tommy@86.43.150.138) |
| 17:21.41 | TheH | hey |
| 17:21.58 | TheH | i have a problem with my BT ISDN / b410p / MISDN config |
| 17:22.13 | TheH | i setup a incoming route but it rings the phone ones and then just drops |
| 17:22.57 | [TK]D-Fender | TheH: pastebin a failed call at verbose 10 |
| 17:22.59 | [TK]D-Fender | ~pb |
| 17:22.59 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:23.01 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 17:24.52 | TheH | TK : http://pastebin.com/m1b8ee8c4 |
| 17:25.01 | TheH | TK : Dont mind the macro :) |
| 17:27.03 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 17:27.09 | *** part/#asterisk colin2007 (n=info@82-171-111-153.ip.telfort.nl) |
| 17:27.34 | *** join/#asterisk [hC] (n=hardcore@190.10.10.76) |
| 17:28.05 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 17:28.44 | [TK]D-Fender | TheH: You've got a cause code 16 in there which seems to indicate that the isdn end hung up. |
| 17:28.53 | TheH | TK : And this is my outgoing call (which also fault.. http://pastebin.com/m3633a426 |
| 17:28.58 | TheH | soo would it be a BT issue ? |
| 17:29.45 | kombi_ | I confess: I did "brctl addif br0 eth0; brctl addif br0 tap0; ifconfig tap0 0.0.0.0 promisc up" After i hit return all ongoing calls were interrupted and the box does not accept or make calls ever since, even after several reboots.. |
| 17:30.13 | kombi_ | I don't see how those brctl commands could affect asterisk or the digium hardware though.. |
| 17:31.22 | TheH | TK : And this is my misdn.conf http://pastebin.com/m767a497f |
| 17:31.35 | seanbright | outtolunc: $7k? try $1,799. |
| 17:31.53 | TheH | TK: i configured it using TE_PTP which seems fine and only default context misdn and then in the misdn context all the ports 1-4 |
| 17:32.46 | *** join/#asterisk fogo (n=fogo@72.8.104.15) |
| 17:32.54 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-182-83-rndf-esr-5.dynamic.isadsl.co.za) |
| 17:33.29 | TheH | TK: Also cause code 16 is normal call clearing which would be weird cause i dont "clear" the call it semes the system clears it before i can pickup |
| 17:34.00 | [TK]D-Fender | TheH: [Jul 3 18:24:41] WARNING[3588]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory |
| 17:34.26 | [TK]D-Fender | TheH: Might be bombing because you don't have a supportable channel the way you're calling from console. Don't waste time dialing like that. use a softphone. |
| 17:34.27 | TheH | TK : Thats ok , thats just because there is no sound card in the server. i have this on a bunch of other servers as well |
| 17:35.27 | vgster | ~centos52bug |
| 17:35.27 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 17:35.37 | TheH | TK: I also added a misdn check before my calls |
| 17:35.42 | TheH | TK exten => s,1,misdn_check_l2l1(g:intern|2) |
| 17:35.42 | TheH | exten => s,2,Dial(misdn/g:intern/${ARG1}) |
| 17:36.43 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 17:36.50 | [TK]D-Fender | TheH: I'm not particularly knowledgeable about ISDN (esp BRI). Still remove 1 potential isse by testing with a softphone on one end. |
| 17:37.22 | TheH | yes did this as well... but same problem occurs.. :( and i did do about 6 other BRI installs for various companies.. |
| 17:37.32 | TheH | i think its a BT Telco issue |
| 17:38.03 | [TK]D-Fender | TheH: Possible, I can't really advise further. Sorry |
| 17:38.09 | *** join/#asterisk keith4__ (n=keith@d-65-175-190-83.cpe.metrocast.net) |
| 17:38.29 | TheH | TK No worries thanks |
| 17:39.10 | huey23 | [TK]D-Fender: are you speechless? |
| 17:39.20 | puzzled | speexless |
| 17:39.31 | keith4__ | VoicePulse provides sample configurations to use with their service. I guess I have to set my outgoing callerid, and the line they suggest is "exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=0000000000)", and similar for 'name' instead of 'num'. Is this syntax correct for 1.4? |
| 17:39.42 | [TK]D-Fender | huey23: No, I've got a professional writer on-call 24/7 :) |
| 17:39.49 | huey23 | [TK]D-Fender: :) |
| 17:39.50 | *** join/#asterisk Great_Anta_Baka (i=c419fff6@gateway/web/ajax/mibbit.com/x-fda0f952038e40bf) |
| 17:40.01 | [TK]D-Fender | keith4yes |
| 17:40.09 | x86 | guys I just found some awesome MoH music :) |
| 17:40.11 | x86 | <PROTECTED> |
| 17:40.36 | keith4__ | no quotes or anything? doesn't seem to be working |
| 17:41.52 | [TK]D-Fender | keith4__: And you don't seem to be showing anything. Curious parallels. |
| 17:42.02 | ManxPower | keith4__: Telcos don't accept outgoing callerid name stuff |
| 17:44.32 | keith4__ | [TK]D-Fender: http://rafb.net/p/lC7Wdd30.html |
| 17:45.00 | keith4__ | top is an excerpt from Voicepulse-provided conf |
| 17:45.35 | keith4__ | I'm sip'd in, calling out to my cell phone. call comes through without CID |
| 17:46.07 | [TK]D-Fender | keith4__: You have a US cell? |
| 17:46.12 | keith4__ | yes |
| 17:46.22 | [TK]D-Fender | keith4__: Located in the US? |
| 17:46.30 | keith4__ | yes |
| 17:46.49 | keith4__ | server in CA, I'm currently in NH |
| 17:46.51 | Great_Anta_Baka | x86: that is some mad mad mad stuff |
| 17:46.52 | [TK]D-Fender | keith4__: Ok, well first, why do you have a "+" in front of the # you are dialing? |
| 17:46.55 | Great_Anta_Baka | love it to bits |
| 17:47.28 | keith4__ | [TK]D-Fender: that is a good question. let me investigate |
| 17:47.34 | [TK]D-Fender | keith4__: And have you just done another test from something like a land-line to confirm that you do get CID in the first place? |
| 17:48.57 | keith4__ | it's in their sample conf: http://rafb.net/p/syB9E835.html |
| 17:48.57 | keith4__ | yes, I get CID from the land-line here, and from other cells |
| 17:50.02 | [TK]D-Fender | keith4__: shouldn't have a + |
| 17:50.50 | [TK]D-Fender | keith4__: and what do you see for the #? |
| 17:51.30 | keith4__ | call log on cell phone shows: Name: "Unknown" Number "No Caller ID" |
| 17:51.38 | keith4__ | I'll take out the + and see what happens |
| 17:53.00 | keith4__ | doesn't make a difference |
| 17:53.26 | keith4__ | call still comes through, though. so obviously the + is frivolous |
| 17:54.44 | [TK]D-Fender | keith4__: pastebin the SIP debug for your call. |
| 17:54.55 | keith4__ | odd that they only have the + in two of the dial statements, but not the third: https://connect.voicepulse.com/samples/extensions.sample |
| 17:55.06 | keith4__ | [TK]D-Fender: ok |
| 17:59.12 | *** join/#asterisk quaqo (n=quaqo@89-97-102-198.ip17.fastwebnet.it) |
| 18:00.54 | keith4__ | [TK]D-Fender: http://rafb.net/p/JI12kv86.html I'm NAT'd behind 65.175.190.83, server is 206.71.169.49 |
| 18:02.36 | [TK]D-Fender | keith4__: I'm suspecting you have a "fromuser=" in your sip.conf entry... |
| 18:03.20 | [TK]D-Fender | keith4__: If so, comment it out and retest |
| 18:04.48 | keith4__ | checking |
| 18:05.10 | keith4__ | ah, crap |
| 18:05.15 | keith4__ | it's in their sample conf! |
| 18:05.46 | keith4__ | https://connect.voicepulse.com/samples/sip.sample |
| 18:05.47 | keith4__ | bastards |
| 18:06.47 | [TK]D-Fender | keith4__: is that to say that that's what it was and is now funcitoning? |
| 18:06.52 | [TK]D-Fender | functioning* |
| 18:07.10 | *** join/#asterisk pputman- (n=centrex@c-68-62-214-146.hsd1.al.comcast.net) |
| 18:07.57 | keith4__ | it's to say... indeed there is a fromuser, but I can't make an outbound call with fromuser commented out |
| 18:08.41 | kombi_ | misdn show stacks says L2Link DOWN. I need that to be up for it to work, any ideas? |
| 18:08.57 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 18:09.11 | keith4__ | http://rafb.net/p/cbQrjT91.html |
| 18:09.43 | keith4__ | which means... it's not registering to voicepulse without 'fromuser'? |
| 18:10.02 | [TK]D-Fender | keith4__: calling has nothing to do with registereing. PB your config |
| 18:10.03 | keith4__ | oh, it is. all 3 show as sip peers |
| 18:10.37 | keith4__ | ok |
| 18:13.12 | *** join/#asterisk MACscr (n=Mark@c-98-214-107-162.hsd1.il.comcast.net) |
| 18:13.29 | MACscr | any recommendations on a quality wifi sip phone? |
| 18:15.34 | [TK]D-Fender | MACscr: Bset out there is the Hitachi's to date |
| 18:15.39 | [TK]D-Fender | best* |
| 18:15.52 | keith4__ | [TK]D-Fender: http://rafb.net/p/8Se11c31.html |
| 18:16.00 | MACscr | [TK]D-Fender: model? |
| 18:16.15 | keith4__ | shit |
| 18:16.20 | [TK]D-Fender | MACscr: IP3000 IIRC |
| 18:16.21 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-52-188.lns10.syd7.internode.on.net) |
| 18:16.24 | MACscr | thanks |
| 18:16.25 | *** part/#asterisk trafim (n=reallyma@212.200.84.70) |
| 18:18.32 | keith4__ | [TK]D-Fender: sorry, http://rafb.net/p/bVL7Zo83.html |
| 18:18.41 | keith4__ | that's without the 'fromuser' lines, obviously |
| 18:19.38 | *** join/#asterisk Great_Anta_Baka (i=c636caf6@gateway/web/ajax/mibbit.com/x-df49f29f4e4054db) |
| 18:19.45 | [TK]D-Fender | keith4__: try swapping commenting out the fromuser & username |
| 18:19.59 | keith4__ | ok |
| 18:21.43 | keith4__ | no good. fails to auth |
| 18:21.50 | keith4__ | but fails differently from before |
| 18:22.05 | keith4__ | http://rafb.net/p/DVAfzU74.html |
| 18:23.45 | [TK]D-Fender | keith4__: ok, I'm out of ideas for a bit |
| 18:24.13 | keith4__ | alright, well thanks for the effort. i'll call voicepulse later, and see if this is just detined to fail |
| 18:24.37 | outtolunc | have you tried setting the CALLERID(name)= (to nothing) |
| 18:25.47 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 18:25.47 | keith4__ | nope. will try |
| 18:26.30 | keith4__ | wtf. why did that work?! |
| 18:26.47 | keith4__ | outtolunc: thanks! |
| 18:26.54 | outtolunc | because they do not expect the invite to a context/user |
| 18:27.00 | [TK]D-Fender | keith4__: Guess they don't like the name. |
| 18:27.10 | outtolunc | gafachi is like that also |
| 18:27.57 | keith4__ | weird |
| 18:28.24 | [TK]D-Fender | outtolunc: Good call. |
| 18:28.43 | keith4__ | now that I look at their sample conf, they have 000000000 for name and num |
| 18:29.36 | outtolunc | the 'name' will always be looked up by the far end anyways.. so it is just passing to the provider you are clearing it out |
| 18:30.24 | outtolunc | if pstn <G> |
| 18:33.29 | *** part/#asterisk MACscr (n=Mark@c-98-214-107-162.hsd1.il.comcast.net) |
| 18:33.54 | keith4__ | thanks guys. this calls for a beer. later! |
| 18:34.41 | outtolunc | happy 4th |
| 18:34.58 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:38.29 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.58.30) |
| 18:39.27 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 18:45.49 | *** join/#asterisk anthm (n=anthm@mbc0736d0.tmodns.net) |
| 18:47.29 | *** join/#asterisk eXistenZ (n=HOME@unaffiliated/existenz) |
| 18:47.34 | eXistenZ | hey [TK]D-Fender |
| 18:47.47 | *** join/#asterisk AC-Jay (n=Jay@rrcs-24-106-28-178.west.biz.rr.com) |
| 18:48.33 | AC-Jay | greetings. can I get help here if my issue lies with asterisk suddenly not running AGI script properly? |
| 18:49.24 | [TK]D-Fender | AC-Jay: perhaps. PASTEBIN is your friend. |
| 18:49.26 | [TK]D-Fender | ~pb |
| 18:49.26 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:50.43 | AC-Jay | well, I don't think it's a code issue either in my dialplan or my php script. if I run the script from the command line, it works fine. when I have asterisk call it via AGI, nothing seems to happen. unforunately, agi debug doesn't show me the info I'd like to see (setting and retrieving variables) |
| 18:51.50 | outtolunc | check the permissions |
| 18:51.58 | AC-Jay | honestly, I'm not sure exactly what the problem is, however I do have the same problem on a new asterisk box I built to replace the one in question. I'm sure I did something wrong, but I can't figure it out |
| 18:52.19 | AC-Jay | outtolunc: tried that already. even tried setting everything to 0777 and no luck |
| 18:52.44 | outtolunc | 755 |
| 18:52.56 | AC-Jay | that's what they are currently set at |
| 18:52.59 | outtolunc | and make sure the user:group is that of your asterisk service |
| 18:53.18 | AC-Jay | much to your chagrin (I'm sure), asterisk is running as root |
| 18:53.21 | AC-Jay | ducks |
| 18:53.41 | AC-Jay | I haven't had much luck running it as non-root. seems to screw everything up for me |
| 18:54.51 | outtolunc | did you put it in your /var/lib/asterisk/agi-bin dir or path it out in the dialplan? |
| 18:55.19 | eXistenZ | [TK]D-Fender, how can I check out whether passing the caller id from the spa to * works? |
| 18:55.31 | AC-Jay | outtolunc: yes. in /var/lib/asterisk/agi-bin |
| 18:55.49 | AC-Jay | the CLI says it's launching the script and that it completed, returning 0 |
| 18:55.59 | outtolunc | well then pastebin the script |
| 18:56.03 | outtolunc | so we can see it |
| 18:56.08 | AC-Jay | ok one sec |
| 18:56.25 | AC-Jay | keep in mind if I run the script from the command line, I get no errors |
| 18:57.35 | *** join/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 18:57.52 | UCFmethod | ~centos52bug |
| 18:57.53 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 18:58.18 | [TK]D-Fender | eXistenZ: LOOK at it. |
| 18:58.34 | *** join/#asterisk Drag0n` (n=Drag0n@net20.quickoffice.com) |
| 18:59.14 | eXistenZ | where |
| 18:59.39 | eXistenZ | [TK]D-Fender, is there a special ckl |
| 18:59.44 | eXistenZ | command for it |
| 18:59.53 | AC-Jay | outtolunc: http://pastebin.com/d7953be77 |
| 19:00.06 | [TK]D-Fender | eXistenZ: look at the SIP traffic, or NoOp it in your dialplan. |
| 19:00.22 | twisted | AC-Jay: what does agi debug tell you? |
| 19:00.43 | kombi_ | after long years of seamless functioning, I can't get L2Link up on a Digium b410. Is this card broken? |
| 19:00.51 | eXistenZ | [TK]D-Fender, sip traffic in sip set debug? |
| 19:00.59 | outtolunc | in is in, not under |
| 19:01.00 | [TK]D-Fender | eXistenZ: yes |
| 19:01.04 | AC-Jay | nothing, and that's the problem. it used to show the assigning and retrieval of variables. it says nothing of the sort now. some info that I dont' think is relevant, but I can pb it if you'd like |
| 19:01.20 | twisted | looks like you're setting things twice |
| 19:01.35 | twisted | not that that's an issue |
| 19:01.50 | AC-Jay | yeah I read about a bug on the phpagi bug tracker that suggested trying that if you were having problems |
| 19:02.02 | AC-Jay | something about it not taking the first time for whatever reason |
| 19:02.37 | *** join/#asterisk kc7wsu (i=80bb00b2@gateway/web/ajax/mibbit.com/x-e008d843392340cd) |
| 19:03.12 | *** join/#asterisk minime (n=afg_ch@84-73-144-128.dclient.hispeed.ch) |
| 19:03.22 | [TK]D-Fender | AC-Jay: $command = "GET /voip/voip.php?cmd=get_cid&data=$acct[data] HTTP/1.1"; <- why do I feel this is not the correct way to reference a variable in-line in PHP? |
| 19:03.55 | [TK]D-Fender | AC-Jay: Do you see your webserver bing hit for the correct query? |
| 19:03.59 | [TK]D-Fender | being* |
| 19:04.28 | twisted | did you run php -l on the script to make sure it's got no syntax errors? |
| 19:04.45 | AC-Jay | [TK]D-Fender, that is the correct syntax, and no, no hits from my webserver |
| 19:04.50 | AC-Jay | unless I run it from the command line |
| 19:05.24 | AC-Jay | twisted: yes |
| 19:05.26 | [TK]D-Fender | AC-Jay: And you're not running PHP in "quiet mode" |
| 19:06.22 | AC-Jay | I've tried both |
| 19:06.31 | AC-Jay | with and without the -q flag |
| 19:06.44 | AC-Jay | which I guess doesn't really exist if you look at the output php provides |
| 19:06.59 | outtolunc | cp the darn thing to the agi-bin dir and restart asterisk |
| 19:07.13 | huey23 | [TK]D-Fender: i looked at lumen and sphinx yesterday, i think i just need to worry about getting a good stable system before i worry about that stuff...there is quite a bit that goes on behind the scenes |
| 19:07.13 | AC-Jay | outtolunc: tried that too |
| 19:07.18 | outtolunc | add a verbose statement before sock |
| 19:07.40 | shido6 | if I do a du -h | grep G > bigfiles.txt and I just want to find out the folders/files that are taking up "Gigs" of space, whats the next step? |
| 19:07.49 | AC-Jay | outtolunc: verbose statements don't do anything, I don't believe. like echoing something won't show in asterisk |
| 19:08.14 | outtolunc | if you have agi debug on yes it does |
| 19:08.19 | outtolunc | (or should) |
| 19:08.37 | AC-Jay | hmm. let me try again |
| 19:09.10 | outtolunc | i use verbose stuff in perl agis all the time |
| 19:09.38 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 19:09.48 | *** part/#asterisk kc7wsu (i=80bb00b2@gateway/web/ajax/mibbit.com/x-e008d843392340cd) |
| 19:10.01 | *** join/#asterisk kc7wsu (i=80bb00b2@gateway/web/ajax/mibbit.com/x-e008d843392340cd) |
| 19:10.11 | AC-Jay | outtolunc: echoing "hello world" didn't show anything in the agi debug |
| 19:10.22 | outtolunc | i didn't say echo |
| 19:10.30 | outtolunc | shakes his head |
| 19:10.37 | BCS-Satori | Has anyone used a wireless sip phone, that is user friendly, that sounds and performs well for a business enviorment |
| 19:11.37 | AC-Jay | sorry, I must have misunderstood what you said |
| 19:11.41 | AC-Jay | what did you mean? |
| 19:11.43 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 19:11.48 | outtolunc | read the phpagi api |
| 19:11.55 | outtolunc | search 'verbose' |
| 19:12.00 | [TK]D-Fender | BCS-Satori: ... |
| 19:12.02 | [TK]D-Fender | ~wifivoip |
| 19:12.03 | jbot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 19:12.22 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 19:12.33 | [TK]D-Fender | BCS-Satori: Unrecommended. If you have little choice, go for the Polycom/Spectralink or Hitachi |
| 19:13.34 | twisted | [TK]D-Fender: heh, i have both here, the user frendliest of the two would be the hitachi |
| 19:13.37 | huey23 | [TK]D-Fender: how come noone recommends a Cisco? |
| 19:13.45 | BCS-Satori | [TK]D-Fender: Yea, I was looking at the Spectralink ones, I don't like it anymore then you do, but they really want one for travel |
| 19:14.13 | [TK]D-Fender | BCS-Satori: Travel? Exactly what they are BAD at. |
| 19:14.17 | kc7wsu | I personally like my Aastra 57i CT |
| 19:14.27 | huey23 | we had an Aastra |
| 19:14.30 | huey23 | it died |
| 19:14.32 | [TK]D-Fender | BCS-Satori: see "nat traversal / security settings, etc" |
| 19:14.38 | AC-Jay | outtolunc: nothing on the verbose |
| 19:14.44 | [TK]D-Fender | kc7wsu: that isn't WiFi |
| 19:16.21 | kc7wsu | True, but if you can find a wifi that will even work very well, why not just goto an ata or aastra with something that will work better. unless you have a huge area an multiple networks to cross |
| 19:16.49 | [TK]D-Fender | kc7wsu: should follow what he's been saying... |
| 19:16.50 | BCS-Satori | [TK]D-Fender: what do you think of bluetooth phones if they exist, have looked yet, that connect to a laptop that has a vpn tunnel back to the pbx? |
| 19:17.06 | [TK]D-Fender | BCS-Satori: Can definietly work. |
| 19:17.18 | [TK]D-Fender | BCS-Satori: Hec, just a BT headset & USB dongle. |
| 19:18.22 | outtolunc | AC-Jay: if your CLI is at verbose higher than your verbose statement and nothing is showing asterisk is not executing it.. did you use the extension inthe dialplan? |
| 19:20.26 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.139) |
| 19:22.18 | *** part/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 19:23.05 | AC-Jay | outtolunc: cli is at 5, $agi->verbose defaults to 1. nothing in asterisk, I assumed it's not executing the script for some reason. I cannot figure out why, however. yes, I called the extension. |
| 19:23.59 | AC-Jay | this is driving me nuts. it's probably something very simple and yet I can't seem to figure it out. ugh |
| 19:24.03 | [TK]D-Fender | AC-Jay: PASTEBIN <--------- |
| 19:24.21 | AC-Jay | you bet. one sec. cli scrolls a lot ;) |
| 19:24.35 | *** join/#asterisk iNetForce (n=f@adsl-074-246-021-235.sip.mia.bellsouth.net) |
| 19:27.03 | AC-Jay | curse the stupid CLI. grrr |
| 19:28.23 | AC-Jay | [TK]D-Fender, outtolunc: http://pastebin.com/d66656074 |
| 19:28.34 | AC-Jay | was not quick enough to get the initial dial command |
| 19:28.44 | UCFmethod | Has anyone played around with app_sms and gotten it to work? I think I need a SMSC configured, but I am not quite sure. |
| 19:28.54 | AC-Jay | got everything after it however, before the stupid CLI autoscrolled and screwed me over :D |
| 19:30.08 | AC-Jay | I'm not 100% sure how to fully decipher an agi debug so I'm hoping I missed something simple and you guys will taunt me mercilessly for it |
| 19:31.34 | minime | scratching head |
| 19:33.12 | AC-Jay | and here's the get_cid.php code http://pastebin.com/d72ca0b3b |
| 19:33.41 | AC-Jay | the double assigning is deliberate |
| 19:35.30 | twisted | hey AC-Jay |
| 19:35.40 | AC-Jay | yes/ |
| 19:35.41 | AC-Jay | ? |
| 19:35.43 | twisted | http://phpagi.sourceforge.net/phpagi2/docs/__examplesource/exsource__root_phpagi-2.14_examples_weather.php_b1b0e9f9a44f9db10926291632cfeaf4.html |
| 19:35.44 | *** part/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 19:35.49 | twisted | check that out for examples on pulling data from the web |
| 19:36.18 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.228.111) |
| 19:36.19 | huey23 | [TK]D-Fender: i need help adding an extension, where should i start? :) |
| 19:36.37 | AC-Jay | my code did work up until a couple hours ago |
| 19:36.42 | AC-Jay | then mysteriously stopped |
| 19:36.43 | [TK]D-Fender | huey23: Step 1) Pull your head out of your ass. |
| 19:36.46 | [TK]D-Fender | :D |
| 19:37.00 | twisted | AC-Jay: hmmm.... sounds fishy |
| 19:37.07 | twisted | AC-Jay: drive isnt' full is it? |
| 19:37.16 | AC-Jay | no |
| 19:37.23 | AC-Jay | and calling the script from the command line works |
| 19:37.26 | twisted | no logfiles 4gb or larger? |
| 19:37.27 | huey23 | [TK]D-Fender: wow .. thanks, that helps |
| 19:37.34 | AC-Jay | so it doesn't appear to be anything on my webserver |
| 19:37.35 | *** join/#asterisk fnordus (n=dnall@70.71.224.2) |
| 19:37.48 | AC-Jay | no, I run logrotate on everything |
| 19:37.49 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 19:38.06 | twisted | something had to change for things to stop working |
| 19:38.13 | twisted | what was the last thing you did on the system before it stopped? |
| 19:38.19 | [TK]D-Fender | huey23: It should. This allows for enhanced vision by removing everything obstructing your eyes! |
| 19:38.35 | anonymouz666 | anyone ever saw asterisk 1.4.21.1 consuming 9%/12% (looking through top) even if there's NO USE |
| 19:38.42 | AC-Jay | on the asterisk box, nothing. on my webserver, recompiling apache |
| 19:38.42 | anonymouz666 | CPU |
| 19:38.44 | *** join/#asterisk L-info (n=L-info@g0962184.demon.co.uk) |
| 19:39.22 | AC-Jay | I'd be inclined to think it was a problem with my webserver, if everything else didn't work fine |
| 19:39.36 | AC-Jay | including running the script *on the asterisk box* from the command line |
| 19:40.23 | iNetForce | Guys call parking is not giving me the extesion where the call is parks |
| 19:40.39 | huey23 | [TK]D-Fender: what country are you located? |
| 19:40.43 | iNetForce | i know it goes to 701 because it is the default first extesion but i dont get the information from the box |
| 19:40.45 | iNetForce | any idea? |
| 19:42.07 | [TK]D-Fender | iNetForce: What "information"? |
| 19:42.30 | *** join/#asterisk rabelais (n=blank@unaffiliated/rabelais) |
| 19:42.41 | macros73 | Anyone recommend software that will high-light phone numbers in web pages and allow click-to-dial for your Asterisk extension? |
| 19:43.06 | [TK]D-Fender | macros73: There are firefox extensions for this. Easily searchable. Go look. |
| 19:43.26 | outtolunc | AC-Jay: where are you loading the phpagi classes (because the AGI isn't in that file you pasted) |
| 19:43.28 | iNetForce | Fender I do not get the extension where the call was parked |
| 19:43.39 | macros73 | [TK]D-Fender: Firefox extensions I can find. I've been looking for the past couple of hours for Internet Explorer-compatible software. |
| 19:43.44 | iNetForce | I am supposed to hear " the car was parked at 701" or something like that |
| 19:43.55 | AC-Jay | outtolunc: are you asking to see the dialplan? |
| 19:43.59 | iNetForce | i do not get to hear that on the line, howerver, the call parks |
| 19:43.59 | [TK]D-Fender | macros73: If you can't find it... maybe it simply doesn't exist |
| 19:44.04 | iNetForce | if i call 701 the call is there |
| 19:44.06 | AC-Jay | or did you want the php code? |
| 19:44.15 | macros73 | [TK]D-Fender: Maybe it's in a box somewhere along with your tact. :D |
| 19:44.30 | [TK]D-Fender | iNetForce: How are you doing this park? describe the phones, and the complete process. |
| 19:44.44 | iNetForce | i have gxp2000 phones |
| 19:44.54 | iNetForce | i call one of the extension and pick up the call there |
| 19:45.01 | outtolunc | AC-Jay: http://sourceforge.net/project/showfiles.php?group_id=106629&package_id=114938 |
| 19:45.09 | [TK]D-Fender | macros73: http://www.google.ca/search?hl=en&q=internet+explorer+asterisk+click-to-dial&btnG=Google+Search&meta= |
| 19:45.12 | iNetForce | i go to the phone where the call was originated from and hit transfer 700 |
| 19:45.26 | iNetForce | the car goes to the parking lot but i do not hear where it was parked |
| 19:45.32 | AC-Jay | out, what was the link for? |
| 19:46.05 | outtolunc | think of it it this way |
| 19:46.06 | [TK]D-Fender | intralanman: What phone model? |
| 19:46.13 | [TK]D-Fender | iNetForce: rather |
| 19:46.38 | drfreeze | Ok, I'm getting a short high pitch on my phones, then a hangup |
| 19:46.48 | macros73 | [TK]D-Fender: Thanks. I've been working with IPdialer, was hoping to find something better...no such luck unless I convince the corporate masters to adopt Firefox. |
| 19:46.49 | outtolunc | if i write an perl based agi that is meant to use asterisk-perl, i would have to install asterisk perl or make its classes available somehow to that agi |
| 19:47.08 | AC-Jay | I'm sorry, I misunderstood. I am using PHPAGI |
| 19:47.19 | outtolunc | you have a php script 'named' phpagi but no phpagi classes (at least not shown) |
| 19:47.35 | AC-Jay | line 6. $agi = new AGI(); |
| 19:47.45 | outtolunc | yeah, magic |
| 19:47.52 | outtolunc | think about that for a sec |
| 19:47.59 | AC-Jay | phpagi.php has the agi classes |
| 19:48.10 | outtolunc | not the file you pasted |
| 19:48.37 | outtolunc | oh i think i got it now |
| 19:48.58 | iNetForce | Am i doing it wrong>? |
| 19:49.03 | outtolunc | you got the phpagi.php in the subdir and some other php file in agi-bin |
| 19:49.15 | [TK]D-Fender | iNetForce: DETAILS. <- |
| 19:49.35 | iNetForce | gxp2000 phones,digium aa50 |
| 19:49.42 | iNetForce | the appliance |
| 19:49.48 | iNetForce | default configuration |
| 19:49.54 | Qwell | iNetForce: Call support. |
| 19:49.59 | [TK]D-Fender | iNetForce: Make sure you are doing an ATTENDED transfer, and not a BLIND transfer. |
| 19:50.24 | AC-Jay | outtolunc: phpagi has several files, so I put them all in the phpagi directory |
| 19:50.56 | iNetForce | i call from 6000 to 60001, I pick up the call at 6001 and from 6000 i press transfer and dial 700. I wait there until the call goes to the parking lot, however, Isupposed to hear on the handset where the callgets parked " the call is parked at 701" but I do not hear this |
| 19:51.23 | iNetForce | am i doing it wrong |
| 19:51.34 | [TK]D-Fender | iNetForce: I just told you something specific to look at. |
| 19:54.28 | outtolunc | AC-Jay: btw.. it is working here, you sure you chown root:root /var/lib/asterisk/agi-bin/phpagi/ and also chmod 755 /var/lib/asterisk/agi-bin/phpagi/phpagi.php and reloaded asterisk |
| 19:55.00 | AC-Jay | I've done both. I will try it again quick once people get off the phones :) |
| 19:55.46 | outtolunc | there is an invalid command but.. it does run, agi debug does output and so does using verbose |
| 19:55.48 | outtolunc | AGI Rx << VERBOSE "test test" 3 |
| 19:55.48 | outtolunc | <PROTECTED> |
| 19:56.05 | macros73 | http://www.tttelecom.nl/ProductenDiensten/Producten/AsteriskDialAnnounceToolADAT/tabid/418/language/en-US/Default.aspx may work |
| 19:56.12 | *** part/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 19:56.19 | AC-Jay | what is the invalid command? |
| 19:56.45 | outtolunc | you think asterisk tells you that .. haha <G> that would be useful |
| 19:56.59 | AC-Jay | oh I thought you were telling me :P |
| 19:57.05 | outtolunc | have to comment out lines or move the verbose in sets to find it |
| 19:57.44 | AC-Jay | christ. my bosses are like freakin highschool girls |
| 19:57.51 | AC-Jay | get off the damn phone! |
| 19:58.07 | *** join/#asterisk TrentCreek (n=kvirc@red1.cs.panam.edu) |
| 19:58.22 | AC-Jay | I'm half tempted to just restart asterisk on him |
| 19:58.39 | TrentCreek | try it on her instead |
| 19:59.06 | outtolunc | doesn't like the #!/usr/sbin/php |
| 19:59.28 | outtolunc | actually other way around |
| 20:00.27 | AC-Jay | are you telling me or suggesting that may be the problem? |
| 20:00.34 | AC-Jay | my shebang line matches "which php" |
| 20:01.07 | outtolunc | yeah without either it doesnt run at all |
| 20:01.08 | TrentCreek | shebangs shebangs shemoves |
| 20:01.14 | outtolunc | but with it gives the error |
| 20:01.23 | AC-Jay | what was the error? |
| 20:01.44 | outtolunc | probably doesn't like the extra stuff i'm tossing at it.. |
| 20:02.16 | AC-Jay | nothing, and no call to my webserver either. |
| 20:02.35 | AC-Jay | chowning and chmoding and restarting didn't fix it |
| 20:03.13 | TrentCreek | now what would be the advantage of setting up mySQL just for *? |
| 20:04.11 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 20:04.28 | [TK]D-Fender | TrentCreek: What do you want to DO with it? |
| 20:05.43 | *** join/#asterisk iNetForce (n=f@adsl-074-246-021-235.sip.mia.bellsouth.net) |
| 20:05.53 | TrentCreek | well...I was just going over the install guide at voip-info and it was metioning about setting it up for it...as an option |
| 20:05.56 | iNetForce | FEnder i figured it out but it takes too much time |
| 20:06.11 | iNetForce | I need to put line 1 on hold, select line and dial 700, then hit trasnfer and line 1 |
| 20:06.30 | iNetForce | is there an easier way to do an attended transfer with a gxp2000? |
| 20:06.34 | [TK]D-Fender | iNetForce: You need to do an ATTENDED transfer. |
| 20:06.50 | [TK]D-Fender | iNetForce: Go read your phone's manual |
| 20:07.26 | [TK]D-Fender | TrentCreek: forget about mysql until you see a real need for it. |
| 20:07.52 | TrentCreek | [TK]D-Fender: Groovy..thanks...well back to setting up asterisk2billing |
| 20:13.00 | robf | most awesome use of hello world http://kerneltrap.org/node/6715 |
| 20:16.11 | *** join/#asterisk murdock_ut (n=chatzill@70.99.184.194) |
| 20:20.35 | TrentCreek | thank you |
| 20:20.55 | *** part/#asterisk drwatson (n=afg_ch@84-73-144-128.dclient.hispeed.ch) |
| 20:22.50 | huey23 | [TK]D-Fender: have fun blowing shit up, making people feel worthless, and being the zen master of the utterly obvious |
| 20:22.57 | *** join/#asterisk mike-ekim (n=digiport@72-19-13-198.idstelcom.net) |
| 20:22.58 | huey23 | [TK]D-Fender: i'm out |
| 20:23.15 | mike-ekim | I have some .vox files, if I have asterisk with allow=ulaw on a provider, will the .vox file play out properly or would I need to convert to .ulaw format |
| 20:23.32 | Qwell | mike-ekim: I don't know what .vox format is. You'll need to convert them to something |
| 20:23.40 | mike-ekim | oh. |
| 20:23.50 | [TK]D-Fender | huey23: whee! |
| 20:24.23 | TrentCreek | i'm in |
| 20:24.28 | [TK]D-Fender | Qwell: Really ancient "standard" |
| 20:24.43 | huey23 | [TK]D-Fender: how do you make your definition come up? ~tk? |
| 20:25.10 | TrentCreek | float |
| 20:25.11 | *** join/#asterisk wigyanpy (n=wigyanpy@120.89.104.6) |
| 20:25.25 | *** part/#asterisk wigyanpy (n=wigyanpy@120.89.104.6) |
| 20:25.52 | *** join/#asterisk browser (n=afg_ch@84-73-144-128.dclient.hispeed.ch) |
| 20:26.17 | Qwell | "Cat 5 Resistant Data Center". wtf? |
| 20:26.28 | huey23 | who? |
| 20:26.38 | Qwell | that's one of the oddest whois comments I've ever seen |
| 20:26.45 | [TK]D-Fender | Qwell: Means that no cat5 will be able to enter. you wouldn't want the hackers to be be able to get to them, do you? :) |
| 20:27.02 | Qwell | huey23: Your provider. very odd. |
| 20:27.24 | [TK]D-Fender | Alrighty, heading home, BBIAB |
| 20:27.26 | huey23 | ohh, we are rated category 5 |
| 20:27.27 | outtolunc | category 5 as in mucho wind <G> |
| 20:27.34 | huey23 | yea |
| 20:27.41 | Qwell | oh, haha |
| 20:27.46 | huey23 | :) |
| 20:27.53 | Qwell | that makes *SO* much more sense |
| 20:28.02 | Qwell | I was thinking like...you know...cat5 |
| 20:28.10 | huey23 | hurricane, tornado, and earthquake |
| 20:28.17 | Qwell | yeah, I got it |
| 20:28.20 | huey23 | :P |
| 20:28.26 | Qwell | you can see the confusion, I'm sure |
| 20:28.45 | seanbright | damn el nino |
| 20:28.54 | huey23 | i can, but if you are in the data center business, you can differentiate |
| 20:29.00 | Qwell | I suppose |
| 20:29.12 | TrentCreek | mean..you need at least cat 6 to enter ;-) |
| 20:29.21 | Qwell | TrentCreek: indeed ;) |
| 20:29.34 | huey23 | we would like it that way, but, you can only push customers so far :P |
| 20:30.11 | Qwell | huey23: so, what makes it cat5 resistant? |
| 20:30.17 | outtolunc | would take thinnet if it were free <G> |
| 20:30.19 | huey23 | the structure |
| 20:30.20 | TrentCreek | gosh darn it..looks like a trip to Fry's tomorrow |
| 20:30.24 | Qwell | no windows, brick, etc? |
| 20:30.25 | huey23 | just the way it's built |
| 20:30.52 | Qwell | and who determines that? is it tested? |
| 20:30.53 | seanbright | popsicle sticks and rubber bands |
| 20:31.16 | *** join/#asterisk ta^3 (n=tacvbo@189.151.76.105) |
| 20:31.17 | huey23 | i am not completly sure, it is 18" reinforced concrete, that's about all i know |
| 20:31.45 | TrentCreek | Maybe it's cheaper to have a host that is in a place that has no bad weather...maybe like Arizona? |
| 20:31.56 | Qwell | TrentCreek: good luck getting bandwidth |
| 20:32.06 | huey23 | NO! ...there are no high winds there |
| 20:32.09 | Qwell | (the answer, of course, is Silicon Valley) |
| 20:32.26 | TrentCreek | yeah..they just got power outages |
| 20:32.35 | macros73 | TrentCreek: Scorpions in AZ, they get into everything. And when you try to clean them out, wow baby, wear thick gloves. |
| 20:32.36 | huey23 | and illegals |
| 20:32.51 | ta^3 | Trying to start Asterisk with the minimum amount of modules (modules.conf: autoload = no), I've discovered that Asterisk is unable to bridge both Zap calls if res_features.so is not loaded |
| 20:32.52 | TrentCreek | hehe |
| 20:32.52 | outtolunc | we got lots of fires currently <G> |
| 20:33.07 | Qwell | ta^3: this is correct |
| 20:33.11 | Qwell | (and documented) |
| 20:33.11 | TrentCreek | then they can post: "We are Scorpion and Illegal Resistant |
| 20:33.14 | huey23 | welll, we have rain |
| 20:33.33 | huey23 | is anything illegal resistant right now? |
| 20:33.50 | Qwell | "illegal"? |
| 20:33.58 | TrentCreek | no...just like the ants |
| 20:34.06 | TrentCreek | get into everything |
| 20:34.10 | huey23 | lol |
| 20:34.21 | ta^3 | Qwell: Hum, I'm ashamed to ear that, where I can find where it's documented? |
| 20:34.30 | huey23 | Qwell: to be blunt...Mexican Nationals |
| 20:34.39 | *** join/#asterisk Segnale007 (n=Segnale0@host163-249-dynamic.23-79-r.retail.telecomitalia.it) |
| 20:34.39 | Qwell | huey23: Take the bigotry elsewhere. |
| 20:34.50 | huey23 | no bigotry, just honesty |
| 20:34.56 | TrentCreek | Stating facts is not bigotry |
| 20:35.04 | huey23 | sorry, i'll go be honest somewhere else |
| 20:35.32 | Qwell | ta^3: good question |
| 20:35.39 | Qwell | I'm pretty sure the book talks about it |
| 20:35.40 | TrentCreek | So calling a drug dealer would be better to sya he is an "Unlicensed Pharamsist?" |
| 20:35.49 | huey23 | :) |
| 20:36.12 | AC-Jay | what is the command to disable agi debugging? |
| 20:36.18 | huey23 | TrentCreek: take that nasty mouth elsewhere please |
| 20:36.25 | huey23 | cya ladies |
| 20:36.37 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 20:37.23 | Qwell | ta^3: most modules should have a dependency on res_features |
| 20:37.53 | Qwell | ta^3: what version? |
| 20:38.16 | ta^3 | Qwell: 1.4.21.1 |
| 20:38.54 | ta^3 | AC-Jay: > agi debug off |
| 20:38.54 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 20:39.33 | TrentCreek | agi flame on |
| 20:40.19 | AC-Jay | ta^3: that did not work |
| 20:41.04 | AC-Jay | nm got it. agi no debug |
| 20:41.11 | TrentCreek | http://www.asteriskguru.com/tutorials/cli_cmd_14.html |
| 20:41.28 | TrentCreek | agi debug off - Disable AGI debugging |
| 20:42.23 | ta^3 | AC-Jay: seems that you are still using 1.2.X |
| 20:43.20 | AC-Jay | I am on my old box. |
| 20:43.28 | AC-Jay | guess I shoulda mentioned that, eh? :P |
| 20:43.30 | TrentCreek | .093 beta |
| 20:46.28 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 20:50.50 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:52.21 | *** part/#asterisk AC-Jay (n=Jay@rrcs-24-106-28-178.west.biz.rr.com) |
| 20:53.55 | pikachu2000 | hi all |
| 20:54.23 | pikachu2000 | Anyone here heard of a problem where asterisk disconnects when a remote pbx does a call transfer? |
| 20:54.51 | [TK]D-Fender | pikachu2000: What "remote PBX", and what kind of "transfer"? How are they conencted? |
| 20:56.34 | pikachu2000 | I dont know the make of the remote pbx but typically it goes like this 1) A call is placed to a customer/supplier/3rd party 2) The third party rep answers. They then try and transfer the call to someone else at the 3rd party. When they do the transfer asterisk disconnects |
| 20:57.03 | pikachu2000 | i.e. the third party is doing a transfer and asterisk disconnects |
| 20:57.37 | pikachu2000 | I cant see anything in the log files but typically by the time I get notified of the problem the logs has got a lot more entries in it. |
| 20:57.56 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 20:58.29 | [TK]D-Fender | pikachu2000: And I asked how they were connected. |
| 20:58.56 | pikachu2000 | over a pri line |
| 20:59.03 | TrentCreek | Cat 5 resistence is fu-tile |
| 20:59.36 | pikachu2000 | the asterisk box has pri -- the other side depends on their setup. |
| 20:59.39 | [TK]D-Fender | pikachu2000: You're going to have to provide CLI output for the failure possibly with PRI debug info as well. |
| 21:00.06 | *** join/#asterisk fetcher (n=jnh@ip-72-55-165-168.static.privatedns.com) |
| 21:00.46 | fetcher | anyone using Voicepulse Connect for IAX termination? |
| 21:01.16 | pikachu2000 | ok --- I will try and turn logging up a bit and see what I can see. I was just hoping someone would have a pointer where to possible look for the problem. I must admit that I havent seen this at other clients. Its just this one who is reporting it. They recently did an upgrade of asterisk |
| 21:01.50 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 21:01.54 | pikachu2000 | i.e trixbox so who knows what they done. I hate it when clients get adventerous. I will check trixbox forums too. |
| 21:02.48 | [TK]D-Fender | pikachu2000: Well.. you have no details, so there is nothing you can ask anyone at this point. |
| 21:02.53 | *** join/#asterisk angryuser (n=sdfsdf@78.115.250.180) |
| 21:03.14 | angryuser | the beep only to voicemail goes with 's" option ? |
| 21:03.28 | angryuser | * 1.4 |
| 21:03.52 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 21:03.55 | [TK]D-Fender | angryuser: If you don't tell it to play a prompt, yes |
| 21:04.04 | *** join/#asterisk hi365_m (n=hi365@213.151.56.96) |
| 21:04.13 | pikachu2000 | true -- the log files i say didnt have anything of interest in them -- will have to dig deeper |
| 21:04.37 | [TK]D-Fender | pikachu2000: CLI ouptu, not logs. Log files are typically worthless |
| 21:04.54 | angryuser | yep, thanks |
| 21:05.12 | pikachu2000 | how do you capture the cli output though. This might sound stupid but it scrolls by really fast |
| 21:05.19 | pikachu2000 | ? |
| 21:06.09 | angryuser | read faster ? xD |
| 21:06.20 | [TK]D-Fender | pikachu2000: copy & paste |
| 21:07.08 | riddlebox | wohoo, finally got remote access to my last asterisk install |
| 21:07.10 | pikachu2000 | phew --- its a busy asterisk box. It normally scrolls off screen before I have a chance to highlight and do the copy. |
| 21:07.29 | pikachu2000 | I will see if we can reproduce it in the lab |
| 21:07.33 | angryuser | ow , have you heard about elastix, their last release is nice |
| 21:07.35 | pikachu2000 | with less traffic |
| 21:07.43 | fetcher | pikachu2000: one way is to run your Asterisk console isnide a GNU 'screen' session |
| 21:07.53 | [TK]D-Fender | angryuser: centos + FreePBX. nothing to write home about |
| 21:07.54 | angryuser | for the *gui* based asterisk |
| 21:08.04 | pikachu2000 | hi fetcher yes can do that |
| 21:08.06 | fetcher | pikachu2000: then press control-A, 'H' (capital) to turn on screen's logging feature |
| 21:08.17 | outtolunc | set your logger.conf for full (add verbose), then reload it |
| 21:08.18 | pikachu2000 | ok -- cool thanks for the tip |
| 21:08.34 | [TK]D-Fender | or pick any client that has a "copy all to clipboard". |
| 21:08.42 | [TK]D-Fender | (PuTTY) |
| 21:08.55 | Qwell | or just use the logger features built into Asterisk |
| 21:08.58 | angryuser | [TK]D-Fender: but they have extra options like vtiger/sugar/spark;openfire/hylafax & more integrated |
| 21:09.13 | angryuser | i think it's nice |
| 21:09.15 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 21:09.35 | pikachu2000 | Will try --- but is really difficult with an intermittent fault. I will see what I can do. Will need to put a lot more time into solving this one. |
| 21:10.19 | angryuser | [TK]D-Fender: good for lazy persons like me |
| 21:10.46 | outtolunc | fooboot's <G> |
| 21:11.17 | pikachu2000 | yeah we use elastix at one of our clients. its nice because it uses misdn by default while trixbox --- last time I looked doesn't |
| 21:11.46 | angryuser | ah yes misdn is a plus |
| 21:12.03 | angryuser | for *some* installs |
| 21:12.23 | Zuchmir | is there a way i can do a Dial(SIP/user1...) and Dial(SIP/user2...) on the same call (or equivelent - to cause an outgoing conference when a call come in)? |
| 21:12.40 | angryuser | pikachu2000: any stability issues ? |
| 21:13.37 | [TK]D-Fender | Zuchmir: try "Page". Not sure if its bi-directional though. |
| 21:14.01 | pikachu2000 | nope -- solid as a rock -- no calls from client. They were using trixbox and we used to battle a bit as we had to recompile everything to support misdn -- this meant no updates etc. When their box died from a disk failure we took the opportunity to put in elastix |
| 21:14.11 | Zuchmir | tk: thanks |
| 21:16.01 | *** join/#asterisk eXistenZ (n=HOME@unaffiliated/existenz) |
| 21:16.05 | angryuser | pikachu2000: i am thinking to install one soon, with couple b410p's so misdn really matter for me, thanks for info, btw, do they use callback with internal agenda ? |
| 21:16.39 | riddlebox | is there a way to tell if someone is on any zap channels? |
| 21:16.46 | eXistenZ | [TK]D-Fender, even if I enable sip set debug, it displays nothing in the CLI when I make a pstn call, what might be the reason |
| 21:16.52 | [TK]D-Fender | riddlebox: "cor show channels concise" |
| 21:17.33 | pikachu2000 | hey angryuser -- I haven't used it with the internall agenda or call back feature |
| 21:17.49 | [TK]D-Fender | eXistenZ: if sip debug is enabled and you don't see anything, then either youu have a networking (firewall,etc) problem (which you shouldn't), or you didn't set you device up right to talk to * |
| 21:18.26 | eXistenZ | [TK]D-Fender, I haven't even enabled the firewall |
| 21:18.39 | eXistenZ | [TK]D-Fender, and the device works quite well, I mean the extensions work |
| 21:18.41 | angryuser | pikachu2000: if they coul improuve it a bit (import csv, multi level acces) it would be a really nice one |
| 21:18.58 | [TK]D-Fender | eXistenZ: No SIP traffic means NO TALKING. It is clearly not right. |
| 21:18.59 | angryuser | improve |
| 21:19.16 | pikachu2000 | What exactly does "internal agenda" do? |
| 21:19.36 | eXistenZ | [TK]D-Fender, ah, there should be an answer? |
| 21:19.38 | angryuser | <pikachu2000> standard callback with .call generated |
| 21:19.49 | eXistenZ | [TK]D-Fender, I mean, there should be a connection between the pstn and line1? |
| 21:19.57 | [TK]D-Fender | eXistenZ: Your device is not set up right. |
| 21:20.01 | angryuser | <pikachu2000> but it is pretty basic now |
| 21:20.08 | [TK]D-Fender | eXistenZ: no, they do not conenct to each other. |
| 21:21.20 | eXistenZ | [TK]D-Fender, http://pastebin.com/mc62ccd6 |
| 21:21.31 | eXistenZ | [TK]D-Fender, that shows that the pstn is set up correctly |
| 21:21.54 | pikachu2000 | anyone here used vicidial? |
| 21:22.11 | *** part/#asterisk oej (n=olle@ns.webway.se) |
| 21:22.22 | [TK]D-Fender | eXistenZ: No, that shows only a small part of the picture. |
| 21:22.39 | eXistenZ | [TK]D-Fender, it doesn't show the caller id there though |
| 21:22.56 | [TK]D-Fender | eXistenZ: that doesn't show me anything concerning your problem. |
| 21:23.26 | eXistenZ | [TK]D-Fender, how can I find out what's the problem |
| 21:23.37 | [TK]D-Fender | eXistenZ: Fix your settings on the device |
| 21:24.08 | eXistenZ | [TK]D-Fender, which device |
| 21:24.16 | eXistenZ | [TK]D-Fender, the spa or the asterisk |
| 21:24.40 | [TK]D-Fender | eXistenZ: If you are placing a call on the device and * doesn't see it its your DEVICE |
| 21:25.05 | eXistenZ | [TK]D-Fender, why can I see it in the message logs |
| 21:25.21 | [TK]D-Fender | eXistenZ: what "logs"? |
| 21:26.21 | ManxPower | eXistenZ: Perhaps you misunderstand. It appears your device is not even TRYING to connect to Asterisk |
| 21:26.43 | eXistenZ | ManxPower, doch doch, check out the ip |
| 21:27.15 | [TK]D-Fender | eXistenZ: "check out the ip"? |
| 21:27.50 | eXistenZ | yeah, in http://pastebin.com/mc62ccd6 |
| 21:28.12 | [TK]D-Fender | eXistenZ: [TK]D-Fender>eXistenZ: No, that shows only a small part of the picture. <--- |
| 21:28.41 | [TK]D-Fender | eXistenZ: there are OTHER settings to tell the SPA to send incoing calls to * |
| 21:28.43 | gr0mit | which decvice is this? |
| 21:28.48 | gr0mit | ah, SPA |
| 21:28.50 | gr0mit | ugh. |
| 21:29.05 | gr0mit | took me a long time to get this talking to asterisk |
| 21:31.01 | eXistenZ | [TK]D-Fender, how about '[Jul 3 19:25:24] NOTICE[3362] chan_sip.c: Call from 'pstn' to extension '12' rejected because extension not found. |
| 21:31.02 | eXistenZ | ' |
| 21:31.10 | eXistenZ | I typed a wrong extension to get this message |
| 21:31.41 | [TK]D-Fender | eXistenZ: If you see nothing in CLI with SIP debug, then you haven't enabled it. |
| 21:32.09 | [TK]D-Fender | eXistenZ: go make sure you're at verbose 10, with sip debug enabled |
| 21:32.23 | eXistenZ | how can I change it to verbose 10 |
| 21:32.50 | [TK]D-Fender | eXistenZ: "set verbose 10". |
| 21:35.03 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 21:35.37 | eXistenZ | [TK]D-Fender, weird, still nothing |
| 21:35.43 | eXistenZ | I will try it using putty |
| 21:35.46 | eXistenZ | ssh |
| 21:35.58 | [TK]D-Fender | eXistenZ: And what were you doing before? |
| 21:36.13 | eXistenZ | through web interface |
| 21:36.21 | [TK]D-Fender | eXistenZ: GARBAGE |
| 21:36.52 | [TK]D-Fender | eXistenZ: and "web interface" doesn't say aWHAT you were looking at in it. |
| 21:36.55 | eXistenZ | [TK]D-Fender, what is this error: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 21:37.13 | angryuser | *headshot* |
| 21:37.15 | [TK]D-Fender | eXistenZ: means either * isn't running, or the user you are logged in as has no rights |
| 21:37.24 | pikachu2000 | ok got some log message that may be relevant 'Transfered/Local/300@from-internal-64e0,2<ZOMBIE>' in macro 'dial' |
| 21:38.23 | eXistenZ | [TK]D-Fender, << Asterisk already running on /var/run/asterisk.ctl. Use 'asterisk -r' to connect. |
| 21:38.24 | eXistenZ | <PROTECTED> |
| 21:38.38 | [TK]D-Fender | eXistenZ: the go connect |
| 21:38.48 | eXistenZ | ah now |
| 21:38.48 | eXistenZ | with sudo |
| 21:38.50 | angryuser | *headshot* |
| 21:39.11 | [TK]D-Fender | angryuser: Get a bigger gun. |
| 21:39.20 | angryuser | i miss everytime |
| 21:39.30 | [TK]D-Fender | angryuser: You should work on that... |
| 21:39.34 | angryuser | ah! blank bullets |
| 21:39.47 | eXistenZ | [TK]D-Fender, works in ssh :) |
| 21:39.49 | [TK]D-Fender | angryuser: Only so many free Stormtrooper positions left these days... |
| 21:40.24 | eXistenZ | web interface is a real crap |
| 21:40.28 | eXistenZ | I though it works :/ |
| 21:40.52 | [TK]D-Fender | eXistenZ: Sure ti does.. see how much of our time its wasted? |
| 21:40.59 | angryuser | i will think about it |
| 21:41.09 | pikachu2000 | mmm --- seems that it might be a freepbx issue |
| 21:41.10 | eXistenZ | [TK]D-Fender, http://pastebin.com/m6d0e947 |
| 21:42.01 | eXistenZ | [TK]D-Fender, the number is there :) << 0527342620@192.168.0.3 >> |
| 21:42.10 | *** part/#asterisk Rem| (n=remi@bg-fw2out.monmouth.com) |
| 21:42.17 | [TK]D-Fender | eXistenZ: Looking for 12 in pstn (domain 192.168.0.3) SIP/2.0 404 Not Found <- says it all. Go look and find out why you don't have an exten to match what they dialed. |
| 21:42.29 | eXistenZ | I know |
| 21:42.40 | eXistenZ | there is extension 1234 |
| 21:42.42 | eXistenZ | I was just testing |
| 21:43.06 | [TK]D-Fender | eXistenZ: ....... |
| 21:43.41 | eXistenZ | [TK]D-Fender, I want simply to divert anonymous callers to /dev/null |
| 21:43.49 | eXistenZ | :p |
| 21:44.22 | [TK]D-Fender | eXistenZ: "that's nice" |
| 21:44.22 | eXistenZ | I will try to find out how |
| 21:45.25 | [TK]D-Fender | eXistenZ: congratulations on wasting a complete ^#%$ing hour on NOTHING. |
| 21:45.39 | eXistenZ | [TK]D-Fender, the web interface is to blame ;( |
| 21:46.27 | [TK]D-Fender | eXistenZ: and you've been warned to keep out of that garbage. |
| 21:48.19 | *** part/#asterisk SparFux (n=raoul@e182021086.adsl.alicedsl.de) |
| 21:49.02 | eXistenZ | [TK]D-Fender, Well, thank you for everything, I should hop to bed now, 12:48AM over here :o |
| 21:49.02 | riddlebox | [TK]D-Fender, I have a site that uses zaptel-1.4.9.2, do you think it is needed to keep it updated, or as long as it works let it go? |
| 21:49.06 | eXistenZ | tomorrow I will try to find out how to block anonymous calls |
| 21:49.31 | [TK]D-Fender | riddlebox: if it works, why change it... |
| 21:49.31 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:50.32 | *** join/#asterisk hardwire (n=hardwire@rdbk-9064.mtaonline.net) |
| 21:50.56 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 21:51.14 | eXistenZ | [TK]D-Fender, are you from the us? |
| 21:51.32 | [TK]D-Fender | eXistenZ: No. |
| 21:55.29 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:59.21 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 22:12.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:13.50 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) |
| 22:18.43 | *** join/#asterisk bsaxon (n=bsaxon@249.sub-75-202-41.myvzw.com) |
| 22:23.28 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 22:24.11 | *** join/#asterisk xenonex (n=xenonex@89.218.235.89) |
| 22:26.04 | *** join/#asterisk dlynes (n=daniel@S01060016b68219f1.vs.shawcable.net) |
| 22:27.29 | *** join/#asterisk adorah (n=Michael@87.69.57.141.cable.012.net.il) |
| 22:29.53 | dlynes | ~centos52bug |
| 22:29.54 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 22:36.17 | *** join/#asterisk Superbartt (n=bart@ip503c8e6d.speed.planet.nl) |
| 22:37.03 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 22:38.06 | *** join/#asterisk Great_Anta_Baka (i=c419fff6@gateway/web/ajax/mibbit.com/x-9a013f85dc49896c) |
| 22:38.30 | dlynes | ~xppbug |
| 22:38.43 | dlynes | ~zaptel14bug |
| 22:39.01 | browser | g'evening pals. I want to give music on hold from an internet .mp3 streaming source. I am on * 1.2.27. The doku on voip-wiki is at least outdated if not confusing. |
| 22:39.59 | dlynes | browser: icecast |
| 22:41.03 | browser | dlynes: I know icecast as a streaming server. are you positive it can pick up a stream from an external source and re-stream it to * ? |
| 22:41.26 | dlynes | browser: yes...afaik, asterisk has had that capability since asterisk 1.0 |
| 22:41.49 | dlynes | browser: you just have to make sure that format_mp3.so is installed |
| 22:42.04 | browser | dlynes: ok. let me do some googling... |
| 22:42.25 | dlynes | browser: try some voip-info'ing instead |
| 22:45.49 | browser | dlynes: icecast takes me back to where I was; http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf#Exampleusingicecastampshoutcaststreams |
| 22:47.08 | dlynes | browser: so where in lies the problem, then? |
| 22:47.18 | browser | dlynes: madplay on RHEL 5 x86_64 didn't convince me. It requires some libraries only commonly available for i386 |
| 22:47.20 | dlynes | browser: seems like you got the same location I did: http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 22:47.38 | dlynes | browser: ah |
| 22:48.05 | dlynes | browser: you never mentioned that you were trying to get all of this working on RHEL 5 x86_64 |
| 22:48.56 | browser | dlynes: my bad. I am following the track with mpg123, which is back with some development activity. |
| 22:50.21 | browser | Let me ask the question differently then. Has anyone used mpg123 in a recent version (not the 0.59r widely purported) ? |
| 22:51.00 | browser | http://www.mpg123.de/asterisk.shtml |
| 22:51.07 | *** part/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 22:53.08 | browser | question more generic: Does * only work if mpg123 was present during the compilation or has someone used it in a moh custom context and can give an idea of syntax. |
| 22:54.18 | *** join/#asterisk qdk (n=qdk@87.48.132.114) |
| 23:01.14 | *** join/#asterisk Zyna (n=Zyna@p54BCDA7E.dip.t-dialin.net) |
| 23:01.23 | Zyna | anyone remember me? |
| 23:01.32 | Zyna | just wanted to announce, that I have passed my finals |
| 23:01.39 | Zyna | thx 4 all your help folks |
| 23:02.10 | Zyna | specially to [TK]D-Fender |
| 23:04.37 | *** part/#asterisk Zyna (n=Zyna@p54BCDA7E.dip.t-dialin.net) |
| 23:06.50 | bkruse | jbot: [TK]D-Fender++ |
| 23:15.37 | angryuser | finals of what ? |
| 23:17.19 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.58.30) |
| 23:19.48 | mwalling | ~book |
| 23:19.49 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 23:23.45 | *** join/#asterisk eXistenZ (n=HOME@unaffiliated/existenz) |
| 23:23.52 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 23:23.57 | eXistenZ | [TK]D-Fender, still there? :) |
| 23:26.35 | *** join/#asterisk mihinomenest (n=argh@24-180-124-128.dhcp.aldl.mi.charter.com) |
| 23:27.24 | jaytee | Pregnant man gives birth to baby girl!!! http://www.katu.com/news/22871014.html?video=YHI&t=a |
| 23:27.44 | jaytee | just when you thought you'd seen and heard it all |
| 23:29.33 | eXistenZ | I was trying to use PrivacyManager to block anonymous calls, but it simply allows both anonymous and non-anonymous : here is the log with non-anonynomus - http://pastebin.com/md359d63 and this one for anonymous: http://pastebin.com/m15067c5 |
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