| 00:00.22 | Sponge_bob | how do you fix echo on the far side? |
| 00:02.56 | TripleFFFF | use the force |
| 00:03.47 | Sponge_bob | ? |
| 00:09.51 | rob0 | phigwork: I'm getting those too. Time to turn down the verbosity. :) |
| 00:11.53 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 00:18.18 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 00:19.13 | *** join/#asterisk ivanfm (n=ivanfm@201.52.162.52) |
| 00:35.29 | dlynes_home | CunningPike: btw...since we switched to sangoma, we've got absolutely zero problems now |
| 00:35.34 | *** join/#asterisk knarfly (n=billtill@c-69-180-98-189.hsd1.fl.comcast.net) |
| 00:39.19 | dlynes_home | phigwork: i've gotten that on startup before...it's usually because the main engine hasn't come up yet |
| 00:39.35 | knarfly | Can anyone help me with X-Lite setup? |
| 00:40.10 | knarfly | dlynes_home: Howdy! |
| 00:40.32 | dlynes_home | hey |
| 00:41.25 | knarfly | dlynes_home: Tried to setup an X-Lite at the office but it won't register with my * at home. Had any experience with this. It runs SIP. |
| 00:42.25 | dlynes_home | knarfly: nope...never used x-lite |
| 00:43.36 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:43.55 | knarfly | dlynes_home: The tech at Connect-Ten says is has to be my server setup. All other calls seem to work okay. In and out from FWD and outbound using splitinfinity to the PSTN. |
| 00:55.59 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 00:58.01 | dlynes_home | Yeah...I have no idea....like I said...I've never used x-ten |
| 00:58.07 | dlynes_home | erm x-lite |
| 00:58.41 | dlynes_home | but the first place i would check is your asterisk logs for an invalid registration |
| 00:58.49 | dlynes_home | your username or password probably isn't correct |
| 00:59.08 | Aurs | does anyone here have in-depth knowledge of res_odbc? :) |
| 00:59.19 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 00:59.19 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:00.03 | Aurs | is it normal that asterisk stops talking to the db when i restart the db? (postgres) |
| 01:00.27 | pdthome | knarfly: x-lite has problems if you have multiple interfaces with IPs bound when it tries to figure out what IP it's "listening" on |
| 01:00.30 | Aurs | if i do a "odbc show" in CLI it starts working again |
| 01:00.45 | pdthome | knarfly: also check the settings for stun and try it with/without and see if you get any different results |
| 01:01.18 | *** join/#asterisk mbit (n=nothing9@218-214-57-65.people.net.au) |
| 01:01.48 | mbit | hey guys when connecting a e1 card to a existing PBX is the zaptel configuration much different to plugging it into the PSTN? |
| 01:02.12 | pdthome | dlynes_home: do you do faxing with the sangoma analog, I haven't had much luck with that so far |
| 01:02.36 | pdthome | dlynes_home: it seems to be pretty random whether it works or not |
| 01:02.42 | dlynes_home | haven't tried yet |
| 01:02.54 | dlynes_home | I'll probably get a chance to try in the next couple of weeks |
| 01:03.27 | dlynes_home | I've got a card with an fxo and and fxs module on it |
| 01:03.31 | dlynes_home | so i'll be trying on it |
| 01:03.45 | pdthome | this is my first analog faxing experience. I have been doing everything via PRI |
| 01:04.50 | dlynes_home | Well, let's take a look at your zapata.conf file |
| 01:04.54 | dlynes_home | it might reveal something |
| 01:05.16 | dlynes_home | also, do you have a hardware echo canceller on the sangoma card? |
| 01:05.33 | dlynes_home | are you trying to faxing over voip? or just through an fxo port bridged to an fxs port? |
| 01:19.34 | knarfly | pdthome: Thanks. I have tried it with and without stun but no go. I only have one interface in the * box. Any other advice? |
| 01:21.20 | knarfly | pdthome: I'm worried that I don't have the port forwarding set right. I'm forwarding 5060, 5082, 9999-20001, 2727 and 4956. I found this info somewhere but can't find it again. |
| 01:22.14 | pdthome | dlynes_home: sorry I wondered off to answer the phone |
| 01:22.27 | dlynes_home | doesn't matter |
| 01:22.29 | pdthome | dlynes_home: doing rxfax |
| 01:22.40 | pdthome | leme figure out which pbx that was |
| 01:22.43 | dlynes_home | pdthome: ah...that's highly unreliable under any circumstances |
| 01:22.54 | knarfly | pdthome: And do I forward upd or tcp? I originally had it as tcp but then when I began testing tonight I figured it should be udp. I admit I don't know what I''m doing, just reading and following docs. |
| 01:22.55 | pdthome | i have very good luck with PRI |
| 01:22.56 | dlynes_home | pdthome: try iaxymodem and hylafax instead |
| 01:25.09 | bkw_ | the guy that wrote iaxmodem is coming to cluecon to speak |
| 01:25.20 | pdthome | knarfly: should be udp |
| 01:25.21 | bkw_ | Lee Howard aka redder86 |
| 01:26.16 | knarfly | pdthome: I switched it to udp. Still X-Lite remote won't register. X-Lite on the LAN will register just fine. |
| 01:27.31 | pdthome | dlynes_home: one thing I do currently with rxfax is context based faxing aka 5551234 -> bob@domain.com 5551111 -> suzy@domain.com, is that possible with iaxmodem + hylafax? |
| 01:29.22 | knarfly | dlynes_home: may I inquire about iaxmodem? I use hylafax religiously but have been afraid to link it up with *. The Hylafax mailing list mod said he doesn't recommend it....! |
| 01:30.24 | dlynes_home | bob@domain.com? suzy@domain.com? huh? |
| 01:30.33 | dlynes_home | what do email addresses have to do with faxing? |
| 01:30.48 | pdthome | fax -> tiff -> pdf |
| 01:30.51 | pdthome | email |
| 01:31.13 | dlynes_home | that's not faxing then |
| 01:31.17 | dlynes_home | that's only receiving faxes |
| 01:31.25 | dlynes_home | yes, you can do that with hylafax, too |
| 01:31.46 | dlynes_home | out of everyone i've talked to on here |
| 01:32.03 | dlynes_home | I've heard a lot more people reporting success with iaxmodem and hylafax than rxfax/txfax |
| 01:32.13 | pdthome | cool, hell I'll try anything once |
| 01:32.15 | dlynes_home | rxfax/txfax has always been problematic at best |
| 01:32.27 | pdthome | it isn't working right now so it couldn't get worse :) |
| 01:32.43 | dlynes_home | pdthome: yeah...what i was going to be trying with sangoma |
| 01:32.49 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 01:32.52 | dlynes_home | pdthome: was fxo/fxs bridging for faxing |
| 01:32.54 | dlynes_home | not email |
| 01:33.04 | pdthome | ah |
| 01:33.08 | dlynes_home | email is something i plan to do on the pri, but not on analog |
| 01:33.24 | pdthome | have a couple of customers that are really small that want it |
| 01:33.29 | pdthome | trying to find a way to make it happen for them |
| 01:33.45 | pdthome | right now they just have one of the pbx lines tapped by the fax machine (yuck) |
| 01:33.52 | *** join/#asterisk kcio (n=kcio@c92509a4.rjo.virtua.com.br) |
| 01:33.59 | kcio | hello |
| 01:34.06 | pdthome | yo |
| 01:34.25 | kcio | is there a way I can make calls from my cell phone using a bt adapter trough the asterisk?I want to make calls, not receive |
| 01:35.08 | pdthome | some phones can be used as microphone, speaker combo's for a PC that is probably the closest you will get |
| 01:35.24 | pdthome | a soft phone using the cell phone as a mic/speaker via bluetooth |
| 01:35.55 | kcio | pdthome, u sure about that?Ive heard that its possible |
| 01:36.28 | pdthome | i haven't seen anything, but I don't know everything :) |
| 01:37.45 | kcio | :) |
| 01:38.28 | *** join/#asterisk XARiUS (n=bdarcy@66-146-191-242.skyriver.net) |
| 01:38.31 | pdthome | it boggles my mind that iaxmodem can be better than rxfax you would think just the opposite... rxfax directly interfaces spandsp with the pbx, iaxmodem is a "real" voip connection interfaced with spandsp |
| 01:38.43 | pdthome | the rxfax code must completely suck |
| 01:38.55 | pdthome | (no offense if the author is in here :) |
| 01:39.06 | *** part/#asterisk knarfly (n=billtill@c-69-180-98-189.hsd1.fl.comcast.net) |
| 01:39.23 | CvR | Gnnn, HEAD does not compile with libpri support, because HAVE_LIBPRI is not defined anymore by AST_EXT_LIB_SETUP. Only HAVE_PRI is defined. |
| 01:39.26 | XARiUS | hey guys, anyone have the config files (mass deployment) for the IP501's? I just ordered 5 and they came with the latest firmware, but Polycom's site only has the "old" configs for donwload :( |
| 01:40.09 | pdthome | i do like this implementation better, no patching asterisk to make it work |
| 01:40.28 | XARiUS | maybe the old work just fine, they're for 1.5.3, phones came with 1.6.3.. |
| 01:40.52 | pdthome | http://jackhammer.org/spip_ssip_sip_1_6_6_revB_sig.zip |
| 01:41.03 | XARiUS | thank you MUCHLY pdt :) |
| 01:41.08 | pdthome | np |
| 01:42.17 | dlynes_home | pdthome: is that something like tappin' dat ass? |
| 01:42.42 | pdthome | dlynes_home: eh? |
| 01:42.46 | *** part/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 01:42.48 | dlynes_home | pdthome: the author is 'coppice' |
| 01:43.00 | dlynes_home | pdthome right now they just have one of the pbx lines tapped by the fax machine (yuck) |
| 01:43.02 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 01:43.03 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 01:43.11 | pdthome | O, man nice lag on that one ;) |
| 01:43.30 | pdthome | it's just as dirty as tappin' dat ass |
| 01:44.00 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 01:44.44 | pdthome | you have to be careful what is in the dialplan, asterisk is faster at answering than the fax machine |
| 01:45.01 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 01:45.02 | *** part/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 01:45.04 | pdthome | if you jump into ivr, etc.. to fast it freaks out |
| 01:45.06 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
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| 01:45.52 | pdthome | dlynes_home: thanks for the iaxmodem idea, i'll have to give that a shot tomorrow. I'll let you know how it turns out |
| 01:46.16 | pdthome | time to go hunt down some food |
| 01:47.13 | dlynes_home | ok |
| 01:48.22 | Agrajag- | gday. using the Echo command i hear myself about 0.5 - 0.75 seconds after i talk. this is on a local lan, ping time to the box running asterisk is about 0.140. using iaxcomm and xten softphones, tried gsm and ulaw. i also have an spa-3000 and the phone connected to the fxs on that doesn't have the problem - echo is pretty much real time |
| 01:48.22 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 01:49.14 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 01:50.25 | rene- | hi, i am looking to the SIP Authorization Header part of a Register message, and there is this part that reads: .......algorithm=MD5,response="...." |
| 01:50.26 | rene- | why MD5? does that means i need to define a md5secret in asterisk? asterisk is refusing to register the peers (realtime mysql) |
| 01:50.27 | FuriousGeorge | hey all. im trying to get this tdm400p on its own IRQ, but unfortunately linux seem to ignore the settings in my bios. slot 4 is the only one that doesnt have to share an IRQ, and slot 4 is where the card is at, but its being put on 3 with the USB and something else |
| 01:50.50 | rene- | FuriousGeorge, why dont you disable USB |
| 01:51.05 | FuriousGeorge | rene-: its my home computer |
| 01:51.10 | FuriousGeorge | i should have mentioned that |
| 01:51.48 | FuriousGeorge | ive tried passing acpi=noirq to the bios |
| 01:52.02 | FuriousGeorge | i mean: to grub |
| 01:52.17 | rene- | broken bios? have you tried juggling the card to the other sockets? |
| 01:52.25 | rene- | err, slots |
| 01:53.12 | FuriousGeorge | my bios isnt broken, linux is just ignoring the settings. yeah, if i move the slot the irq changes but its just being shared by something else |
| 01:53.15 | FuriousGeorge | with something else |
| 01:53.24 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
| 01:54.18 | rene- | what about disabling serial/parallel interfaces onboard video so you can free up some ircs |
| 01:54.24 | rene- | irqs |
| 01:54.31 | FuriousGeorge | thats the thing, i need everything else |
| 01:54.37 | rene- | everything? |
| 01:54.39 | FuriousGeorge | at least all the things that are taking an irq |
| 01:54.40 | FuriousGeorge | yeah |
| 01:54.43 | FuriousGeorge | i already disabled serial |
| 01:54.48 | FuriousGeorge | i need parallel |
| 01:54.51 | FuriousGeorge | need usb |
| 01:54.53 | rene- | you should have gotten a sangoma board |
| 01:54.54 | FuriousGeorge | need sound, video |
| 01:55.19 | FuriousGeorge | rene-: maybe, but it works fine. it just annoys me that i got all these free irqs and no one knows how to force linux to use them |
| 01:55.39 | rene- | not even people in #kernel? |
| 01:56.41 | FuriousGeorge | good idea, ill try in there |
| 01:59.11 | dlynes_home | FuriousGeorge: it's not a matter of whether linux can be forced to use them |
| 01:59.18 | dlynes_home | FuriousGeorge: more a matter of forcing your bios to use them |
| 01:59.24 | FuriousGeorge | its if i can bend the spoon? |
| 01:59.38 | FuriousGeorge | dlynes_home: i have the bios set to use irq7 for pci slot 4 |
| 01:59.41 | FuriousGeorge | where my tdmlives |
| 01:59.43 | dlynes_home | FuriousGeorge: you'd run into the same problem with windows |
| 01:59.58 | dlynes_home | and irq7 is shared with your printer port |
| 02:00.12 | dlynes_home | are you using acpi or apic? |
| 02:01.12 | FuriousGeorge | im pretty sure its acpi |
| 02:01.28 | FuriousGeorge | the one under "Power Management" in kernel's menuconfig |
| 02:01.34 | dlynes_home | that's acpi |
| 02:02.10 | dlynes_home | apic allows you to have 32-bit(?) irqs |
| 02:02.42 | FuriousGeorge | dlynes_home: speaking of printer. i got an LCD device that came with my case that interfaces via parallel port. i dont have a /dev/lp0, and i dont see lp0 taking an irq, i do have Parallel support in the kernel, and the LCD device works fine despite all this. wierd huh |
| 02:02.49 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 02:03.11 | dlynes_home | you probably have a /dev/lp |
| 02:04.41 | rene- | dlynes_home: if a sip device tries to register to asterisk sending lgorithm=MD5 |
| 02:04.44 | rene- | sorry |
| 02:05.02 | rene- | sending algorithm=MD5 in its authorization header |
| 02:05.18 | rene- | do i need to define a md5secret for that device in asterisk> |
| 02:05.19 | rene- | ? |
| 02:06.10 | FuriousGeorge | dlynes_home: no dev lp either |
| 02:06.11 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 02:06.23 | FuriousGeorge | the only thing in dev that starts with l are loop* |
| 02:06.26 | rene- | or do every sip authentication uses MD5 and md5secret is just a way of not storing plain passwords in asterisk? |
| 02:06.30 | FuriousGeorge | no /dev/paraport either |
| 02:06.49 | Telamon | How much bandwidth does a ULAW call *actually* take, when you include overhead and fragmentation and such? |
| 02:07.03 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 02:07.16 | dlynes_home | FuriousGeorge: no idea then...could've sworn it was lp, or lp0 or something...i havne't used a parallel printer in i don't know how long |
| 02:07.39 | dlynes_home | FuriousGeorge: you sure it's a parallel cable and not a 25-pin serial cable, eh? |
| 02:07.41 | Telamon | FuriousGeorge: You need to load parport, parport_pc, and lp if you are using Linux. |
| 02:07.50 | Telamon | load = modprobe |
| 02:08.12 | FuriousGeorge | dlynes_home: well whatever it works. 25 pin serial? no its definately a parallel this is a new case |
| 02:08.24 | dlynes_home | a new case? |
| 02:08.32 | rene- | Telamon: IIRC it was in the vecinity of 100Ks |
| 02:08.33 | dlynes_home | I didn't even think new cases still had parallel ports |
| 02:08.47 | Telamon | rene-: K as in kytes ort kbits? |
| 02:08.57 | FuriousGeorge | Telamon: i just use paraport for the lcd device which is working so ill leave it alone. the last thing i need is another irq taken. actually i was just talking about how to force linux to use a certain irq for irq7 |
| 02:09.00 | rene- | kbits |
| 02:09.10 | FuriousGeorge | someone said with windows i would have the same problem. maybe but its easy enough to change |
| 02:10.17 | FuriousGeorge | someone else said i can just put irq=7 in my modules.conf file, but i use modules.autoload.d/lernel2.6 to load my module. i guess i can try it |
| 02:14.07 | P-NuT | Hi all, I have a budgetone 101. I've just upgraded the firmware and now it has no dialtone and spits out 403 on the screen whenever you try to do something. Has anyone else had this problem? |
| 02:17.35 | FuriousGeorge | no luck so far |
| 02:19.01 | FuriousGeorge | i fournd somewhere in modules.conf where something is aliased to wctdm. i didnt put it there, so i guess my distro generated that part of that file for me. i tried putting an "option wctdm irq=7" in there and no luck |
| 02:19.25 | CvR | Does anybody have an * connected via a PRI to a DMS100? I just can't get it to work and the people from APUA can't point me in the right direction.. The dms100 just presents me with an immediate RELEASE when * replies with CALL PROCEEDING to the dms100's SETUP. |
| 02:19.28 | dlynes_home | no...your zaptel install did that, FuriousGeorge |
| 02:19.46 | FuriousGeorge | dlynes_home: well that was helpful |
| 02:20.09 | rene- | FuriousGeorge, will you be able to set the IRQ via the zaptel driver? |
| 02:20.25 | FuriousGeorge | rene-: the zaptel mopdule isnt using an irq |
| 02:20.28 | FuriousGeorge | just wctdm |
| 02:20.40 | FuriousGeorge | which is the one im trying to point to my free irqs with no luck |
| 02:20.46 | dlynes_home | rene-: you can't change the irq with a driver parameter; only force it if it's not autodetected |
| 02:21.05 | rene- | dlynes_home: ahh |
| 02:21.06 | FuriousGeorge | dlynes_home: how can i force it if it is autodetected |
| 02:22.22 | dlynes_home | FuriousGeorge: if the zaptel driver supports it, you can use the method you used; but that only forces it to use the irq it's already using; it doesn't tell it to use a different one |
| 02:22.49 | dlynes_home | FuriousGeorge: in order to get it to use a different one, you have to tell the chipset to put it on a different irq, and you do that through your bios setup |
| 02:23.09 | dlynes_home | FuriousGeorge: or, if your motherboard/chipset support it, use apic |
| 02:23.17 | rene- | cool |
| 02:23.23 | dlynes_home | FuriousGeorge: that'll guarantee each piece of hardware has a unique irq |
| 02:23.44 | dlynes_home | FuriousGeorge: because it'll use 64-bit irqs then, instead of 16-bit irqs |
| 02:23.53 | FuriousGeorge | dlynes_home: thats the thing i _already_ set it in my bios to be on a given irq |
| 02:24.00 | FuriousGeorge | its just being ignored |
| 02:24.05 | dlynes_home | FuriousGeorge: use apic then |
| 02:24.11 | dlynes_home | FuriousGeorge: and that's apic, not acpi |
| 02:25.12 | dlynes_home | beautiful |
| 02:25.23 | dlynes_home | got my wireless adapter up and running under linux on my laptop now :) |
| 02:26.37 | FuriousGeorge | dlynes_home: did you mean use apic instead of acpi or with acpi? |
| 02:27.04 | FuriousGeorge | my bios seems to support it, and i have a bunch of apic related stuff enabled in the kernel, but im not sure which one would make a difference, if it would |
| 02:27.06 | *** part/#asterisk mbit (n=nothing9@218-214-57-65.people.net.au) |
| 02:27.26 | dlynes_home | FuriousGeorge: you can use it with acpi, or without it |
| 02:27.57 | dlynes_home | FuriousGeorge: apic options will not configurable unless you have smp support enabled in the kernel |
| 02:28.27 | dlynes_home | FuriousGeorge: you'll know if your system supports it or not |
| 02:28.31 | FuriousGeorge | dlynes_home: i think its the other way. if i do have smp no apic |
| 02:28.49 | dlynes_home | FuriousGeorge: nah...all smp systems have apic, and some non-smp systems also have apic |
| 02:29.19 | FuriousGeorge | Symbol: X86_UP_APIC [=y] Prompt: Local APIC support on uniprocessors Defined at arch/i386/Kconfig:231 Depends on: !SMP && !X86_VISWS && !X86_VOYAGER |
| 02:29.33 | FuriousGeorge | !SMP means not smp doint it |
| 02:29.35 | FuriousGeorge | dont it |
| 02:29.38 | dlynes_home | FuriousGeorge: anyways...if you do cat /proc/interrupts after you've configured apic, you'll see some stuff in there |
| 02:29.45 | dlynes_home | There's two apic options |
| 02:29.52 | dlynes_home | one for uniprocessor, and one for smp |
| 02:30.10 | dlynes_home | there's apic and local apic |
| 02:30.17 | dlynes_home | apic is smp; local apic is uniprocessor |
| 02:30.27 | FuriousGeorge | dlynes_home: i must have it working then, because i already see stuff when i cat /proc/interrupts thats how i know its sharing an irq |
| 02:31.09 | dlynes_home | FuriousGeorge: you see a bunch of io-apic stuff in cat /proc/interrupts? |
| 02:31.23 | FuriousGeorge | dlynes_home: well i only have one processor so i guess apic is out of the question. |
| 02:31.43 | FuriousGeorge | dlynes_home: i didnt have Local APIC IO built into kernel. was thinking of trying that |
| 02:31.48 | dlynes_home | you don't understand |
| 02:31.56 | dlynes_home | you can have apic on both smp and uniprocessor |
| 02:32.00 | FuriousGeorge | IO-APIC support on uniprocessors |
| 02:32.03 | dlynes_home | it's very rare to have it on uniprocessor |
| 02:32.07 | dlynes_home | but it does exist |
| 02:32.17 | FuriousGeorge | but its not the same as local apic |
| 02:32.43 | dlynes_home | anyways...just try whatever combinations you can try |
| 02:32.44 | FuriousGeorge | ? |
| 02:32.48 | dlynes_home | and hope for the best |
| 02:33.04 | dlynes_home | i've only ever had one uniprocessor machine that had apic support |
| 02:36.33 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 02:36.39 | FuriousGeorge | im finding links that seem to suggest my mb supports it |
| 02:36.56 | FuriousGeorge | this IO APIC support might be the thing i need |
| 02:37.13 | dlynes_home | FuriousGeorge: well, apic might fix your problem, and it might now |
| 02:37.16 | dlynes_home | s/now/not/ |
| 02:37.29 | dlynes_home | FuriousGeorge: you're using a digium card, right? |
| 02:37.43 | FuriousGeorge | yeah |
| 02:37.55 | dlynes_home | Yeah...it's got about 95% chance of fixing your problem |
| 02:37.58 | dlynes_home | but at this point |
| 02:38.02 | dlynes_home | You don't have much of an option |
| 02:38.32 | dlynes_home | I'm guessing you don't have a lot of pci slots in that machine either, right? |
| 02:38.34 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 02:38.39 | FuriousGeorge | dlynes_home: it just irks me that i have so many free irqs and no real way to force linux to use one |
| 02:38.50 | ManxPower | It's call ACPI |
| 02:38.51 | FuriousGeorge | dlynes_home: i got 4, and one of those offset other pci slots |
| 02:39.00 | FuriousGeorge | ManxPower: both exist |
| 02:39.19 | ManxPower | setpci doesn't help you? |
| 02:39.30 | FuriousGeorge | setpci? |
| 02:40.26 | ManxPower | setpci lets yu do cool things like change the PCI bus priority of various devices. rumor has it that you can even change the IRQ of a device, I've never done it |
| 02:40.55 | dlynes_home | ManxPower: do you need acpi enabled to make use of setpci? |
| 02:41.07 | ManxPower | dlynes_home, I doubt it. |
| 02:41.10 | FuriousGeorge | dlynes_home: i do have it enabled |
| 02:41.11 | dlynes_home | ah |
| 02:41.20 | dlynes_home | well, apic should solve his problem, too |
| 02:41.30 | dlynes_home | with less screwing around |
| 02:41.36 | ManxPower | yes, it should. |
| 02:42.45 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 02:42.55 | FuriousGeorge | ill see if it works |
| 02:43.01 | FuriousGeorge | be back in a minute |
| 02:43.05 | *** part/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 02:43.23 | rene- | If a device tries to authenticate using MD5 (sip) to asterisk does it matters weather asterisk sip entry for that user has a md5secret or a secret? |
| 02:43.23 | *** join/#asterisk LoneShadow (n=a@c-67-188-235-220.hsd1.ca.comcast.net) |
| 02:44.52 | ManxPower | I'll bet it does |
| 02:45.33 | rene- | Manx: you bet it does? |
| 02:46.07 | rene- | see i havent done anything special to this PAP2 unit, it is new so that s probably the reason is being naughty to me |
| 02:46.36 | rene- | but i do see .....algorithm=MD5,response="...." in the authentication header for the Register request |
| 02:46.46 | rene- | what do you say? |
| 02:47.02 | rene- | i could add the column |
| 02:47.08 | rene- | to sip realtime |
| 02:48.17 | rene- | if i do not use realtime authentication works |
| 02:49.53 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 02:50.18 | FuriousGeorge | dlynes_home: the bad news is that it still ignores the settings in my BIOS, the good news is that it has its own IRQ now |
| 02:50.25 | rene- | cool |
| 02:50.29 | FuriousGeorge | i kinda wish it wasnt the last irq though :) |
| 02:50.30 | rene- | what did ya do? |
| 02:50.51 | FuriousGeorge | doesnt that mean its the hardware with the lowest proiority in my system |
| 02:51.08 | FuriousGeorge | rene-: i enabled apic |
| 02:51.15 | rene- | ahh |
| 02:51.15 | dlynes_home | FuriousGeorge: when it's using apic, the irq settings you have in your bios are irrelevant |
| 02:51.16 | rene- | nice |
| 02:51.27 | dlynes_home | FuriousGeorge: the irq settings you have in your bios are for 16-bit irqs |
| 02:51.36 | FuriousGeorge | rene-: is it? |
| 02:51.43 | FuriousGeorge | dlynes_home: is it bad that its the last irq? |
| 02:52.11 | FuriousGeorge | it got 21 |
| 02:52.14 | dlynes_home | yes, but is it bad that it's working now? |
| 02:52.27 | FuriousGeorge | dlynes_home: it was working before too |
| 02:52.36 | dlynes_home | then why all the concern? |
| 02:52.40 | FuriousGeorge | dlynes_home: whats better sharing irq3 or owning irq 21? |
| 02:52.48 | ManxPower | IRQs in the BIOS and in lspci are BEFORE ACPI is enabled |
| 02:53.01 | ManxPower | owning 21 |
| 02:53.04 | FuriousGeorge | dlynes_home: im just trying to take control of my interupts |
| 02:53.05 | rene- | Furious did your card worked when it was sharing the irq? |
| 02:53.16 | FuriousGeorge | ManxPower: i mean that figuratively, its alone on 21 |
| 02:53.33 | dlynes_home | FuriousGeorge: sharing irq 3 if it's a sangoma; owning 21 if it's a digium |
| 02:53.36 | FuriousGeorge | rene-: yeah it did, this is just a learing experience and a chance to get pissed of at my system |
| 02:53.45 | rene- | heh |
| 02:53.48 | FuriousGeorge | *pissed off |
| 02:54.36 | FuriousGeorge | i guess i can try once without apic or acpi and just setting it in the bios |
| 02:54.43 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 03:01.07 | FuriousGeorge | my buddy just said something about a vector table and interrupts and it not mattering if its on 21 or 2 but then again he's full of sh!t sometimes |
| 03:02.56 | FuriousGeorge | here we go |
| 03:03.00 | *** part/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 03:05.47 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 03:07.15 | rene- | my problem was due not activating rt caching, it resolved my issue |
| 03:08.38 | dlynes_home | woah...cool |
| 03:08.43 | dlynes_home | new laptop is 6400 bogomips :) |
| 03:11.10 | rene- | is it a dualcore? |
| 03:11.20 | rene- | core duo |
| 03:11.40 | Qwell | dlynes_home: all 3.2ghz x86 cpus are 6400 bogomips :p |
| 03:11.54 | LoneShadow | what does this error mean "dialparties.agi: Extension 100 cf is disabled" ? |
| 03:11.59 | dlynes_home | yeah...core duo... |
| 03:12.15 | dlynes_home | Qwell: it's a 1.67Ghz |
| 03:12.31 | dlynes_home | sempron duo core |
| 03:12.33 | dlynes_home | erm |
| 03:12.35 | dlynes_home | centrino |
| 03:12.37 | Qwell | eh? |
| 03:12.38 | Qwell | oh, heh |
| 03:12.42 | Qwell | I was gonna say... |
| 03:12.45 | Qwell | you have an Intel AMD? |
| 03:12.45 | dlynes_home | hehe |
| 03:13.11 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 03:13.20 | dlynes_home | the judge tossed out 200 bogus claims of sco against ibm |
| 03:13.21 | dlynes_home | heh |
| 03:13.38 | rob0 | 3596.75 / 3591.27 on my Dual Core AMD Opteron(tm) Processor 165 (server) |
| 03:13.49 | *** part/#asterisk CvR (n=CvR@cw.global-player.com) |
| 03:14.11 | rob0 | I think that's a 1.8G CPU. |
| 03:15.16 | [TK]D-Fender | LoneShadow : Means stop using AMP/FreePBX |
| 03:15.25 | file | [TK]D-Fender: cute answer |
| 03:15.53 | LoneShadow | found the problem, my db is messed up =/ |
| 03:16.29 | rene- | bye ppl |
| 03:16.44 | LoneShadow | [TK]D-Fender: I guess I will have to deal with debugging things myself if I use Freepbx :) |
| 03:16.58 | *** part/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 03:18.18 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 03:18.58 | *** part/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 03:20.43 | [TK]D-Fender | LoneShadow :Odds are if you're using FreePBX then you shouldn't be using it if you're qualified to debug it. Funny logic to it... |
| 03:25.40 | LoneShadow | well I am using it for personal use, just messing around with it to see whats so great about it :P |
| 03:26.18 | LoneShadow | has anyone use video-sip client in here ? |
| 03:26.22 | LoneShadow | used* |
| 03:31.27 | LoneShadow | [TK]D-Fender: if I have a sip connection established between 2 sip clients, can I use another sip client to join the conversation ? would that be a conference call ? |
| 03:32.06 | LoneShadow | like if I have 2 sip phones in my house, both will ring. But if I pick one, can I also pick the other and join the same conversation ? |
| 03:32.14 | X-Rob_ | No. |
| 03:32.53 | LoneShadow | I am trying to make the 2nd sip client a video soft phone :) |
| 03:33.14 | X-Rob_ | [TK]D-Fender, it's an issue. We put as much debugging in there as we can, but peopel use it because they don't have the skills to drive * natively. Any suggestions? 8) |
| 03:34.00 | [TK]D-Fender | LoneShadow : No. You'd have to place a call to the other phone and have that phone do the 3-way for you |
| 03:35.29 | LoneShadow | I have 2 msntv boxes running linux, and 2 spa3k phones, trying to see how to integrate them. So that I can video conferencing unit on my tv for $30-50 on each side :) |
| 03:35.34 | *** part/#asterisk file (n=file@216.237.114.82) |
| 03:35.35 | [TK]D-Fender | X-Rob_ : Suggestings for what? |
| 03:35.39 | *** join/#asterisk file (n=file@216.237.114.82) |
| 03:35.46 | X-Rob_ | <[TK]D-Fender> LoneShadow :Odds are if you're using FreePBX then you shouldn't be using it if you're qualified to debug it. Funny logic to it... |
| 03:36.25 | [TK]D-Fender | X-Rob_ : You mean suggestions on how to improve/repair it? |
| 03:36.56 | X-Rob_ | How to make it easier for people to debug |
| 03:37.06 | X-Rob_ | but if you don't use it, then not much point asking you, now I think about it. |
| 03:38.09 | [TK]D-Fender | X-Rob_ : Who says circular logic is confusing... I find it incredibly ENLIGHTENING ;) |
| 03:38.35 | X-Rob_ | heh |
| 03:44.28 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001217014b92.vc.shawcable.net) |
| 03:48.31 | *** part/#asterisk Mattwj2005 (n=Matt@user-12l3n74.cable.mindspring.com) |
| 03:53.38 | *** join/#asterisk Ritalin- (i=dave@c-68-47-199-178.hsd1.tn.comcast.net) |
| 03:53.47 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 03:57.11 | *** join/#asterisk zepmantra (i=dsadsa@203.215.100.96) |
| 03:58.06 | zepmantra | hello just wanna ask, i have here openvox a400p22, my question is how will i know where to place fxo line or fxs? its not indicated |
| 03:59.50 | Ritalin- | is that like the TDM400? |
| 04:00.42 | zepmantra | yes.. |
| 04:01.10 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 04:01.35 | Trazz | what does everyone recommend for FXS cards or products to connect to * / |
| 04:01.36 | zepmantra | or i should be the one to set what channels on zapata.conf? |
| 04:01.58 | Ritalin- | zep: looking at the card with the PCI connector at the bottom what colors are the daughter cards? green/green/red/red ? |
| 04:02.25 | Ritalin- | do you mean ... physically hooking up the card or configuring asterisk? |
| 04:02.46 | zepmantra | Ritalin- its red red green green |
| 04:02.47 | *** join/#asterisk docelmo (n=docelmo@55-65.126-70.tampabay.res.rr.com) |
| 04:03.22 | Ritalin- | http://www.digium.com/images/products/hardware/tdm400p.jpg |
| 04:03.24 | Ritalin- | opposite of that ? |
| 04:03.34 | zepmantra | yes opposite... |
| 04:03.44 | docelmo | opposite of what? |
| 04:04.03 | docelmo | Maybe a Sipura 3000 or Grandstream 486 or something.. |
| 04:04.05 | zepmantra | of that digium card .kpeg |
| 04:04.06 | Sedorox | its a rj45 plug with a female PCI adapter? |
| 04:04.27 | docelmo | I thought the TDM400P was the FXO/FXS card |
| 04:04.40 | Sedorox | it can be both |
| 04:04.43 | Ritalin- | zep: the top port is the left module. so in the picture i just showed you it goes FXS/FXS/FXO/FXO (#1/#2/#3/#4) |
| 04:04.46 | Sedorox | iirc |
| 04:04.54 | docelmo | ANYWHO.. |
| 04:05.01 | docelmo | ~seen damin |
| 04:05.19 | jbot | damin is currently on #asterisk, last said: 'Dr-Linux: WTF?'. |
| 04:05.20 | docelmo | Where is jbot? |
| 04:05.20 | zepmantra | ic ... so my card is the otherway around ... |
| 04:05.27 | docelmo | haha |
| 04:05.35 | pdthome | <PROTECTED> |
| 04:05.36 | docelmo | DAMIN! QUIT DRINKING |
| 04:05.37 | Ritalin- | zep: yeah you should have (from top to bottom) FXO/FXO/FXS/FXS |
| 04:05.54 | zepmantra | i got it ... thanks Ritalin- :) |
| 04:06.04 | Ritalin- | zep: assuming it's the same as the TDM400 :) |
| 04:06.06 | docelmo | I have a TDM4210 @ home in my computer |
| 04:06.06 | Sedorox | ahahah |
| 04:06.11 | Ritalin- | which it looks like it |
| 04:07.22 | Sedorox | I need a TE110P |
| 04:07.49 | docelmo | hmmm.. check out voipconnection |
| 04:07.55 | docelmo | I usually buy from those guys |
| 04:08.06 | docelmo | I got a TE410P and the thing is rock solid |
| 04:08.38 | Sedorox | I just dont' have >$200 right now :p |
| 04:08.45 | Sedorox | plus I wouldn't need it perm.. but anyway |
| 04:08.47 | Sedorox | time for bed...... |
| 04:15.17 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 04:15.40 | Trazz | whats the command to call an extension from the CLI ? |
| 04:15.59 | CunningPike | Trazz: There isn't one |
| 04:16.09 | Trazz | ouch ok |
| 04:16.10 | dlynes_laptop | CunningPike, yes ther is |
| 04:16.15 | Trazz | i thought there was |
| 04:16.23 | CunningPike | Oh? I stand corrected |
| 04:16.28 | dlynes_laptop | dial |
| 04:16.34 | Trazz | i have done it b4 but dont remember the command |
| 04:16.39 | dlynes_laptop | but you need chan_phone installed, I think |
| 04:16.44 | CunningPike | Ah |
| 04:17.13 | dlynes_laptop | it's really only useful on a desktop machine, not a real server |
| 04:17.19 | CunningPike | dlynes_laptop: And now you have another bloody nick ;) |
| 04:17.26 | dlynes_laptop | haha |
| 04:17.33 | Trazz | i tried the dial command and no luck |
| 04:17.42 | CunningPike | dlynes_laptop: You realize what you're doing to my auto-complete, don't you? |
| 04:17.44 | dlynes_laptop | Trazz, load chan_phone.so |
| 04:17.48 | dlynes_laptop | heh |
| 04:18.13 | dlynes_laptop | Trazz, you might additionally need to load chan_alsa.so |
| 04:18.30 | CunningPike | dlynes_laptop: I'm playing with the AsteriskGuru Queue Statistics atm |
| 04:18.33 | Trazz | unable to load those modules :( |
| 04:18.52 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 04:19.05 | dlynes_laptop | Trazz: i guess you're screwed until you can then |
| 04:19.15 | dlynes_laptop | Trazz: why would you want to dial from the cli, anyways? |
| 04:19.26 | Trazz | to do testing from colo |
| 04:19.39 | dlynes_laptop | CunningPike: you mean for agent queues? |
| 04:19.53 | Trazz | or do test from CLI where i dont have extensions with all the dialplans |
| 04:19.53 | CunningPike | dlynes_laptop: Yes |
| 04:20.07 | dlynes_laptop | Trazz: set up a dial command for when you call up the pri, have it do something special for your did |
| 04:20.21 | Trazz | ok |
| 04:20.23 | dlynes_laptop | ah...yeah...i haven't had a use for agents and queues yet |
| 04:20.33 | CunningPike | Trazz: Or use a softphone like SJPhone that lets you quickly switch profiles |
| 04:20.52 | dlynes_laptop | I just had my first customer that needed call parking |
| 04:21.09 | CunningPike | dlynes_laptop: With Polycoms? Or using the asterisk feature? |
| 04:21.11 | dlynes_laptop | most of our customers' needs are quite basic |
| 04:21.18 | dlynes_laptop | using the asterisk feature |
| 04:21.20 | CunningPike | Green, green, yellow |
| 04:21.24 | dlynes_laptop | i'm using aastra 9133i's |
| 04:21.45 | Trazz | ok |
| 04:27.57 | *** join/#asterisk BugKham (i=BugKham@202.8.86.164) |
| 04:28.57 | BugKham | do I need zaptel or any timing source for moh? |
| 04:32.13 | *** join/#asterisk orlock (n=jwr@202.44.174.4.static.nexnet.net.au) |
| 04:33.10 | orlock | In the sip registratipn session, does anybody know what is used to set the Contact: field? |
| 04:33.57 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 04:36.22 | pdthome | i have a wierd problem with the parking |
| 04:36.34 | pdthome | if the user waits too long to finish the transfer the call gets hung up |
| 04:36.53 | pdthome | they have to finish the transfer right after the extension is announced or it dies |
| 04:37.04 | CunningPike | BugKham: Yes - if you have no hardware for timing, you should use ztdummy |
| 04:38.53 | *** join/#asterisk b00gz1 (n=b00gz@d233-124-245.col.wideopenwest.com) |
| 04:40.08 | *** join/#asterisk extremescholar (n=robinson@c-69-242-176-251.hsd1.mo.comcast.net) |
| 04:40.43 | extremescholar | How do I go about re-opening a bug? |
| 04:43.09 | docelmo | uhh bugs.digium.com |
| 04:45.14 | *** part/#asterisk extremescholar (n=robinson@c-69-242-176-251.hsd1.mo.comcast.net) |
| 04:45.37 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:47.59 | BugKham | CunningPike, thanks |
| 04:48.45 | CunningPike | np |
| 04:51.35 | BugKham | CunningPike, thought it was only used with meetme |
| 04:52.10 | Agrajag- | gday. using the Echo command i hear myself about 0.5 - 0.75 seconds after i talk. this is on a local lan, ping time to the box running asterisk is about 0.140, using iaxcomm and xten softphones, tried both gsm and ulaw. i also have an spa-3000 and the phone connected to the fxs on that doesn't have the problem - echo is pretty much real time |
| 04:52.28 | Agrajag- | why might i get that latency? |
| 04:53.28 | CunningPike | BugKham: MOH and MeetMe both need a timing source |
| 04:55.18 | BugKham | CunningPike, 2.6 kernel seems to provide internal kernel timer |
| 04:55.41 | BugKham | CunningPike, but I just installed ztdummy =) |
| 04:56.15 | CunningPike | BugKham: It does - but it supplies timing to ztdummy. Pre-2.6, you needed to have USB enabled to provide timing to ztdummy |
| 04:56.15 | orlock | heh |
| 04:56.20 | orlock | feline world domination |
| 04:56.31 | orlock | you werent running no-cat on it were you? :) |
| 04:56.42 | orlock | doh |
| 04:57.26 | *** join/#asterisk livinded (n=livinded@pool-68-238-132-220.sea.dsl-w.verizon.net) |
| 04:57.48 | BugKham | CunningPike, ok |
| 04:58.23 | livinded | whats the difference between the svn snapshot and the other svn source? |
| 04:58.44 | livinded | is the snapshot the stable one? |
| 05:01.57 | livinded | anyone are that can answer that for me? |
| 05:02.07 | livinded | around* |
| 05:05.55 | hads|home | A snapshot is typically just a, erm.. snapshot of the SVN tree at a particular point in time. |
| 05:06.27 | livinded | hads: which is a better choice to run? |
| 05:06.50 | hads|home | The release version. 1.2.9.1 I think |
| 05:07.12 | hads|home | Or if you want to test etc. then SVN trunk. |
| 05:07.24 | livinded | hads|home: alright |
| 05:07.36 | hads|home | Or you could run the asterisk 1.2 SVN trunk I guess. |
| 05:08.02 | livinded | i guess i'll just stick with the stable release |
| 05:08.02 | livinded | i don't want things breaking |
| 05:09.50 | *** join/#asterisk droops (n=root@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 05:10.18 | dlynes_laptop | hads: there's also 1.2 branch |
| 05:10.30 | dlynes_laptop | hads: no such thing as 1.2 trunk |
| 05:10.37 | hads|home | Yeah, that's what I meant :/ |
| 05:10.41 | dlynes_laptop | hads: but i think you meant just plain trunk |
| 05:11.05 | hads|home | <hads|home> Or you could run the asterisk 1.2 SVN trunk I guess. |
| 05:11.18 | dlynes_laptop | 1.2 branch is the 1.2.10 prospect |
| 05:11.23 | dlynes_laptop | trunk is the 1.4 prospect |
| 05:11.23 | hads|home | should have said 1.2 *branch* |
| 05:12.01 | hads|home | Indeed. And I lovely prospect it is :) |
| 05:12.13 | dlynes_laptop | livinded: the snapshot is usually a snapshot of trunk at a particular point in time |
| 05:12.43 | livinded | so should i just go with the stable release if i want something thats going to work? |
| 05:12.49 | dlynes_laptop | livinded: so iow, it might not even compile |
| 05:12.56 | dlynes_laptop | livinded: yes |
| 05:13.04 | livinded | alright |
| 05:13.07 | dlynes_laptop | livinded: always go with the latest release tarball |
| 05:13.47 | dlynes_laptop | livinded: only start looking at 1.2 branch if you're finding a bug in the release that you've heard is fixed in 1.2 branch |
| 05:14.02 | livinded | alright |
| 05:14.20 | livinded | ohh this is bad |
| 05:14.25 | dlynes_laptop | livinded: you should never look at trunk for a non-test system |
| 05:14.27 | dlynes_laptop | livinded: ? |
| 05:14.47 | livinded | well i'm installing asterisk for a friend and i'm getting all these errors |
| 05:15.01 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 05:15.02 | dlynes_laptop | put them on pastebin so someone can help you |
| 05:15.05 | livinded | i've had bad luck for everything i've ever used debian for |
| 05:15.06 | dlynes_laptop | ~pb |
| 05:15.07 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 05:15.13 | livinded | slackware never gives me a problem |
| 05:15.17 | dlynes_laptop | :) |
| 05:15.36 | livinded | i'm anciously awaiting slack 11 |
| 05:15.42 | dlynes_laptop | I even run slackware on my laptop |
| 05:15.53 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 05:15.59 | livinded | i had slack on my laptop but i got tired of building packages for everything |
| 05:16.04 | hads|home | Both are nice. BSD init is good though. |
| 05:16.15 | livinded | i have fedora core 5 on here and i'm not terribly impressed |
| 05:16.23 | dlynes_laptop | I only build a package for asterisk |
| 05:16.28 | livinded | i was thinking about giving freebsd a try |
| 05:16.34 | dlynes_laptop | so i don't have to recompile it on every deploymnet |
| 05:16.56 | dlynes_laptop | why do you feel the need to build a whole bunch of packages for slackware? |
| 05:17.12 | livinded | i need to build a new box this summer so i can get asterisk back up, my ox has been down for 4 or 5 months now and i haven't touched it since |
| 05:17.30 | livinded | dlynes_laptop: becuase it makes updating much easier |
| 05:17.37 | livinded | and easier to remove |
| 05:17.47 | dlynes_laptop | and why can't you find those packages already? |
| 05:18.01 | livinded | dlynes_laptop: i can find some |
| 05:18.09 | livinded | but not for everything |
| 05:18.14 | dlynes_laptop | livinded: you know about www.linuxpackages.net, right? |
| 05:18.21 | livinded | ya, thats where i go |
| 05:18.34 | dlynes_laptop | ah |
| 05:18.44 | dlynes_laptop | you're weird though :) |
| 05:18.48 | CunningPike | :P |
| 05:18.56 | livinded | i wish pat would build a amd64 build i don't wanna run a 3rd party build of slack for my desktop |
| 05:19.16 | dlynes_laptop | livinded: i don't think that's going to happen any time soon |
| 05:19.20 | livinded | me either |
| 05:19.34 | dlynes_laptop | livinded: a sunsparc official distro will probably happen sooner |
| 05:19.42 | livinded | i haven't found a linux distro i like enough to use as my desktop besides slackware |
| 05:19.55 | dlynes_laptop | heh |
| 05:19.58 | livinded | hey droops |
| 05:20.00 | droops | whats up all |
| 05:20.04 | droops | hey livinded, whats up |
| 05:20.24 | livinded | trying to stay connected to this wifi |
| 05:20.25 | livinded | i |
| 05:20.49 | livinded | i'm sitting outside in a tshirt and its 53 degrees or so |
| 05:20.51 | CunningPike | Brrrrr |
| 05:20.56 | livinded | theres no connectivity in the hotel room |
| 05:21.00 | dlynes_laptop | livinded: see...now that's what you get for dumping slackware...you get an unstable wifi :p |
| 05:21.14 | CunningPike | Or Yowza! depending on your temperature scale |
| 05:21.22 | livinded | dlynes_laptop: actually i think the fedora package is my problem |
| 05:21.30 | dlynes_laptop | EXACTLY! |
| 05:21.31 | *** join/#asterisk eaperezh (n=erick@201.218.117.178) |
| 05:21.45 | livinded | i never had problems on slackware with the madwifi drivers |
| 05:21.52 | dlynes_laptop | madwifi? |
| 05:21.59 | livinded | atheros based wifi cards |
| 05:22.06 | dlynes_laptop | ah |
| 05:22.11 | P-NuT | Hi all, |
| 05:22.21 | P-NuT | IAX extension dialing out through a sip gateway. |
| 05:22.22 | livinded | the cards are great for certain things but i hate using it for everything |
| 05:22.23 | dlynes_laptop | yeah...i'm just using the intel ipw2945 drivers |
| 05:22.26 | P-NuT | I just get congestion.. |
| 05:22.31 | P-NuT | any ideas? |
| 05:22.34 | eaperezh | good night to all. Can I ask someone here to provide some assistance with a weird problem with a tdm400 (4fxo) card? |
| 05:22.42 | livinded | mosty great for hacking the planet |
| 05:22.52 | livinded | :P |
| 05:23.07 | dlynes_laptop | yeah...wep cracking :) |
| 05:23.34 | livinded | did you get the fragmentation attack to compile on linux |
| 05:23.37 | hads|home | eaperezh: What's the problem? |
| 05:23.48 | dlynes_laptop | never tried |
| 05:23.53 | livinded | i couldn't and neither could the other guys at my lug who tried |
| 05:24.22 | livinded | its suposed to be usable on linux and freebsd but its dependent on bsd headers |
| 05:24.32 | eaperezh | Hi hads: dialing outbound with my tdm400 works prefectly. However dialing from the pstn to the tdm400 has a problem. I hear (pstn side) a nasty loud pitch tone |
| 05:25.42 | eaperezh | hads: I setup a simple context to test the tdm400, it waits 3 secsonds, then Answers() then plays "hello world" |
| 05:25.53 | livinded | i think the problem is he doesn't have headers or kernel source :( and of course there are no headers for his kernel version |
| 05:26.09 | livinded | at least in the apt repository |
| 05:26.26 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 05:26.38 | eaperezh | hads: however as soon as the first rings arrives to the TDM and BEFORE the Answer() part, I from the pstn side start to hear a nasty fax-like tone that continues until i hang up. |
| 05:26.57 | Trazz | is there anyway to turn down the volume on the iVR announcement? |
| 05:27.17 | livinded | Trazz: lower the levels on the recording? |
| 05:27.34 | Trazz | the recording was standard -16db |
| 05:27.36 | Trazz | already |
| 05:27.41 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 05:28.08 | hads|home | eaperezh: And this happens on all of the lines plugged into the card? |
| 05:28.33 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 05:28.53 | *** join/#asterisk mrichmanM (n=richmanm@c-24-20-125-148.hsd1.mn.comcast.net) |
| 05:28.53 | eaperezh | hads: i have two pstn lines, so i tested connecting then to the 4 fxo ports. |
| 05:29.10 | CunningPike | P-NuT: What exactly is the issue? |
| 05:30.08 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:30.31 | hads|home | eaperezh: So the problem occurs on both lines? Have you switched the lines around into the different modules on the card and tested again? |
| 05:30.47 | mrichmanM | i have just installed business edition and it does not start also is their a GUI for the config? |
| 05:31.18 | livinded | mrichmanM: the gui available is bad |
| 05:31.43 | eaperezh | hads: Yes, pstn 1 and 2 on tdm ports 1,2 or 3,4 or 2,3 or 1,4 |
| 05:31.47 | livinded | it criples asterisk and makes it a pain to do anything manually |
| 05:31.51 | mrichmanM | but better than nothing? |
| 05:32.06 | livinded | mrichmanM: its much better to make manual configs |
| 05:32.14 | eaperezh | hads: I then used a phone to test the line and the line is fine (at least for a standard phone) |
| 05:32.27 | livinded | macros can really simplify also |
| 05:32.45 | hads|home | eaperezh: Sounds quite strange. |
| 05:32.50 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 05:33.15 | *** join/#asterisk skraelings001 (n=skraelin@201.230.212.253) |
| 05:33.20 | skraelings001 | hi |
| 05:33.23 | eaperezh | hads: indeed. I called digium this morning and they recommended disabling usb at bios level. I did it, but did not help. |
| 05:34.19 | hads|home | eaperezh: Is the card sharing interrupts at all? Also, does the console output anything useful? |
| 05:34.24 | skraelings001 | anyone that played with AGI ? pls, let me knnow |
| 05:35.25 | eaperezh | hads: I can show you the output. I also have a 1megabyte file with the "tone" recorded. Shall we go private or shall I post it here? |
| 05:36.08 | eaperezh | hads: no int sharing at all. it's a linux box with a x86_64 compilation (nocona) |
| 05:36.55 | hads|home | eaperezh: pastebin the output of the console if you like. I don't know if It'll help though. Sounds more like a hardware issue. |
| 05:37.02 | mrichmanM | so what is the gui I can use with business edition |
| 05:37.10 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 05:38.31 | hads|home | eaperezh: It sounds like you are on the right track talking to Digium support though. |
| 05:39.16 | *** join/#asterisk seb- (n=seb@cpe-72-132-242-171.san.res.rr.com) |
| 05:39.51 | eaperezh | hads: doing pastebin |
| 05:42.35 | eaperezh | hads: pastebin kind of slow from here.. |
| 05:43.50 | eaperezh | hads: it's only a few lines.... |
| 05:43.54 | eaperezh | Verbosity was 3 and is now 35 |
| 05:43.55 | eaperezh | <PROTECTED> |
| 05:43.55 | eaperezh | <PROTECTED> |
| 05:43.55 | eaperezh | <PROTECTED> |
| 05:43.55 | eaperezh | <PROTECTED> |
| 05:43.55 | eaperezh | <PROTECTED> |
| 05:43.57 | eaperezh | <PROTECTED> |
| 05:43.59 | eaperezh | <PROTECTED> |
| 05:44.01 | eaperezh | <PROTECTED> |
| 05:44.03 | eaperezh | <PROTECTED> |
| 05:44.03 | hads|home | DOn't paste it here. |
| 05:44.21 | eaperezh | hads: that's it...ony,ten lines.... |
| 05:44.44 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 05:45.53 | dlynes_laptop | eaperezh: there's more than one pastebin |
| 05:45.56 | dlynes_laptop | ~pb |
| 05:45.58 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 05:46.19 | eaperezh | hads: the sound begins here: Starting simple switch on 'Zap/2-1' |
| 05:46.55 | hads|home | eaperezh: The output looks normal. It sounds like a lower level issue. |
| 05:48.28 | eaperezh | jb: pastebin.ca worked..... pastebin.com is giving timeouts. |
| 05:48.31 | eaperezh | http://pastebin.ca/80224 |
| 05:50.58 | eaperezh | how can i enable debugging on zap channel? |
| 05:51.17 | dlynes_laptop | debug on |
| 05:51.42 | dlynes_laptop | or zap debug channelnum (when a call is established) |
| 05:52.02 | eaperezh | No such command 'zap debug' (type 'help' for help) |
| 05:52.28 | dlynes_laptop | maybe it was debug channelnum then |
| 05:53.58 | hads|home | debug channel blah |
| 05:53.59 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 05:54.44 | eaperezh | where di the debug got written? |
| 05:55.21 | hads|home | check out logger.conf for the gory details |
| 05:57.51 | *** join/#asterisk Tusker (n=tusker@203.117.94.152) |
| 05:57.54 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 05:58.42 | eaperezh | hads: http://pastebin.ca/80230 |
| 06:00.46 | hads|home | eaperezh: There isn't anything there that looks odd to me. Like I said, it sounds like it's at a lower level than Asterisk |
| 06:00.49 | *** join/#asterisk adorah (n=Administ@87.69.72.228) |
| 06:00.57 | *** part/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 06:02.05 | Tusker | heya guys, I'd like to implement follow-me functionality, that allows me to dial in using SIP, and modify the follow me settings ? I've seen http://www.voip-info.org/wiki-Asterisk+Tips+follow+me, but i'm not sure how to have it work with SIP incoming... |
| 06:02.12 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 06:02.19 | eaperezh | hads: funny. the board works at 50% capacity |
| 06:02.33 | eaperezh | hehe |
| 06:02.46 | hads|home | eaperezh: What do you mean by 50% capacity? |
| 06:03.16 | eaperezh | hads: SIP Phones can go out to the pstn using that card without a problem. |
| 06:03.35 | hads|home | Oh, you mean outgoing but not incoming. |
| 06:03.41 | eaperezh | however people from the pastn cannot call in because of that noise i hear... |
| 06:04.49 | hads|home | eaperezh: Have you tried shifting the card around to different PCI slots? |
| 06:04.53 | tengulre11 | Hi,all! |
| 06:05.22 | tengulre11 | how to config a simple iax service? |
| 06:05.29 | eaperezh | well, no. I will do that tomorrow because i exhausted what i think are all "software" issues. |
| 06:06.07 | eaperezh | BRB |
| 06:10.43 | *** part/#asterisk seb- (n=seb@cpe-72-132-242-171.san.res.rr.com) |
| 06:12.56 | tengulre11 | eaperezh, are u active? |
| 06:13.16 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 06:13.38 | *** join/#asterisk dan__t (n=dant@ip70-162-170-55.ph.ph.cox.net) |
| 06:13.45 | dan__t | 'evening. |
| 06:15.20 | tengulre11 | dan__t, 'evening too. |
| 06:15.38 | tengulre11 | dan__t, do u konw how to config a simple iax service? |
| 06:17.49 | dan__t | No. |
| 06:17.52 | dan__t | The manual probably does. |
| 06:18.50 | eaperezh | hads: im back.... Outgoing (sip to zap) goes OK. pstn to zap is the problem |
| 06:19.52 | eaperezh | tengulre11: hi. |
| 06:20.08 | tengulre11 | eaperezh, hi, :) |
| 06:21.36 | dan__t | Is the O'Reilly book any good? |
| 06:22.03 | X-Gen | morning freaks |
| 06:22.05 | drray | yes |
| 06:22.15 | eaperezh | tengulre11: an incoming iax? |
| 06:22.16 | drray | the orielly book is an 8 out of 10 |
| 06:22.33 | dan__t | saw that. |
| 06:26.04 | tengulre11 | eaperezh, incoming and outgoing! |
| 06:26.24 | tengulre11 | eaperezh, I want to test it in lan! |
| 06:26.33 | eaperezh | tengulre11: http://pastebin.ca/80237 |
| 06:27.23 | tengulre11 | eaperezh, thank you very much! |
| 06:29.46 | eaperezh | tengulre11: post resubmitted with changes: http://pastebin.ca/80238 |
| 06:31.08 | eaperezh | tengulre11: a good tutorial site: http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html |
| 06:33.49 | *** join/#asterisk lorinc (n=ang@caracas-3088.adsl.interware.hu) |
| 06:37.41 | droops | hey whats the phone term, for calling a number and then playing a recording? |
| 06:38.57 | *** join/#asterisk operat0r (n=h0msar@adsl-219-206-48.asm.bellsouth.net) |
| 06:39.07 | operat0r | so what does vonage run ? asterisk sphere ? |
| 06:39.37 | Agrajag- | droops: freakin annoying |
| 06:41.08 | droops | isnt that the truth |
| 06:41.35 | |oranjia| | can someone help me with a spot of php AGI? |
| 06:41.37 | Agrajag- | "rude" is the technical term |
| 06:42.19 | operat0r | mmm gimmie AGI |
| 06:46.45 | *** join/#asterisk Un1x (i=seann@CPE00016c29e15a-CM00080d40ee4c.cpe.net.cable.rogers.com) |
| 06:47.05 | Un1x | anyone know any voip providers wich will let me use my own hardware.... |
| 06:49.15 | droops | what kind of hardware? |
| 06:49.51 | Un1x | spa-3000 |
| 06:50.37 | Un1x | i was hopinh one inside of canada |
| 06:50.47 | Un1x | that allow unlimted nationwide calling |
| 06:50.52 | Agrajag- | my spa-3000 is pissing me off! |
| 06:50.53 | Un1x | or one in toronto |
| 06:51.16 | Un1x | lol looks simple |
| 06:51.20 | Un1x | instructions where simple |
| 06:51.26 | Un1x | but it was alot smaller then expected |
| 06:51.30 | droops | ahh |
| 06:51.30 | Un1x | and the pic of spa i seen |
| 06:51.33 | Agrajag- | spa-3000 does sip so any voip provider that does sip should work. or you can make your spa-3000 go to an asterisk box and make that register with the provider |
| 06:51.37 | Un1x | i guess the pic was before linksys bought it |
| 06:51.45 | Agrajag- | yeah it's simple but getting the quality good isn't so |
| 06:51.50 | Un1x | oh |
| 06:51.52 | Un1x | heh yea |
| 06:51.57 | Un1x | but im looking for proividers |
| 06:51.59 | Agrajag- | nope i have a linksys and an older one, they're the same |
| 06:52.02 | Un1x | thaty allow use of own hardware |
| 06:52.05 | Un1x | like vonage |
| 06:52.11 | Un1x | they dotn allow use of own equipment |
| 06:52.12 | Un1x | :S |
| 06:52.22 | Un1x | but if i could find smaller companys im sure theyd allow it |
| 06:52.23 | droops | you might want to check out boradvoice |
| 06:52.28 | droops | err |
| 06:52.29 | Un1x | u mean |
| 06:52.30 | droops | broadvoice |
| 06:52.30 | Un1x | broad |
| 06:52.56 | droops | im not so good at typing, especially on my laptop in bed |
| 06:53.07 | Un1x | wow broadvoice is good |
| 06:53.09 | Un1x | :D |
| 06:53.17 | Un1x | unlimted calling to norway canada etc :P |
| 06:53.19 | Un1x | this will be good |
| 06:53.32 | Un1x | The BroadVoice Unlimited World plan lets you talk all you want. For just $19.95 per month you get unlimited calling to 21 countries including the United States. |
| 06:54.06 | droops | sometimes the service gets sketchy, but it has gotten alot better |
| 06:54.10 | droops | and its dirt cheap |
| 06:54.30 | Un1x | yea but it says |
| 06:54.35 | Un1x | look this page |
| 06:54.41 | Un1x | there charging for adapter :S |
| 06:54.44 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 06:54.46 | Un1x | http://www.broadvoice.com/rateplans_unlimited_world.html |
| 06:55.16 | Un1x | nvm |
| 06:55.21 | Un1x | in signup it gives u option |
| 06:55.27 | Un1x | weather to use ur own sip device |
| 06:56.31 | Un1x | i'll sing up with them after i get another sb5100 :( |
| 06:57.44 | *** join/#asterisk P4C0 (n=paco@201.224.103.235) |
| 06:58.35 | P4C0 | hello guys, does anyone know if there's a way of knowing if the pbx is asterisk by taking a look at the sip traffic? in other ways, does asterisk set like a user-agent field or similar to the sip package??? |
| 07:00.30 | Un1x | droops: know any other companys like Broadvoice ;)? |
| 07:00.56 | dpryo | P4C0: Astersisk sets user-agent, yes |
| 07:01.12 | P4C0 | dpryo: thanks ;) |
| 07:01.27 | operat0r | r |
| 07:04.28 | Agrajag- | does anyone happen to know if the spa-3000 supports more than incoming call on the pstn line? |
| 07:04.33 | *** join/#asterisk tparcina (n=tparcina@lns02-1594.dsl.iskon.hr) |
| 07:06.54 | |oranjia| | operat0r: with php AGI script i cannot seem to get the channel status |
| 07:07.26 | *** join/#asterisk e-ddie (n=oa@62.61.144.75.generic-hostname.arrownet.dk) |
| 07:07.27 | walhala | what does this error mean " decodeMP3: Junk at the beginning of frame" |
| 07:08.50 | CoffeeKid | good question, i get that too :) |
| 07:09.11 | |oranjia| | I do fwrite(STDOUT," CHANNEL STATUS Zap/1-2") and then fgets(STDIN,3000) but it hangs :( |
| 07:09.14 | CoffeeKid | it always happens when i have too many people "on hold" |
| 07:10.27 | CoffeeKid | wish i knew how to fix it :( |
| 07:10.28 | operat0r | |oranjia| I need some agi script if you got em /msg me :) |
| 07:10.55 | |oranjia| | operat0r give me a second :) |
| 07:13.43 | droops | Un1x, no, they did what i needed |
| 07:13.48 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 07:14.41 | *** join/#asterisk Zerthimon (n=Zerthimo@80.74.110.153) |
| 07:16.24 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 07:16.43 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 07:19.17 | operat0r | |oranjia| what does it do ? |
| 07:19.52 | |oranjia| | operat0r: it basically gets an available channel |
| 07:20.08 | |oranjia| | the channel that has the highest rank |
| 07:20.18 | |oranjia| | we want to get a good spread over all our channels |
| 07:20.23 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 07:20.38 | |oranjia| | but i can't seem to get the status of a channel... |
| 07:20.41 | |oranjia| | :( |
| 07:22.01 | *** join/#asterisk nfi|ermes (n=nfi_erme@217.220.121.62) |
| 07:22.26 | operat0r | |oranjia| I got a few ideas for some great AGI scripts but I have not started |
| 07:22.49 | |oranjia| | ah ok |
| 07:22.58 | |oranjia| | but do you know how to get channel status? |
| 07:23.18 | operat0r | Hell no im a asterisk noob :) ill look for ya tho |
| 07:25.02 | operat0r | |oranjia| 1st get it working with out for loop then get it working with sql |
| 07:25.19 | operat0r | |oranjia| does the command work in the console ? |
| 07:25.40 | _4d4m_ | anyone any idea why my polycom 601 periodically does not ring when inbound calls are coming through? |
| 07:26.03 | operat0r | how do I jump on the console to run a AGI script or chnnel stats etc if asterisk is alreay running ? |
| 07:26.08 | _4d4m_ | or coudl anyone point me in the direction of the admin guide as its not avail. online from polycom or my supplier |
| 07:27.17 | *** join/#asterisk Koshatul (n=evangeli@CPE-138-217-188-144.qld.bigpond.net.au) |
| 07:27.17 | *** join/#asterisk damned (n=vpol@prior.lanck.net) |
| 07:27.46 | operat0r | nm |
| 07:27.50 | operat0r | asterisk -r |
| 07:28.22 | operat0r | how can I test that command CHANNEL STATUS |
| 07:32.50 | operat0r | |oranjia| got any other AGI scripts I can look at ? |
| 07:33.49 | operat0r | |oranjia| try channel list 1st before status |
| 07:34.55 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 07:37.21 | |oranjia| | operat0r: will send you stuff as soo as i can :) |
| 07:37.45 | *** join/#asterisk sengland (n=info@dsl081-072-084.sfo1.dsl.speakeasy.net) |
| 07:38.05 | operat0r | |oranjia| just take it out of the loop make sure its not sometihng in the loop or syntax |
| 07:39.06 | |oranjia| | operat0r: even out of the loop it gives me trouble :( |
| 07:39.15 | |oranjia| | and the agi script is called in the dialplan btw |
| 07:39.25 | |oranjia| | the console commands |
| 07:39.37 | |oranjia| | sorry the console commands are not the same as the agi commands |
| 07:39.39 | operat0r | |oranjia| what about channel list ? can you list ok ? |
| 07:40.01 | |oranjia| | operat0r: i can't seem to get any output from asterisk in the script |
| 07:40.13 | operat0r | sip debug |
| 07:40.40 | operat0r | and when you run asterisk run in int -vvvvv mode of corse |
| 07:51.33 | *** join/#asterisk jetway2008 (n=asd@60.48.163.124) |
| 07:53.41 | jetway2008 | could someone pls recommend a cheap clone fxs card |
| 07:56.31 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:01.19 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 08:04.56 | SplasPood | Can anyone suggest to me why setting callerid=Unavailable in the [general] section of sip.conf wouldn't set that callerid for inbound sip calls that have their CID withheld? |
| 08:13.42 | Zerthimon | hi, can I use t and i extensions in [macro] ? |
| 08:14.13 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 08:29.01 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 08:29.40 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 08:35.46 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 08:44.06 | *** join/#asterisk zepmantra (i=wahhh@203.215.100.96) |
| 08:44.13 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 08:55.16 | *** join/#asterisk dudes (n=dudes@71-87-34-39.dhcp.stcd.mn.charter.com) |
| 08:55.20 | *** join/#asterisk RoyK[at] (n=roy@chello080109196173.3.graz.surfer.at) |
| 08:55.30 | Agrajag- | is anyone else using iaxcomm and what sort of latency do you get to asterisk? im on the same lan and get around about 1/2 second with iax. with twinkle (sip) i get a bit less, but it's still very noticeable |
| 08:55.52 | dudes | try kiax |
| 08:55.59 | dudes | or diax if windows |
| 08:57.00 | RoyK[at] | dudes: any mac client available? |
| 08:57.19 | dudes | not sure |
| 08:57.23 | *** join/#asterisk skymarshal (n=stefan@www.wintermeyer.de) |
| 08:57.38 | RoyK[at] | Agrajag-: that sounds really strange... |
| 08:57.39 | dudes | I use linux (x86) |
| 08:57.55 | dudes | why is that strange |
| 08:58.01 | dudes | because I don't use a MAC |
| 08:58.03 | Agrajag- | just tried kiax, same result as iaxcomm |
| 08:58.17 | dudes | then update your shit |
| 08:58.18 | Agrajag- | about 1/2 second |
| 08:58.33 | skymarshal | While trying to install the Junghanns BRI-stuff I get the following message: make[1]: Entering directory `/usr/src/linux-2.6.5-7.257-obj/i386/default' make[1]: *** No rule to make target `modules'. Stop. -- What did I do wrong? |
| 08:58.47 | RoyK[at] | dudes: strange as in half a second latency |
| 08:58.53 | Agrajag- | dudes: talking to me? what shit in particular do i need to update? |
| 08:59.18 | RoyK[at] | skymarshal: rotfl. 2.6.5????? |
| 08:59.30 | dudes | that you have a issue of half a second tells me a lot |
| 08:59.39 | skymarshal | RoyK[at]: it is a SuSE 9.3 - SLES |
| 08:59.55 | Agrajag- | dudes: enlighten me please |
| 09:00.43 | RoyK[at] | skymarshal: 2.6.5 is like > 2 years old |
| 09:01.12 | dudes | I've had setups that went from Minnesota to Vegas to the phillipines to Vegas to Minnesota in less than a quarter of a second |
| 09:01.30 | Agrajag- | i also have an spa-3000, and the analogue phone plugged into the fxs port on that gets pretty much no echo (so no latency). it's all on the same network |
| 09:01.52 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:01.52 | skymarshal | RoyK[at]: The client wants to use a SLES and that is the kernel which is on it. But the problem should be something else. But I don't know what. |
| 09:01.57 | RoyK[at] | dudes: i'd make an ethereal trace and see what's there |
| 09:02.19 | RoyK[at] | skymarshal: sounds like you're missing the kernel source |
| 09:03.05 | dudes | I don't need an ehte-trace because I don't have a issue |
| 09:03.20 | skymarshal | RoyK[at]: Nope: lrwxrwxrwx 1 root root 17 Jul 6 12:52 linux-2.6 -> linux-2.6.5-7.244 |
| 09:03.20 | skymarshal | drwxr-xr-x 20 root root 712 Jul 6 12:50 linux-2.6.5-7.244 |
| 09:05.23 | nfi|ermes | i can t pickup a call from an external trunk (isdn zaphfc) |
| 09:07.44 | nfi|ermes | bristuffed |
| 09:09.22 | Agrajag- | gah.. can't figure this out.. sip definitely has much less latency than though |
| 09:09.28 | kay2 | RoyK[at]: do you use MeetMe() ? |
| 09:10.22 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 09:11.10 | MatsK | skymarshal: could it be so that bristuff demands a never kernel ? |
| 09:11.37 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:12.17 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 09:12.22 | *** join/#asterisk ChrisDE4 (n=ChrisDE@88.128.48.246) |
| 09:13.28 | *** join/#asterisk linuX|Reneger (n=linuxren@p5087E8EE.dip.t-dialin.net) |
| 09:13.31 | ChrisDE4 | hi. I have a problem with asterisk 1.2.9 ... everytime when I look in the console and type "show channels" there are several channels up... why? |
| 09:13.43 | linuX|Reneger | hi |
| 09:15.40 | skymarshal | Matsk: I'll send them an e-mail and ask. |
| 09:15.41 | kay2 | ChrisDE4: pastbin des channels |
| 09:15.42 | kay2 | ~pb |
| 09:15.50 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 09:16.01 | RoyK[at] | skymarshal: it still looks like you're missing the source. quite often only the headers are installed there |
| 09:16.41 | linuX|Reneger | is anyone here that understands German? |
| 09:17.14 | skymarshal | RoyK[at]: The directory /usr/src/linux-2.6 has some 236M in it. I think that would be much for just headers. ;-) |
| 09:17.28 | skymarshal | linuxX|Reneger: Go a head! |
| 09:18.36 | RoyK[at] | skymarshal: then i don't know. try an official kernel |
| 09:18.51 | linuX|Reneger | na linux-de is not so specific for asterisk |
| 09:19.05 | RoyK[at] | erm |
| 09:19.07 | RoyK[at] | asterisk-de |
| 09:19.12 | linuX|Reneger | ah okay thanks |
| 09:19.14 | RoyK[at] | :) |
| 09:19.27 | *** part/#asterisk skymarshal (n=stefan@www.wintermeyer.de) |
| 09:19.36 | linuX|Reneger | there is no one |
| 09:19.59 | linuX|Reneger | than i try it with my bad english |
| 09:20.14 | jetway2008 | could all ATA device be used with asterisk??? |
| 09:20.28 | dudes | most likely ca |
| 09:20.29 | dudes | n |
| 09:20.40 | *** join/#asterisk skymarshal (n=stefan@www.wintermeyer.de) |
| 09:20.45 | dudes | doesn't mean all though |
| 09:24.16 | linuX|Reneger | i have an connecetion problem between an extern caller an my intern Voip Phone, a caller is call trough my Junghans octobri card, an go?s to an voip extension(snom360 Voip phone) the call was connected, but no one hear anything? |
| 09:24.46 | linuX|Reneger | the software is mISDN with de actuell asterisk |
| 09:25.20 | jetway2008 | is the confugration different for each ata |
| 09:25.55 | linuX|Reneger | what do you mean with ata? |
| 09:26.11 | linuX|Reneger | bchannel_activate: BC Not properly upsetted (state:BCHAN_SETUP) addr:50020402 this is my message on the asterisk CLI |
| 09:26.59 | jetway2008 | ATA |
| 09:27.03 | jetway2008 | adapter |
| 09:28.06 | linuX|Reneger | the ISDN adapter? |
| 09:29.19 | jetway2008 | analog phone |
| 09:29.28 | jetway2008 | to asterisk ..so need ATA adapter |
| 09:29.46 | ChrisDE4 | http://channels.debian.net/paste/3060 |
| 09:29.51 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
| 09:29.53 | ChrisDE4 | so this channel is up... |
| 09:29.54 | jetway2008 | pls tell me how do i connect a analog phone to asterisk? |
| 09:29.55 | ChrisDE4 | why? |
| 09:30.32 | ChrisDE4 | plug it into your analog phone card |
| 09:30.49 | *** join/#asterisk AuPix (n=AuPix@mail.aupix.com) |
| 09:31.19 | jetway2008 | thats ATA rite |
| 09:31.34 | ChrisDE4 | yes you can use an ata |
| 09:32.03 | jetway2008 | but will all ATA adapter work or are they specific for * |
| 09:32.30 | ChrisDE4 | I think all atas will work.. |
| 09:32.45 | ChrisDE4 | can anyone tell my why this channels is up? http://channels.debian.net/paste/3060 |
| 09:33.23 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:33.27 | jetway2008 | pls explain the procedure... |
| 09:33.51 | jetway2008 | after plug it to ATA ...how do i configure with * |
| 09:34.10 | ChrisDE4 | you have to define a sip user and register the ata with * |
| 09:36.03 | jetway2008 | ok...thx chris |
| 09:36.16 | ChrisDE4 | another channel that is up: http://channels.debian.net/paste/3061 |
| 09:36.28 | ChrisDE4 | its ringing since 16h43m4s ? |
| 09:40.08 | ChrisDE4 | the user isn't registered anymore to asterisk... but the phone of the called person is still ringing? |
| 09:47.59 | ChrisDE4 | anyone there? |
| 09:49.29 | tzafrir | where? |
| 09:50.03 | ChrisDE4 | any idea how to prevent asterisk from keeping open channels? |
| 09:50.05 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 09:50.09 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 09:50.14 | ghenry | how often should you restart *? |
| 09:50.20 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 09:50.48 | tzafrir | soft hangup the channel? |
| 09:51.03 | ChrisDE4 | yes, but why is it open? |
| 09:51.13 | ChrisDE4 | state is ring |
| 09:51.21 | RoyK[at] | ghenry: generally, asterisk shouldn't need a restart |
| 09:51.37 | linuX|Reneger | lol |
| 09:51.39 | ghenry | RoyK[at]: OK, thanks. |
| 09:51.53 | ghenry | Anyone got any tips for callerid in the UK? |
| 09:53.03 | ghenry | got a tdm400p card |
| 09:53.29 | x86 | ghenry: if you use gentoo it has a ukcid patch it applies to the source |
| 09:53.44 | x86 | ghenry: and a 'ukcid' USE flag you can set to apply the patch |
| 09:54.08 | ghenry | So you still need to patch? I read on voip-info that you just set the cid type etc |
| 09:54.11 | ghenry | hmmm |
| 09:54.26 | x86 | i've never used hardware trunks directly, so i'm not sure |
| 09:54.34 | RoyK[at] | ghenry: you can't set callerid on pots lines |
| 09:54.34 | RoyK[at] | that is, using fxs or fxo? |
| 09:54.58 | Agrajag- | so anyone any ideas on debugging this latency? still 1/2 second echo on Echo() with ping times of 0.140ms using iax |
| 09:55.07 | ghenry | yeah, fxo, i.e. receiving a call, just want to pass it on to the SIP/Zap phones |
| 09:55.16 | ghenry | don't want to set it |
| 09:56.41 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 09:57.15 | santoshr | the doco says tht there is a STRFTIME and a strptime function .. but the cli is not agreeing with tht |
| 09:57.41 | ghenry | RoyK[at]: Can you pass on callerid? |
| 09:57.44 | ghenry | on a fxo |
| 09:57.46 | RoyK[at] | what is a 'hardware trunk'? |
| 09:58.42 | santoshr | i want to find the diffenece between two datetime values.. |
| 09:58.50 | santoshr | i hvae both available |
| 09:58.54 | tzafrir | hmmm... just got a bunch of spam "from" kpfleming. Sadly it got past my not so great spam filter. Seem to come from a computer somewhere in India |
| 09:59.10 | RoyK[at] | no |
| 09:59.43 | RoyK[at] | tzafrir: tried dspam? |
| 10:01.17 | X-Rob_ | tzafrir, Mail::SpamAssassin _is_ your friend. |
| 10:01.47 | RoyK[at] | methinks spamassassin sucks compared to dspam |
| 10:02.47 | mut | hells no |
| 10:03.33 | ghenry | so you can't pas on callerid on fxo lines to SIP phones. Gotya |
| 10:04.02 | mut | my SA does a pretty dern good job |
| 10:05.06 | RoyK[at] | mut: tried dspam_ |
| 10:05.07 | RoyK[at] | ? |
| 10:05.21 | mut | yea for a while on some smaller domains |
| 10:05.41 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 10:05.44 | RoyK[at] | it really is far better than SA |
| 10:05.49 | mut | processing like 5k msgs/day |
| 10:05.56 | mut | i still found my SA install better |
| 10:08.00 | santoshr | how do i get difference between two datetiem values.. they are in this format 2006-07-06 15:12:57 |
| 10:09.04 | santoshr | is thr a way to convert these time to epoch times.. i tried strptime ... but tht function does not exists |
| 10:09.18 | mut | dunno maybe i didn't give it a good enough chance |
| 10:09.19 | mut | but |
| 10:09.26 | mut | spam to my inbox is eliminated |
| 10:09.40 | mut | Total spam detected : 139043 ( 88.04%) |
| 10:09.40 | mut | Total ham accepted : 18893 ( 11.96%) |
| 10:10.07 | nfi|ermes | mut, are u sure about channel you are witing ? |
| 10:10.14 | RoyK[at] | http://karlsbakk.net/fun/poor-magpie.jpg |
| 10:10.28 | mut | nfi|ermes? |
| 10:10.59 | nfi|ermes | talking about spam in #asterisk |
| 10:11.06 | mut | yea... |
| 10:11.09 | nfi|ermes | .D |
| 10:11.15 | nfi|ermes | it s not a problem for me |
| 10:12.50 | *** join/#asterisk burnproof (n=burnproo@210.213.244.61) |
| 10:13.11 | kay2 | l |
| 10:13.21 | burnproof | good day, has anyone from you guys encounter error on building asterisk on latest trunk? |
| 10:13.41 | kay2 | Someone could tell me how to create a conferance room dynamically ? ( without having it in meetme.conf) |
| 10:14.51 | burnproof | it seem's like i can't compile with chan_zap modules |
| 10:15.10 | RoyK[at] | burnproof: have you updated zaptel+libpri? |
| 10:15.38 | burnproof | RoyK: yep actually it happens about yesterday build |
| 10:15.50 | burnproof | RoyK[at]: let me check again the latest source |
| 10:18.54 | *** join/#asterisk RoyK[at] (n=roy@chello080109196173.3.graz.surfer.at) |
| 10:22.01 | burnproof | it's working now weird ... |
| 10:23.26 | santoshr | how do i get difference between two datetiem values.. they are in this format 2006-07-06 15:12:57 is thr a way to convert these time to epoch times.. i tried strptime ... but tht function does not exists |
| 10:24.41 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 10:27.17 | tzafrir | X-Rob_, I'm using spamassassin. Maybe I should upgrade. |
| 10:27.36 | X-Rob_ | tzafrir? eh? |
| 10:27.37 | santoshr | is thr a way to call a sqlfunction from asterisk.. |
| 10:28.04 | tzafrir | <X-Rob_> tzafrir, Mail::SpamAssassin _is_ your friend. |
| 10:28.12 | burnproof | santoshr: you could use AGI function on asterisk |
| 10:28.15 | *** part/#asterisk ChrisDE4 (n=ChrisDE@88.128.48.246) |
| 10:29.06 | *** join/#asterisk Bert- (n=bert@LAubervilliers-151-12-81-84.w193-252.abo.wanadoo.fr) |
| 10:29.10 | Bert- | hello there |
| 10:29.13 | RoyK[at] | santoshr: there's a MYSQL app in -addons |
| 10:29.21 | X-Rob_ | tzafrir, uh ok. |
| 10:30.06 | Bert- | I created an account -> [paul]. It works fine, excepted that paul doesn't have extension, so he can't be called. what is the option to set him an extension please ? |
| 10:30.19 | tzafrir | Anyway, I have finally managed to get asterisk to destroy zaptel channels as I disconnect an astribank unit |
| 10:30.31 | tzafrir | Time to update the drivers... |
| 10:30.36 | RoyK[at] | wtf is an astribank? |
| 10:31.22 | E-bola | Can anybody here help me with configurering an asterisk server for multiple companies? |
| 10:31.45 | burnproof | E-bola: just go ahead ask your question here. |
| 10:31.54 | E-bola | How do u separate the companie? |
| 10:31.57 | RoyK[at] | ~realm |
| 10:32.02 | RoyK[at] | E-bola: using sip? |
| 10:32.06 | E-bola | burnproof: i dont have concreate queastion per say |
| 10:32.08 | E-bola | yes using sip |
| 10:32.20 | E-bola | im very new to asterisk, i only have a simple lan only setup working |
| 10:32.26 | Bert- | RoyK[at] : we made it ;) |
| 10:32.26 | RoyK[at] | ~docs |
| 10:32.27 | jbot | well, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 10:32.29 | RoyK[at] | ~doc |
| 10:32.31 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 10:32.31 | burnproof | E-bola: well realm is your friend :) |
| 10:32.32 | RoyK[at] | ~book |
| 10:32.33 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:32.46 | E-bola | burnproof: are extensions specific for a realm, or are they global? |
| 10:33.17 | burnproof | E-bola: look for sip.conf there are peer section regarding this, i think it could be on a peer basis |
| 10:33.24 | burnproof | or even global i think |
| 10:33.29 | RoyK[at] | E-bola: extensions are in contexts, and each realm can use different contexts |
| 10:33.42 | burnproof | E-bola: that's correct |
| 10:33.49 | E-bola | good |
| 10:34.00 | E-bola | then i guess i coudl set up a realm for each orgonisation |
| 10:34.14 | RoyK[at] | yes |
| 10:34.16 | E-bola | And they would then have no way of interfering with ppl not in their own realm? |
| 10:34.31 | burnproof | E-bola: yes because they are on different context |
| 10:34.39 | E-bola | Cool. thanks for the info |
| 10:34.55 | E-bola | Im sure ile have alot more questions over the next couple of weeks. But ile start with reading more docs |
| 10:35.09 | tzafrir | RoyK[at], ask jbot |
| 10:35.29 | burnproof | E-bola: sure you can if you have PSTN involved on your setup it could be on different context too on a per channel basis |
| 10:35.29 | RoyK[at] | ~astribank |
| 10:35.30 | jbot | astribank is probably a USB-connected channel bank made by Xorcom with zaptel drivers. http://xorcom.com/astribank/features.html . Units of 8 FXS, 16 FXS and 8FXS/8FXO are availble. Drivers: http://xorcom.com/drivers/astribank/ . Ask tzafrir |
| 10:35.54 | RoyK[at] | tzafrir: jbot told me to ask you :) |
| 10:36.20 | tzafrir | Well, earlier it was just "ask tzafrir". |
| 10:36.35 | RoyK[at] | but, usb? is that good? does this thing work well? |
| 10:36.55 | tzafrir | Call our office |
| 10:37.14 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:37.47 | E-bola | Ok guys a new question: Why would i use the various telephone hardware pci cards u can use with asterisks? |
| 10:37.51 | E-bola | I dont get what they are used for |
| 10:38.02 | burnproof | guys just asking well i think this is out of the context, w/c do you prefer teliax or voipjet? just asking no flame pls |
| 10:38.08 | E-bola | Is it if u want to use traditional PSTN locally? |
| 10:38.08 | nfi|ermes | how can i set asterisk to not "see" # as end of the string character |
| 10:38.23 | burnproof | E-bola: yes |
| 10:39.10 | tzafrir | nfi|ermes, why do you actually need that? |
| 10:39.22 | E-bola | thanksa burnproof. We are only about 7 ppl |
| 10:39.29 | E-bola | so i think we will just get new phones or adapters |
| 10:39.30 | puzzled | hi |
| 10:39.45 | tzafrir | If you know the number of digits, you can set up a proper dialplan pattern |
| 10:39.49 | burnproof | E-bola: :) |
| 10:40.04 | nfi|ermes | i need it to compose some strings that require # tone |
| 10:41.54 | nfi|ermes | example: to charge a hpne credit there is a responder that says: to charge , insert your phone number followed by # |
| 10:43.49 | tzafrir | one ugly method: |
| 10:44.15 | tzafrir | exten => _X#,1,something |
| 10:44.25 | tzafrir | exten => _XX#,1,the same something |
| 10:44.29 | tzafrir | etc. |
| 10:45.30 | *** join/#asterisk vivek (n=vivek@unaffiliated/tintin) |
| 10:56.28 | *** join/#asterisk andrebarbosa (n=andrebar@62.48.215.102) |
| 10:57.17 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 10:59.51 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
| 10:59.57 | *** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.162) |
| 11:00.02 | MrChimpy | mroning chaps |
| 11:00.04 | MrChimpy | morning |
| 11:01.58 | *** join/#asterisk danfour (n=fraa@outside.drachenfels.de) |
| 11:04.56 | MrChimpy | Ideally there would be a CLID option to Dial |
| 11:16.17 | jetway2008 | hi |
| 11:16.42 | jetway2008 | which is better freepbx or trixbox? |
| 11:21.09 | *** join/#asterisk pgarcia_br (n=root@200.206.211.121) |
| 11:25.26 | *** join/#asterisk GyrosGeier (n=richter@p54994BF7.dip.t-dialin.net) |
| 11:25.28 | GyrosGeier | hi |
| 11:25.39 | *** join/#asterisk morex (i=morex@host86-133-31-49.range86-133.btcentralplus.com) |
| 11:25.43 | morex | Hey y'all |
| 11:26.00 | morex | Does anybody know how to force an asterisk core dump so I can debug it with gdb? |
| 11:26.08 | GyrosGeier | I'm looking for an embeddable Asterisk configuration frontend. PHP is okay as long as it does not use a database. |
| 11:26.35 | GyrosGeier | morex, you can Ctrl-\ into the terminal it's running in, or simply attach gdb with the -p option |
| 11:27.32 | *** join/#asterisk pgarcia_br (n=root@200.206.211.121) |
| 11:27.47 | morex | Gyros: I've done that, but it's a live system. I need to force a core dump so I can come back to it after restarting Asterisk |
| 11:28.04 | morex | Apparently gcore in gdb will do it, I've just read |
| 11:29.30 | RoyK[at] | morex: killall -11 asterisk :) |
| 11:30.53 | RoyK[at] | GyrosGeier: -p?? docs doesn't mention thatt |
| 11:33.24 | morex | RoyK I'll try that too :-) |
| 11:34.19 | GyrosGeier | RoyK[at], that is IIRC |
| 11:34.33 | GyrosGeier | RoyK[at], it might also be that you just say "gdb $pid" |
| 11:37.00 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
| 11:37.33 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 11:52.56 | tparcina | question: i have this when i do show channels command - http://pastebin.ca/80421 - and there are no calls active. why has this happend? how to make it newer happen again? |
| 11:54.24 | drray | the lines are held open by your provider? |
| 11:56.18 | Zerthimon | anyone able to help with dialplan ? |
| 11:57.06 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
| 11:59.13 | tparcina | drray, as far as i know - no |
| 11:59.40 | tparcina | Zerthimon, shoot, maybe i can help |
| 12:00.08 | Zerthimon | I'm having problem with jumping on ${DIALSTATUS} |
| 12:00.29 | Zerthimon | exten => s,4,Goto(s-${DIALSTATUS},1) |
| 12:00.41 | Zerthimon | exten => s-BUSY,1,Read(OPTION|oz/direct-busy|1) |
| 12:01.11 | Zerthimon | exten => s,3,Dial(${TODIAL},30,rt) |
| 12:01.16 | tparcina | newer used dialstatus in extensions.conf, but sonn i'll have to - it seams like weary usefull |
| 12:02.02 | Zerthimon | it's usefull when it works |
| 12:02.06 | tparcina | and when it's busy it doesn't jump to s-BUSY,1,Read(OPTION|oz/direct-busy|1) ? |
| 12:02.15 | Zerthimon | nope |
| 12:02.30 | Zerthimon | it exits from exten => s,3 |
| 12:02.51 | Zerthimon | doesn't seem to jump to exten => s,4 |
| 12:03.56 | tparcina | use noop(${TODIAL}) - to se what is value of TODIAL variable |
| 12:05.06 | Zerthimon | <PROTECTED> |
| 12:05.16 | Zerthimon | TODIAL sets correctly |
| 12:05.42 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:05.59 | tparcina | fender is here, he'll be able to help you ;))) |
| 12:06.25 | Zerthimon | kewl ^^ |
| 12:07.23 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 12:07.43 | Zerthimon | D-Fender can you help with dialplan ? |
| 12:07.48 | [TK]D-Fender | depends.... |
| 12:07.51 | Zerthimon | cool |
| 12:08.18 | Zerthimon | I'm having problem using ${DIALSTATUS} |
| 12:08.27 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
| 12:08.36 | [TK]D-Fender | pastebin the sections of dialplan thats not working. |
| 12:08.37 | [TK]D-Fender | ~pb |
| 12:08.44 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 12:09.25 | Zerthimon | pasebin eh .. lemme see |
| 12:11.31 | [TK]D-Fender | place you can shove a mountain of text without spamming the channel |
| 12:11.40 | kay2 | Could someone tell me why MeetMe(|Md) does work if there is nothing in meetme.conf ? |
| 12:12.02 | Zerthimon | hmm pb doesn't seem to accept what I'm pasting |
| 12:12.16 | *** join/#asterisk vivek (n=vivek@unaffiliated/tintin) |
| 12:12.46 | Zerthimon | lol they have a problem with a DB |
| 12:13.16 | [TK]D-Fender | Zerthimon : there are a number of PB sites there. ".ca" seems to be the most reliable |
| 12:13.41 | Zerthimon | ok, used .ca |
| 12:13.42 | Zerthimon | http://pastebin.ca/80439 |
| 12:13.45 | *** join/#asterisk RF_MIA (n=Administ@68-235-157-31.miamfl.adelphia.net) |
| 12:16.24 | [TK]D-Fender | Zerthimon : Ok, a minor varient os STDEXTEN, whats not working? |
| 12:16.52 | [TK]D-Fender | Zerthimon : Aside from the fact that it will fail if the phone isn't up, or is anything other than bust/noanswer. |
| 12:17.14 | [TK]D-Fender | Zerthimon : You'll want to take a look at the "catch-all" line in the sample on the WIKI.... |
| 12:17.35 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:17.58 | Zerthimon | the system doesn't jump to line exten => s,4,Goto(s-${DIALSTATUS},1) |
| 12:18.25 | Zerthimon | call hangs up after timeout of exten => s,3,Dial(${TODIAL},30,rt) |
| 12:18.52 | [TK]D-Fender | Zerthimon : pastebin the CLI output of a call like that. |
| 12:19.00 | Zerthimon | ok, in a sec |
| 12:20.16 | *** join/#asterisk sorush20 (n=sorush20@82-43-184-143.cable.ubr07.newm.blueyonder.co.uk) |
| 12:20.48 | Zerthimon | http://pastebin.ca/80445 |
| 12:21.03 | *** join/#asterisk Nettux (n=Quaaaack@adsl-195.137.136.17.tzm.net) |
| 12:21.36 | Nettux | Hello everybody. I need some help with configuring queues |
| 12:21.59 | *** join/#asterisk vlrk (n=vlrk@202.65.134.119) |
| 12:22.31 | [TK]D-Fender | Zerthimon : First you should remove the "r" in your dial... thats normally not a good thing and cheats normal indications |
| 12:22.40 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 12:22.51 | Zerthimon | that doesn't help |
| 12:22.59 | Zerthimon | I tried.. |
| 12:23.06 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 12:23.57 | [TK]D-Fender | Zerthimon : Pastebin your whole dialplan... lets see if something is globally wrong |
| 12:24.09 | Zerthimon | aight, gimme a sec |
| 12:24.23 | [TK]D-Fender | Zerthimon : just for fun though, before doing that, do "show dialplan" and pastbin that context |
| 12:27.10 | tparcina | Zerthimon, do you use priortyjumping, because if you do ti will try to jump n+101 priority |
| 12:27.31 | Zerthimon | http://pastebin.ca/80455 |
| 12:28.11 | Zerthimon | yes I do use priorityjumping ? is that my problem ? |
| 12:28.27 | [TK]D-Fender | Zerthimon : Yes. |
| 12:28.40 | tparcina | yes it is, because when its busy, it will jump to n+101 so it will overjump priority 4 |
| 12:28.42 | [TK]D-Fender | Zerthimon : Priority jumping is very dated. |
| 12:28.53 | DarKnesS_WolF | hello i have a problem with this setup ---> xlite "eyebeam" ----> registering to asterisk <---- Multech Device (MVP210)--->PBX using G729 " included in the device and in eyebeam but when i try to send the DTMF " RFC2833 which supported in both end it's not working fine .. it do dial but but not the correct number and also VAD is disabled on both ends.. any idae? |
| 12:28.56 | vlrk | I know this is not the place to ask related to the phones but i need this info very urgent so iam asking here can any body say me how can we enable a shared line in polycom phone to foward to any other number |
| 12:29.04 | tparcina | Fender: i lead 1:0 :PP |
| 12:29.59 | [TK]D-Fender | tparcina : I suspected as much, but I like seeing the evidence in print first before I execute :) |
| 12:30.40 | tparcina | yes, shure :) |
| 12:30.49 | [TK]D-Fender | vlrk : * does not support SLA.... |
| 12:30.56 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 12:31.14 | RF_MIA | zttest results in zaptel-1.2.6.....are these numbers valid? Does anybody have any feedback? |
| 12:31.27 | Zerthimon | D-Fender: if I remove priorityjumping will it mess up many things ? |
| 12:31.44 | Zerthimon | D-Fender: what should I use instead of PJumping ? |
| 12:32.44 | RF_MIA | I've deployed a half-dozen Dell SC430s with different Digium cards (T1,FXO,FXS,E1, etc.) and they all perform well and call quality is excellent in spite of bad zttest results......any comments? |
| 12:33.05 | [TK]D-Fender | Zerthimon : Should break much. in your dial-outs, get rid of the priority jumping prioritys for congestion and the rest looks pretty clean |
| 12:33.24 | [TK]D-Fender | Zerthimon : ${DIALSTATUS} *IS* instead of priority jumping. |
| 12:34.36 | Zerthimon | D-Fender: Thanks a lot D-Fender that's awesome help |
| 12:35.03 | Zerthimon | I fill like alone in the dark, I appreciate all help |
| 12:35.08 | Zerthimon | fill=feel |
| 12:35.47 | Zerthimon | noone that I know knows asterisk so I have to figure out all by myself :) |
| 12:37.02 | [TK]D-Fender | thats why we're here |
| 12:38.11 | Zerthimon | D-Fender: settiny PJumping to no didn't solve the problem |
| 12:38.49 | *** join/#asterisk hgaillac (n=Harry@183.15.119-80.rev.gaoland.net) |
| 12:39.44 | Zerthimon | the CLI output didn't change either |
| 12:39.48 | [TK]D-Fender | Zerthimon : ratehr than doing that READ for choosing VM / Operator that way, read up on the 2 other "standard" extens : a & o. They are called when the caller presses * and 0 (respectively) while listening to the VM intro meesage before recording to allow a caller to exit VM. Leaving a VM is the TYPICAL response someone would do when calling someone and not getting an answer. Your dialplan should account for the majority scenari |
| 12:40.02 | [TK]D-Fender | Zerthimon : pastebin your extensions.conf |
| 12:40.19 | [TK]D-Fender | Zerthimon : also, what version are you running? |
| 12:41.34 | Zerthimon | sec, let me paste |
| 12:41.40 | *** join/#asterisk PakiPenguin (n=Junaid@linuxpakistan/admin/pakipenguin) |
| 12:41.48 | PakiPenguin | hello there |
| 12:42.18 | Zerthimon | D-Fender: Asterisk 1.2.7.1 |
| 12:42.26 | PakiPenguin | i am trying to make a call thru the manager interface , but it fails for some reason , how can i debug it on the cli or some log file so i can check what is actually happening , i can see my user loggin in and out of the manager interface fine |
| 12:43.08 | Ahrimanes | PakiPenguin: hm i think i have a perlscript to do that.. want that? |
| 12:43.18 | Zerthimon | D-Fender: http://pastebin.ca/80464 |
| 12:43.19 | PakiPenguin | sure Ahrimanes |
| 12:43.33 | Ahrimanes | PakiPenguin: gimme a minute to find it |
| 12:43.38 | PakiPenguin | thanks :) |
| 12:44.22 | *** part/#asterisk RF_MIA (n=Administ@68-235-157-31.miamfl.adelphia.net) |
| 12:45.36 | hgaillac | Hello, I would need help for "outboundproxy" in sip.conf |
| 12:45.44 | morex | exit |
| 12:45.49 | morex | oops :-) |
| 12:46.20 | Ahrimanes | PakiPenguin: install asterisk-perl and use http://pastebin.ca/80465 |
| 12:46.29 | morex | OK really going now. |
| 12:47.10 | uwe | hello, if im moving from asterisk 1.0.x to 1.2.x on another machine ... is there anything special i need to take care about, or would moving the configs be sufficient |
| 12:48.03 | [TK]D-Fender | Zerthimon : Hmmm |
| 12:48.35 | [TK]D-Fender | uwu : Read the WIKI, a lot has changed. Deprecate apps, changes in priority jumping, etc. |
| 12:49.00 | Zerthimon | D-Fender: a dial plan seems ok ? |
| 12:50.39 | [TK]D-Fender | Zerthimon : it "looks" like it doesn't see prio #4 in there.... |
| 12:50.49 | uwe | hmmm, [TK]D-Fender , i seem to search using the wrong keywords, i looked for migrating from asterisk 1.0.x to 1.2.x and other search strings, could you give me a hint for what to look for |
| 12:50.50 | [TK]D-Fender | Zerthimon : do a "reload" and try again |
| 12:51.06 | [TK]D-Fender | uwu "asterisk 1.2" might get you more. |
| 12:51.23 | Zerthimon | D-Fender: I did, also even rebooted the machine |
| 12:51.57 | Zerthimon | D-Fender: yeah, that what's I thought, it doesn't see priority 4 |
| 12:52.22 | [TK]D-Fender | Zerthimon : Something fishy is going on... I'm jsut not sure what.... |
| 12:52.36 | PakiPenguin | thanks Ahrimanes |
| 12:55.40 | Zerthimon | D-Fender: any itea what might be causing it ? |
| 12:56.08 | *** join/#asterisk joelsolanki (n=jnsolank@202.160.161.94) |
| 12:56.45 | *** part/#asterisk hgaillac (n=Harry@183.15.119-80.rev.gaoland.net) |
| 12:59.36 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 12:59.51 | Zerthimon | D-Fender: ok I found it: the dial exits with DIALSTATUS=CANCEL any idea why ? |
| 13:00.10 | *** part/#asterisk sorush20 (n=sorush20@82-43-184-143.cable.ubr07.newm.blueyonder.co.uk) |
| 13:04.42 | joelsolanki | Hi all getting this in sip show channels : 0112715292 1879fe9e378 00102/00000 unkn No Tx: CANCEL |
| 13:05.02 | joelsolanki | dont know why TX: is not removed when call is already disconnected |
| 13:05.12 | *** part/#asterisk mrtwister (n=mrtwiste@107.250.broadband5.iol.cz) |
| 13:05.14 | joelsolanki | can any body tell what is that ? |
| 13:05.15 | Nugget | Don't remove texas! I live there! |
| 13:05.24 | joelsolanki | hehe :) |
| 13:05.43 | Zerthimon | D-Fender: why the dial app exits with DIALSTATUS=CANCEL, any idea why ? |
| 13:05.51 | RoyK[at] | tx == evil |
| 13:06.49 | joelsolanki | what is that Tx: CANCEL ??? |
| 13:07.34 | jbalcomb | why is it not ok to take naps at work? |
| 13:09.00 | Zerthimon | D-Fender: ok, found it, the timeout was too high |
| 13:10.27 | *** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-47-63.w86-213.abo.wanadoo.fr) |
| 13:10.35 | joelsolanki | helo helo |
| 13:10.41 | joelsolanki | <PROTECTED> |
| 13:10.45 | joelsolanki | why do i get this error ? |
| 13:11.06 | jbalcomb | good question |
| 13:11.12 | cybergypsy | hi all , which conf file has the 'number of incoming voip lines' in it ? |
| 13:11.27 | cybergypsy | i have restricted it to 1 and now need to add more |
| 13:13.57 | joelsolanki | ahh nobody is reply to my question :( |
| 13:18.27 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 13:19.18 | *** join/#asterisk Persilon (n=ajolodov@200.123.112.152) |
| 13:19.54 | Persilon | Hello |
| 13:20.09 | Persilon | I'm getting this on the CLI... is it important ? Jul 6 10:02:50 NOTICE[23366]: indications.c:505 ast_unregister_indication_country: Removed default indication country 'ar' |
| 13:22.17 | Bert- | tiens du francais :) |
| 13:23.21 | *** join/#asterisk extremescholar (n=robinson@c-69-242-176-251.hsd1.mo.comcast.net) |
| 13:24.19 | extremescholar | I would like to re-open a bug owned by someone else on bugs.digium.com. I have an account, but I don't see a way to re-open the bug. |
| 13:30.13 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 13:31.51 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
| 13:37.45 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 13:37.57 | *** part/#asterisk extremescholar (n=robinson@c-69-242-176-251.hsd1.mo.comcast.net) |
| 13:38.16 | kmilitzer | Hello everyone ... has anyone a working config for zapata as "Back-To-Back" meaning one channel connected with cross-cable to another? |
| 13:42.50 | RoyK[at] | just set one side to use pri_net signalling and the other to pri_cpe |
| 13:45.02 | kmilitzer | OK, got that far now ... |
| 13:45.40 | kmilitzer | The problem is, I have a card that I suspect is broken, at least I cannot use it with chan_ss7, where it works on another host with the same configs. So I want to try it with chan_zap now |
| 13:45.45 | kmilitzer | As back to back config ... |
| 13:46.06 | *** join/#asterisk ph|ber (n=phiber@slackwaresupport.com) |
| 13:46.26 | ph|ber | anyone have a sample extensions.conf for stripping areacode out of a 10 digit dialed? |
| 13:46.56 | rob0 | * on a laptop: is it overkill? |
| 13:47.15 | rob0 | ph|ber: ${EXTEN:4} |
| 13:47.18 | RoyK[at] | ss7 is just signalling, so it should work with all sorts of pri cards |
| 13:48.28 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 13:49.24 | *** part/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 13:49.27 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 13:49.40 | kmilitzer | RoyK[at]: Yes, I know ... the thing is. I had an ss7 connection working, meaning the E1s were working, MTP-Layer word and SS7 too. Then I started to get CRC error on the MTP-Layer and strange messages regarding the zaptel buffer |
| 13:49.49 | TripleFFFF | what is 'trunk' mode in IAX? |
| 13:50.16 | TripleFFFF | i mean hwat does trunk mode do |
| 13:50.20 | *** part/#asterisk tparcina (n=tparcina@lns02-1594.dsl.iskon.hr) |
| 13:50.22 | RoyK[at] | TripleFFFF: stuff several streams together, send fewer packets |
| 13:50.28 | kmilitzer | RoyK[at]: So I suspected it was a problem with the card, but PRI-Signaling with zaptel seems to work ... |
| 13:50.53 | TripleFFFF | hmm.. is that a good thing ? i mean do i loose cdr records ? or they just aggregate to one ? |
| 13:51.33 | Persilon | hey, is there anyway of using an usb-connected gsm cellular as trunk ? |
| 13:51.44 | tzanger | not that I'm aware of |
| 13:52.48 | Persilon | I saw a way of using a bluetooth connected one as trunk, I tought maybe there was a way of doing the same with usb |
| 13:53.27 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.137.Dial1.SanJose1.Level3.net) |
| 13:54.24 | kmilitzer | Can I run two instances of asterisk on my system? |
| 13:55.23 | ph|ber | so would it be.. exten => _9804NXXXXXX,4,Dial(${LOCALTRUNK}/1${LOCALAC}${EXTEN:4}) |
| 13:55.34 | ph|ber | 804 being the area code |
| 13:56.08 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 13:56.16 | TripleFFFF | yeah |
| 13:56.21 | RoyK[at] | kmilitzer: i don't think there should be any changes with the crcs |
| 13:56.50 | RoyK[at] | TripleFFFF: there'll be no change to CDRs and so on. |
| 13:57.22 | kmilitzer | kmilitzer: Yes, that's what I thought. If there would be a problem with the card, I would also have it with zaptel instead of ss7, but there isn't. So I guess the problem is on the "other side" |
| 13:57.30 | kmilitzer | meant RoyK[at]: |
| 13:57.31 | TripleFFFF | ok |
| 13:57.50 | TripleFFFF | also what is the 0x10 and 0x18 diffe ? i mean whats the tos meaning of them ? |
| 13:58.54 | RoyK[at] | kmilitzer: what sort of card is this? |
| 13:59.00 | kmilitzer | TE205P |
| 13:59.25 | RoyK[at] | k |
| 14:00.43 | Persilon | my boss is bugging me because he thinks there's a way of using his usb-cellphone as asterisk trunk... may anyone tell me whether this is possible or not please ? |
| 14:03.41 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 14:05.16 | [TK]D-Fender | Persilon : does his cellphone act as headphone # speakers? |
| 14:06.04 | *** join/#asterisk Optic (n=dfraser@miso.capybara.org) |
| 14:06.12 | Optic | i have an odd question |
| 14:06.17 | Persilon | [TK]D-Fender: I think he wants outgoing cellphone calls go through it |
| 14:06.18 | Optic | do you know who makes the free music that comes with AMP? |
| 14:06.41 | Persilon | [TK]D-Fender: (lower prices) |
| 14:07.39 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6F38B.versanet.de) |
| 14:07.40 | Nebukadneza | hi |
| 14:07.58 | *** join/#asterisk boddy (n=e@212.58.24.138) |
| 14:08.03 | boddy | hi all |
| 14:08.14 | [TK]D-Fender | Persilon : Didn't answer my question. |
| 14:08.23 | boddy | have any information about linksys pap2 |
| 14:08.48 | boddy | pap2 cant make two call with usig g729 ? |
| 14:08.51 | Nebukadneza | if i have my sipgate (some sip provider) account in sip.conf as [sip1] and dial SIP/<somenumber>@sip1 - why do i get a "No such host: sip1" even if i have a host=<foo> line in [sip1] in the sip.conf? |
| 14:09.06 | [TK]D-Fender | boddy : Probably not like mych of the SPA line. |
| 14:09.54 | Persilon | [TK]D-Fender: I'm sorry... Yes, his cellphone can act as headphone and speaker |
| 14:09.55 | [TK]D-Fender | Nebukadneza : Dial(SIP/sip1/<somenumber>) |
| 14:10.07 | boddy | [TK]D-Fender:I didnt understan exactly |
| 14:10.28 | [TK]D-Fender | Persilon : Then get him a softphone. He may not be able to DIAL/Answer using the phone however. |
| 14:10.36 | Nebukadneza | [TK]D-Fender: oh ok |
| 14:11.24 | MrChimpy | as I understand it a macro call passed to dial won't do what I need, but I'm not sure. |
| 14:11.55 | Persilon | [TK]D-Fender: I think we are talking abou different things... We have a sip trunk, but he wants to use his cellphone as another trunk, so asterisk can use it to make calls |
| 14:14.01 | ph|ber | ok looks like i have a phone issue.. anyone use a polycom 601?? with handset up it will dial 9,1,areacode, number.. but with it down, it doesnt. |
| 14:14.38 | Nebukadneza | thanks - works |
| 14:15.21 | *** join/#asterisk OuterSpace (n=me1@201.255.73.45) |
| 14:17.55 | OuterSpace | hi, i have a menu with option 1 and 2, i want the user to enter one of the options or another extension to dial directly. with: 601 702, it works, but i cant dial extensions 101 and 202 as it thinks its a menu option. how can i solve this? |
| 14:18.19 | [TK]D-Fender | Persilon : AH... forget that :) |
| 14:18.47 | [TK]D-Fender | Persilon : Actually there is a "dock" style product that MIGHT work. Go to the store at www.voxilla.com |
| 14:19.00 | boddy | what is the value of gsm codec like g??? |
| 14:19.10 | Persilon | [TK]D-Fender: I saw a howto to make bluetooth cellphones do what he wants, isn't it possible with usb ? |
| 14:19.14 | [TK]D-Fender | OuterSpace : Easily doable. Pastebin your IVR context and everything linked to it. |
| 14:19.14 | [TK]D-Fender | ~pb |
| 14:19.16 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 14:19.49 | [TK]D-Fender | boddy : GSM isn't a gXXX standard. Most devices don't support it (your PAP2 included |
| 14:20.06 | [TK]D-Fender | boddy : Your only real choices are G711 / G729 |
| 14:20.52 | TripleFFFF | anyone with 7960 ? |
| 14:21.03 | TripleFFFF | my phone hangs on Universal Application loader crap |
| 14:21.59 | boddy | [TK]D-Fender:ok last question is can I buy g723 codec for * |
| 14:22.13 | boddy | if so from where ? |
| 14:22.29 | Bert- | if you have thousands $$ to spend, I suppose you can |
| 14:22.38 | nortex | TripleFFFF, What version of the firmware? |
| 14:22.42 | TripleFFFF | last |
| 14:22.45 | TripleFFFF | no |
| 14:22.57 | TripleFFFF | pos3-07-2-00 |
| 14:23.35 | nortex | I'm on POS-07-5-00, I don't know if it matters, but mine works. |
| 14:23.44 | jbalcomb | Is 'tftpd-hpa' a bad tfto server or something? I'm not getting any syslog entries and when I upload files it tells me they aren't found, as if i'm trying to download them. |
| 14:23.53 | *** part/#asterisk Optic (n=dfraser@miso.capybara.org) |
| 14:24.03 | [TK]D-Fender | boddy : I've never seen it sold anywhere. |
| 14:24.22 | TripleFFFF | nortex can you paste bin the dst settings ? |
| 14:24.34 | TripleFFFF | seems its not getting from my ntpd no more |
| 14:24.40 | TripleFFFF | so no time on phone |
| 14:24.42 | TripleFFFF | no idea why |
| 14:25.03 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:25.19 | TripleFFFF | well i know why.. but.. no time server |
| 14:25.26 | TripleFFFF | then siming out on tftpd |
| 14:26.04 | TripleFFFF | but tftpd is a xinetd crap.. and its running.. udp 0 0 *:tftp *:* |
| 14:26.12 | *** join/#asterisk trbldwine (i=trbldwin@adam.ur.northwestern.edu) |
| 14:27.14 | TripleFFFF | hmm or not |
| 14:27.22 | OuterSpace | [TK]D-Fender : http://bzflag.pastebin.ca/80518 |
| 14:27.43 | nortex | TripleFFFF, Now that you mention it there is no time on my idle screen, maybe the logo cancled it out. |
| 14:27.57 | TripleFFFF | nah |
| 14:28.01 | TripleFFFF | its top left on the black bar |
| 14:28.27 | nortex | TripleFFFF, Mine is blank |
| 14:28.32 | TripleFFFF | me too |
| 14:28.35 | *** join/#asterisk Eecplat (n=ouarf@AStDenis-105-1-33-201.w81-248.abo.wanadoo.fr) |
| 14:28.44 | TripleFFFF | lol but status says.. cant locate tftpd |
| 14:30.07 | Bert- | Hmm Is asterisk able to deals with callback ?? I mean someone call asterisk. Asterisk doesn't answer, but call the initial caller. Then, the caller dial a number, and asterisk dials this number to our SIP account then bridge the two channels together |
| 14:30.13 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:30.21 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 14:30.37 | TripleFFFF | Bert yes |
| 14:30.42 | TripleFFFF | its called a secretary ;) |
| 14:30.59 | TripleFFFF | but theres a service from queue somethign |
| 14:31.03 | TripleFFFF | check wiki for queues |
| 14:31.08 | TripleFFFF | hes spamming all the pages |
| 14:31.18 | [TK]D-Fender | OuterSpace : You need to clean up your includes..... |
| 14:31.42 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:31.42 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:31.42 | Bert- | I don't mean queues |
| 14:31.52 | Bert- | queues works grear |
| 14:31.52 | MrChimpy | trying to bodge what I wanted to do with macro won't work as macro only runs after call connects |
| 14:31.54 | Bert- | great |
| 14:32.22 | OuterSpace | [TK]D-Fender : yes, but anything else ? |
| 14:32.29 | OuterSpace | to do, i mean |
| 14:33.02 | [TK]D-Fender | OuterSpace : Where do calls land on? |
| 14:33.34 | trelane_ | AARGH I'm listening to the acd menu on digium's PBX and I am mentally writing the AGI and such for it in my head as I'm listening |
| 14:34.46 | OuterSpace | [TK]D-Fender : what you mean ? |
| 14:34.48 | *** join/#asterisk drbrown_ (n=keith@63.238.116.149) |
| 14:37.13 | [TK]D-Fender | OuterSpace : http://bzflag.pastebin.ca/80528 |
| 14:38.15 | drbrown_ | how would you grab input from a caller and store it in a variable???? |
| 14:38.37 | [TK]D-Fender | drbrown_ : "show application read" |
| 14:38.51 | drbrown_ | thanks |
| 14:40.43 | TripleFFFF | <PROTECTED> |
| 14:40.52 | TripleFFFF | hmm.. but tftpd still not running.. |
| 14:41.03 | TripleFFFF | iptables is off.. ill reboot i guess |
| 14:41.28 | [TK]D-Fender | WOW, just found a place with cheaper Polycom pricing than Atacomm.... |
| 14:41.58 | TripleFFFF | lol |
| 14:42.04 | TripleFFFF | they maybe selel the 10% defects |
| 14:42.36 | [TK]D-Fender | http://www.telephonydepot.com/IP_Phones_s/1.htm |
| 14:43.06 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:43.13 | [TK]D-Fender | Actually ALL of their prices seem excellent |
| 14:44.00 | [TK]D-Fender | I seriousy wish VoipSupply would clean up the crap on their site.... |
| 14:45.08 | trelane_ | [TK]D-Fender, the atacomm guy gets cool points for stopping by here from time to time |
| 14:45.33 | kmilitzer | OK people, I need your thouhts. I have a TE205P card for use with chan_ss7 in a Server with a P4 3GHz. Over the last few weeks the MTP2 CRC-Errors (MTP2 is one layer above the E1-Layer) with this increased because the zaptel input and output buffers were full or empty or both. This resulted in a complete disfunction of chan_ss7 (no ss7 links could be started). I then tested the card with a cross-cable and zaptel -> No problem. Cross cable and chan_s |
| 14:46.11 | kmilitzer | ... Any thoughts on that? The thing that bothers me is, that it worked once and the CRC-errors increased over time. Otherwise I would have blamed the P4 |
| 14:46.27 | OuterSpace | [TK]D-Fender: thanks a lot, its working, and btw i know where they was landing now heh |
| 14:46.46 | [TK]D-Fender | OuterSpace : I took a (small) chance it was on [incoming]. |
| 14:46.51 | *** join/#asterisk linuX|Reneger (n=linuxren@p5087E8EE.dip.t-dialin.net) |
| 14:47.07 | [TK]D-Fender | OuterSpace : But before you had contexts including themselves in a circle. |
| 14:47.24 | [TK]D-Fender | OuterSpace : like A includes B which in turn includes A again. Can screw stuff up. |
| 14:48.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:49.03 | OuterSpace | it wasnt, it is now, this is my first asterisk, so i'll be around ;-) |
| 14:49.07 | ManxPower | kmilitzer, nobody here uses SS7 |
| 14:50.20 | trelane_ | I've thus far found about 1000 ways to make this tdm24xxp not work. |
| 14:51.25 | kmilitzer | ManxPower: But it seems to me more like a zaptel problem then an ss7 problem. |
| 14:51.39 | Persilon | [TK]D-Fender: I'm still with the usb cellphone thing... what if I connect the cell to a sipphone using it as speaker/headset ? |
| 14:52.38 | *** join/#asterisk argos73 (n=mike@w010.z208036240.chi-il.dsl.cnc.net) |
| 14:52.44 | *** join/#asterisk mrtwister (n=mrtwiste@107.250.broadband5.iol.cz) |
| 14:53.26 | jbalcomb | Anyone doing anything with DHCP to make their different phones pull firmware and configs from different directories on the tftp server? |
| 14:53.59 | mrtwister | hello. have question re: asterisk realtime. sip friends/peers are in table and well working. problem that i have SEVERAL tables and want to use other way, possible in extconfig to use SELECT instead of table name or some function ? MySQL 5. |
| 14:54.22 | *** join/#asterisk hohum (n=dcorbe@mail.interceltelecoms.com) |
| 14:54.58 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:55.06 | jbalcomb | mrtwister: I do not understand your question. |
| 14:55.37 | [TK]D-Fender | Persilon : depends on the the USB driver support in the softphone. |
| 14:55.40 | jbalcomb | mrtwister: Are you asking how to get your information out of asterisks database and into a mysql database? |
| 14:55.46 | mrtwister | extconfig.conf have record |
| 14:55.47 | mrtwister | sipusers => mysql,astelbill,cc_sip_buddies |
| 14:55.47 | mrtwister | sippeers => mysql,astelbill,cc_sip_buddies |
| 14:55.53 | mrtwister | i want to use SQL not tabler |
| 14:56.04 | mrtwister | since have few tables with sip friends |
| 14:56.08 | [TK]D-Fender | jbalcomb : Polycoms allow you to pass the user/pass through DHCP IIRC which would do that. |
| 14:56.37 | Persilon | [TK]D-Fender: The idea is having the usb cellphone connected to a sipphone and using it as speaker/headset, so when I speak from my machine I'll be speaking to the cellphone's headset |
| 14:56.55 | mrtwister | like sipusers=mysql,astelbill,'SELECT * from ......" |
| 14:57.03 | jbalcomb | [TK]D-Fender: I'm looking to catch them on make/model and have dhcpd give them different tftp_root entries |
| 14:57.14 | [TK]D-Fender | Persilon : Check your softphone to see if it'll support it. |
| 14:57.16 | ManxPower | jbalcomb, WHY? |
| 14:57.27 | |oranjia| | hi guys :) |
| 14:57.43 | [TK]D-Fender | jbalcomb : is this for that new app of yours? |
| 14:58.04 | *** join/#asterisk hohum (n=dcorbe@mail.interceltelecoms.com) |
| 14:58.19 | Persilon | [TK]D-Fender: what I'd need in that case is a way of redirecing my sip:XXX call to the cellphone, so it dials the same XXXX telephone |
| 14:58.24 | *** join/#asterisk tamp4x (n=tampon@64.201.13.51) |
| 14:58.29 | [TK]D-Fender | jbalcomb : No way to determine model in DHCP AFAIK. |
| 14:58.31 | *** join/#asterisk idpromnut (n=chris@modemcable157.119-82-70.mc.videotron.ca) |
| 14:58.31 | jbalcomb | ManxPower: c:\inetpub\tftproot\gxp2000 .. ip501 .. c7940 .. sip941, etc |
| 14:58.39 | idpromnut | Hi everyone |
| 14:58.43 | TripleFFFF | whats best linksys to get WRT54G GC OR ? |
| 14:58.50 | [TK]D-Fender | jbalcomb : You may be able to do it based on a partial MAC match. |
| 14:58.53 | jbalcomb | ManxPower too many files to have all phone configs in one dir |
| 14:59.01 | [TK]D-Fender | jbalcomb : But that'd seperate BRAND, not MODEL. |
| 14:59.05 | ManxPower | jbalcomb, so what you are really wanting to do is do this based on the MAC VENDOR. |
| 14:59.08 | jbalcomb | [TK]D-Fender: thats a start, definitely for model |
| 14:59.11 | |oranjia| | anyone got experience in setting up FXO cards? kudzu detects my card but /proc/zaptel doesn't exist :( |
| 14:59.29 | ManxPower | |oranjia|, if kudzu configures the card it won't work with zapte. |
| 14:59.45 | jbalcomb | [TK]D-Fender: i think some of the vendors use the first 2 or 3 digits in the 2nd half to identify the model |
| 14:59.49 | idpromnut | I'm having some issues with a few customers; asterisk is not detecting all DTMF that these customers are dialing (usually the first couple of digits and that's all) |
| 14:59.51 | ManxPower | kudzu will try to load ISDN BRI drivers for the card |
| 15:00.00 | tamp4x | why when i use the reload command asterisk crashes? when i look at /var/log/asterisk/debug, the last message is says, cdr_cuscom.c: Failed to reload configuration file ..... |
| 15:00.02 | idpromnut | Any ideas? I've tried the relaxdtmf option to not avail |
| 15:00.06 | ManxPower | idpromnut, Gains or waits |
| 15:00.07 | tamp4x | cdr_custom.c |
| 15:00.14 | ManxPower | relaxdtmf almost never fixes anything. |
| 15:00.21 | |oranjia| | ManxPower: what do you suggest? is it a driver problem? |
| 15:00.21 | tamp4x | this is asterisk 1.2.9.1 |
| 15:00.24 | idpromnut | ManxPower: higher RxGain? |
| 15:00.27 | ManxPower | idpromnut, you don't tell us if it's incoming or outgoing |
| 15:00.39 | idpromnut | ManxPower: incoming, sorry :) |
| 15:00.44 | ManxPower | |oranjia|, find out how to unconfigure the box |
| 15:00.59 | ManxPower | idpromnut, so call comes User -> PSTN -> Asterisk |
| 15:01.01 | jbalcomb | ManxPower: the only problem with relying on a vendor only organizational structure is I have several grandstream, polycom, and sipura models |
| 15:01.13 | idpromnut | ManxPower: customer makes an inbound call to asterisk; then dials a number |
| 15:01.13 | *** join/#asterisk mmmmmToop (n=mmmmToop@firewall.datapro.co.za) |
| 15:01.15 | ManxPower | jbalcomb, It sucks to be you. |
| 15:01.22 | idpromnut | ManxPower: correct |
| 15:01.31 | ManxPower | idpromnut, Is the customer on the PSTN or on the Asterisk phones? |
| 15:01.34 | idpromnut | ManxPower: where PSTN is provided by PRI |
| 15:01.38 | jbalcomb | ManxPower: As ever, you are a sweetheart. |
| 15:01.42 | ManxPower | idpromnut, so you are using DISA |
| 15:01.44 | idpromnut | ManxPower: PSTN |
| 15:01.50 | idpromnut | ManxPower: Yes |
| 15:01.55 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 15:02.05 | ManxPower | idpromnut, increase or decrease rxgain and txgain. |
| 15:02.09 | *** join/#asterisk joe_acme (i=HydraIRC@mar92-9-82-237-75-54.fbx.proxad.net) |
| 15:02.23 | tamp4x | anyone? |
| 15:02.26 | idpromnut | ManxPower: any clues as to which is the better one to start with? |
| 15:02.36 | idpromnut | ie more likely? |
| 15:02.47 | ManxPower | If Asterisk is just MISSING digits then increase the rxgain, if it is getting double digits then reduce both txgain and rxgain |
| 15:03.07 | ManxPower | idpromnut, Try it until it works. That's what I did. |
| 15:03.42 | ManxPower | My issue was the txgain was far too high, causing echo that was loud enough for Asterisk to think the dame DTMF tone was sent twice. |
| 15:04.05 | |oranjia| | ManxPower: if i remove the card, no configs are loaded. Should I first compile the zaptel drivers and then re-insert the card? |
| 15:04.17 | ManxPower | |oranjia|, I do not know. |
| 15:04.27 | *** join/#asterisk jetway2008 (n=asd@60.50.26.221) |
| 15:04.43 | ManxPower | |oranjia|, the 1st and ONLY time I ever let kudzu config the card was YEARS ago and I have no memory of how I fixed it. |
| 15:04.50 | jetway2008 | hi |
| 15:05.01 | jetway2008 | which is better trixbox or freepbx? |
| 15:05.07 | ManxPower | I may very well have reinstalled from scratch since I use scripted install disks for Mandrake and it would have been a new server |
| 15:05.08 | Qwell | jetway2008: see topic |
| 15:05.12 | |oranjia| | ManxPower: Thanks anyways :) |
| 15:05.28 | trelane_ | I've got a Dell PowerEdge 830 server. Two of the three PCI slots share IRQ's (one with SMBUS one with IDE) I cannot adjust either hte smbus or ide irq's in bios and I need all three pci slots. I can either make asterisk work (to an extent as disk access is borken) or I can make the raid controller work, but not both, does anyone have any suggestions or experience with this platform running asterisk? (the digium card is a 24xxp) |
| 15:05.42 | *** part/#asterisk Nebukadneza (n=daddel9@i3ED6F38B.versanet.de) |
| 15:05.49 | jetway2008 | heello |
| 15:06.03 | TripleFFFF | so |
| 15:06.04 | ManxPower | trelane_, I recommend a TE100P and a channelbank |
| 15:06.13 | TripleFFFF | whats the better one ? wrt54gl ? |
| 15:06.16 | ManxPower | jetway2008, Hello. Read the damn /topic |
| 15:06.39 | ManxPower | TripleFFFF, the L has 2x the flash and 2x the ram. you decide., |
| 15:06.40 | Qwell | TripleFFFF: the netgear is way cooler |
| 15:06.43 | trelane_ | ManxPower, I'd love to do that but at this point I've got the hardware, is the TE100P more resiliant to IRQ issues? |
| 15:06.48 | Qwell | usb ftw |
| 15:07.00 | ManxPower | trelane_, No. |
| 15:07.02 | TripleFFFF | in the linksys i mean |
| 15:07.03 | jetway2008 | u mean i cant ask any other question out of the topic |
| 15:07.05 | jetway2008 | goshhh |
| 15:07.11 | TripleFFFF | ManxPower so WRT54GL ? |
| 15:07.15 | *** part/#asterisk joe_acme (i=HydraIRC@mar92-9-82-237-75-54.fbx.proxad.net) |
| 15:07.20 | trelane_ | ManxPower, then how will it help with the described issue? |
| 15:07.39 | ManxPower | jetway2008, your question is like asking which is better a buckyball or a carbon nanotube. We don't know anything about either one and so cannot answer you. |
| 15:08.00 | ManxPower | trelane_, 1) I do not trust ANY Digium analog card. |
| 15:08.17 | ManxPower | trelane_, 2) The TExxxP cards work well and a good channel bank works well. |
| 15:08.33 | jetway2008 | isnt both similar?? |
| 15:08.33 | ManxPower | 3) if you share IRQs you will have problems. |
| 15:08.41 | trelane_ | I don't have a choice with 3 |
| 15:08.56 | trelane_ | which is why I was asking if anyone was familiar with what has to be a reasonably common platform |
| 15:08.59 | mut | yes you do |
| 15:09.04 | mut | buy a new motherboard |
| 15:09.14 | drray | buy a different motherboard |
| 15:09.15 | ManxPower | jetway2008, I have no idea. All I know is both use complicated scripts that would make rube goldberg proud |
| 15:09.27 | ManxPower | trelane_, If you share IRQs it will not be reliable. |
| 15:09.34 | Qwell | ManxPower: I think he would be ashamed |
| 15:09.41 | jetway2008 | ok |
| 15:10.05 | trelane_ | Im sorry but "buy a new motherboard" doesn't cut it, this is a bog-standard dell server |
| 15:10.13 | Qwell | trelane_: call Dell.. |
| 15:10.17 | trelane_ | Qwell, I did |
| 15:10.19 | jetway2008 | do u know of cheap fxs clone card? |
| 15:10.23 | ManxPower | trelane_, It won't work. Simple as that. |
| 15:10.23 | Qwell | call Gold support |
| 15:10.26 | idpromnut | ManxPower: wild.. thanks :) I'm guessing I can use the ztmonitor app to help tune the gains, correct? |
| 15:10.29 | ManxPower | jetway2008, they do not exist. |
| 15:10.33 | drray | you paid $500-$2300 for a digium card, but wont budge on a $50 mb? |
| 15:10.47 | trelane_ | drray, ITS A DELL SERVER! |
| 15:10.47 | ManxPower | idpromnut, you could, but just increse or decrease by 3 to start out with |
| 15:10.58 | trelane_ | what part of it's a propriatary motherboard isn't therein contained? |
| 15:11.04 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 15:11.08 | ManxPower | drray, if you can find a motherboard to fit the server, I'll bet he'll pay a lot more than $50 to you |
| 15:11.15 | trelane_ | ManxPower, exactly |
| 15:11.23 | jetway2008 | how abt phone jack |
| 15:11.26 | drray | I can, but it comes with an adapter, called a case |
| 15:11.45 | ManxPower | trelane_, your best bet is to read the mailing list archives, then exchange the machine for one that is known to work well and not to share IRQs |
| 15:11.51 | ManxPower | jetway2008, "phone jack" |
| 15:12.25 | *** join/#asterisk ionix (n=ionix@p1200-ipbfp05miyazaki.miyazaki.ocn.ne.jp) |
| 15:12.59 | *** join/#asterisk TeePOG (n=1234@dsl-145-151-26.telkomadsl.co.za) |
| 15:13.06 | ManxPower | trelane_, A Digium card generates 8,000 interrupts per second and has NO onboard buffering. |
| 15:13.07 | TeePOG | hi |
| 15:13.38 | tzafrir | phonejack are of the same price range as the digium cards. So are some some clones of the digium cards |
| 15:14.43 | ManxPower | tzafrir, Didn't Digium remote the phonejack/linekack drivers because they didn't work well? |
| 15:14.50 | tzafrir | no idea |
| 15:15.13 | tzafrir | For some reason they appear on the "supported hardware" page |
| 15:15.17 | jetway2008 | <PROTECTED> |
| 15:15.22 | jetway2008 | yeap |
| 15:16.20 | *** join/#asterisk Delta239 (n=adfadsf@201.226.130.55) |
| 15:17.06 | jetway2008 | any experience with phonejack??? |
| 15:17.40 | jbalcomb | jetway2008 Have you considered `/part` && `/j #freepbx`? |
| 15:18.40 | jetway2008 | hmmm |
| 15:18.47 | jetway2008 | thinking abt it jballcomb |
| 15:20.25 | TripleFFFF | kik\ |
| 15:20.27 | TripleFFFF | lol |
| 15:20.37 | TripleFFFF | hmm backticks work ? |
| 15:20.54 | TripleFFFF | somebody HAS to stpo this india crap call center pos |
| 15:21.08 | jetway2008 | rite |
| 15:21.10 | Qwell | TripleFFFF: Dell? Sprint? Linksys? |
| 15:21.14 | TripleFFFF | just ot a paypal guy try to talk to me.. i said ill put you on hold.. he said no no is ok ..all good to go |
| 15:21.18 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 15:21.18 | TripleFFFF | paypal |
| 15:21.19 | TripleFFFF | lol |
| 15:21.26 | Qwell | ahh |
| 15:21.28 | Qwell | lame |
| 15:21.44 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 15:21.50 | TripleFFFF | nowadays all do it.. its like having sex with a lamb.. no one says they do but secretely.. hehheehe |
| 15:21.51 | TripleFFFF | j/k |
| 15:21.55 | Nivex | When you call a place up and they say "Will you hold please?", what happens if you say "No." ? |
| 15:22.07 | Qwell | Nivex: Then you don't get put on hold. ;) |
| 15:22.10 | TripleFFFF | never tried |
| 15:22.20 | Qwell | Nivex: They say "Sir, in order to get my supervisor, I must put you on hold." |
| 15:22.28 | TripleFFFF | BUT i do run the telmarketers hell script .. and that where i put my parking for spammers |
| 15:22.29 | Qwell | Then you say "too bad, do NOT put me on hold" |
| 15:22.38 | Qwell | "Sir, please 2 be listening to me" |
| 15:22.53 | *** join/#asterisk esculapio__ (n=ESCulapi@200.88.44.66) |
| 15:22.55 | Nivex | Qwell: No, I've actually waited in a queue for a few minutes and then when they answer INITIALLY they ask if they can put me on hold. |
| 15:22.57 | Qwell | "Go get your supervisor, and DO NOT put me on hold." |
| 15:23.04 | Qwell | Nivex: yeah, same thing |
| 15:23.04 | Nivex | One of these days I'm gonna be like "That's what I've BEEN doing!" |
| 15:23.06 | *** part/#asterisk jetway2008 (n=asd@60.50.26.221) |
| 15:23.07 | Qwell | Nivex: They still can't :p |
| 15:23.19 | *** join/#asterisk jetway2008 (n=asd@60.50.26.221) |
| 15:23.31 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001217014b92.vc.shawcable.net) |
| 15:23.45 | Qwell | Nivex: You've gotta learn to bully your way around call centers...heh |
| 15:24.08 | luke-jr_ | Anyone know a *reliable* VoIP company that supports LNP, doesn't require an ATA, and supports POTS call forwarding? |
| 15:24.23 | Nivex | thankfully I don't have to call to call centers that often |
| 15:24.27 | luke-jr_ | in particular, simply to port an old number to forward to a new one cheaply |
| 15:24.29 | Qwell | Nivex: meh |
| 15:24.36 | Qwell | I have to call one probably...twice a month? |
| 15:24.43 | Qwell | just for my own personal reasons |
| 15:25.29 | vader-- | What are you guys using if a secretary needs to see if their boss is on the phone? |
| 15:25.31 | Qwell | hell, just the other night, I got a gold desktop support tech at Dell to support an issue I was having...with a laptop... |
| 15:25.36 | Qwell | when I didn't have a gold warranty... |
| 15:25.42 | Qwell | or ANY warranty, for that matter :D |
| 15:25.49 | vader-- | i have cisco 7940 phones and i need some way for the secretaries and stuff to see if people are on the phone |
| 15:26.01 | Qwell | vader--: use sccp with hints |
| 15:26.34 | Qwell | rob0: no |
| 15:26.45 | vader-- | Qwell im already using sip |
| 15:26.50 | Qwell | vader--: then you can't |
| 15:27.10 | vader-- | are there any advantages to using sccp over sip? |
| 15:27.14 | Qwell | see above |
| 15:27.18 | vader-- | other than that? |
| 15:27.24 | Qwell | yeah, sip sucks |
| 15:28.00 | rob0 | I'm thinking with an ATA or softphones, I could be pretty mobile. :) |
| 15:28.20 | CunningPike | Qwell: SIP has hints...... but I figure you already know that - do Cisco phones not do SIP hints? |
| 15:28.22 | dlynes_laptop | rob0: try http://www.rogers.com/portable |
| 15:28.37 | CunningPike | Morning, dlynes_laptop |
| 15:28.39 | dlynes_laptop | rob0: you can use the internet anywhere...even in yhour car |
| 15:28.52 | dlynes_laptop | morning, cp |
| 15:31.28 | lilalinux | what sip client for linux with video support do you recommend? |
| 15:32.35 | [TK]D-Fender | vader-- : Presence on the phone or web-app. |
| 15:32.41 | Qwell | CunningPike: no, they don't |
| 15:32.51 | CunningPike | Qwell: Bummer |
| 15:33.02 | *** join/#asterisk ambriento (n=ambrient@www.cobranet.com.br) |
| 15:33.19 | [TK]D-Fender | vader-- : Oh yeah.. I already told you about all this! Either write an XML scrip for it, or lookup "cisco presence" on the WIKI to see what you can do on them. |
| 15:33.56 | *** join/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 15:34.31 | dlynes_laptop | CunningPike: why would cisco want to follow a sip standard when they can follow their own sccp standard? :p |
| 15:34.41 | dlynes_laptop | CunningPike: sccp supports presence awareness |
| 15:35.25 | CunningPike | dlynes_laptop: True, but if you're going to provide SIP, why not do it properly? Especially if you're making people pay for it |
| 15:35.53 | dlynes_laptop | because then you get to see that angry grimace on their face when they realize they got screwed |
| 15:36.07 | CunningPike | lol |
| 15:36.17 | [TK]D-Fender | vader-- : WIKI page says no SIP presence. So go make an XML browser scripts for it. |
| 15:37.26 | *** part/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 15:41.07 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 15:41.37 | *** join/#asterisk s0lid (n=s0lid@203.177.12.98) |
| 15:41.42 | *** part/#asterisk s0lid (n=s0lid@203.177.12.98) |
| 15:41.50 | *** join/#asterisk s0lid (n=s0lid@203.177.12.98) |
| 15:42.50 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 15:43.00 | mrtwister | Jul 6 06:40:14 WARNING[27478]: rtp.c:397 ast_rtcp_read: RTP Read too short |
| 15:43.08 | mrtwister | after that it hngup call |
| 15:43.16 | mrtwister | in about 5-10 seconds |
| 15:43.27 | mrtwister | where i can found issue, any ideas ? |
| 15:43.52 | CunningPike | mrtwister: Sounds like your RTP stream is getting broken along the way. What's the latency like on your connection> |
| 15:43.59 | *** part/#asterisk dudes (n=dudes@71-87-34-39.dhcp.stcd.mn.charter.com) |
| 15:44.03 | mrtwister | 250 |
| 15:44.09 | mrtwister | it is DID number from china |
| 15:44.11 | mrtwister | for calling card |
| 15:44.22 | mrtwister | problems only with that DID |
| 15:44.27 | mrtwister | and only if i use DTMF |
| 15:44.41 | CunningPike | mrtwister: Inband DTMF? |
| 15:44.54 | mrtwister | rfc |
| 15:45.24 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 15:45.35 | *** join/#asterisk davewise (i=icechat5@66.251.196.193) |
| 15:45.36 | mrtwister | calling to chinese DID, asterisk respond, pin code .... destination.... beign ring out any destination and hangup |
| 15:45.49 | TripleFFFF | would you guys say nagios is the most used/ best monitoring syste,m ? |
| 15:45.50 | mrtwister | same script under italian, usa did are ok |
| 15:45.56 | *** part/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 15:46.03 | *** join/#asterisk overshard (n=isaac@nc-205-240-45-138.sta.embarqhsd.net) |
| 15:46.17 | davewise | Is anyone using G.729 liscences? |
| 15:46.18 | *** join/#asterisk codestr0m (n=asura@ns2.netsyncro.com) |
| 15:46.24 | tamp4x | anyone here use backtrace on asterisk |
| 15:46.35 | TripleFFFF | dave yes |
| 15:46.37 | CunningPike | mrtwister: No idea, sorry |
| 15:46.44 | mrtwister | :( |
| 15:46.57 | CunningPike | TripleFFFF: We use Intermapper - but it's not cheap..... |
| 15:47.00 | davewise | does anyone know if you can buy like 5 for one box and later add 5 more ? |
| 15:47.19 | davewise | or do they all need to be purchased at once? |
| 15:47.27 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 15:47.54 | codestr0m | If I wanted to do selective call blocking or redirect.. what's the best way to set that up? |
| 15:48.01 | overshard | Hello... I am having trouble getting a server to recognize the zaptel cards when trying to bring them up i get the error ZT_SPANCONFIG failed on span 1... anyone have any clue how i can fix this problem? |
| 15:48.29 | davewise | Is Nagios not a good monitoring tool? |
| 15:50.12 | *** join/#asterisk smackus (n=smackus@63.149.122.94) |
| 15:50.25 | smackus | ok, so I have a new issue this morning :-D |
| 15:50.52 | smackus | has anyone ever had a agent drop a call and hear the message "call time exceeded"? |
| 15:51.07 | smackus | apparently I have set a call time limit and I cannot find where. |
| 15:51.21 | [TK]D-Fender | smackus : Start pastebining everything.... |
| 15:51.43 | smackus | ok. |
| 15:51.54 | smackus | queues and agents .conf? or do you want more? |
| 15:52.43 | [TK]D-Fender | smackus : EVERYTHING related obviously. |
| 15:52.52 | codestr0m | selective call blocking.. how do I set this up? |
| 15:53.27 | *** join/#asterisk ambriento (n=ambrient@www.cobranet.com.br) |
| 15:54.32 | [TK]D-Fender | codestr0m : Selective how? |
| 15:54.43 | *** join/#asterisk vivek (n=vivek@unaffiliated/tintin) |
| 15:54.50 | codestr0m | ex gf calls.. redirect her to a shoo go away message or dev/null |
| 15:54.53 | codestr0m | that sort of idea |
| 15:55.15 | codestr0m | based on from CLI/ANI |
| 15:55.23 | vivek | what's the difference between the caller id standards of Denmark and sweeden ? both use DTMF .... but are listed separatley in spa3000 |
| 15:55.23 | syzygybsd | if I wanted to load test an asterisk box with all 96 lines (92) lines, is there a place where I can call that has 96 lines to test |
| 15:55.49 | codestr0m | syzygybsd: sipp try that |
| 15:55.51 | cybergypsy | can anyone tell me where the number of voip calls limit it ? |
| 15:56.06 | cybergypsy | number of in/out calls at once |
| 15:56.22 | SplasPood | Can anyone suggest to me why setting callerid=Unavailable in the [general] section of sip.conf wouldn't set that callerid for inbound sip calls that have their CID withheld? |
| 15:56.28 | [TK]D-Fender | codestr0m : "show command gotoif" "show function CALLERID" |
| 15:56.34 | vivek | codestr0m: how cruel !!! cruella ;) |
| 15:56.47 | syzygybsd | think they would get annoyed at louie louie playing? (what i use to hear if there is static/crackling) |
| 15:57.03 | SplasPood | syzygybsd: heh I use a milliwatt tone for that :P |
| 15:57.13 | codestr0m | vivek: indeed, but I'm sure this will be effective.. |
| 15:57.17 | [TK]D-Fender | SplasPood : just check it in your incoming context. |
| 15:57.27 | smackus | [TK]D-Fender: http://pastebin.ca/80577 |
| 15:57.33 | syzygybsd | SplasPood: good idea... |
| 15:57.36 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 15:57.48 | *** join/#asterisk salviadud (n=ralfalfa@201.137.164.143) |
| 15:57.49 | codestr0m | so.. now I just need to find the sounds for "This number has been disconnected" |
| 15:58.05 | syzygybsd | I like the idea of having talking because it gives a chance for echo cancelers to work etc |
| 15:58.06 | codestr0m | is that part of asterisk-sounds already? |
| 15:58.26 | SplasPood | [TK]D-Fender: if it's asterisk, then reset it.. yea I could do that... but isn't callerid= in [general] SUPPOSED to do that? |
| 15:58.47 | syzygybsd | codestr0m: I think there is a "this number is no longer in service" |
| 15:59.06 | *** part/#asterisk overshard (n=isaac@nc-205-240-45-138.sta.embarqhsd.net) |
| 15:59.14 | codestr0m | syzygybsd: thanks.. the puzzle is coming together.. I have some work todo now.. thanks |
| 15:59.23 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.238.243.Dial1.SanJose1.Level3.net) |
| 15:59.29 | syzygybsd | or Playback(has-been-disconnected) |
| 15:59.33 | *** join/#asterisk coppice (n=chatzill@61.197.17.210.dyn.pacific.net.hk) |
| 16:00.10 | mrichmanM | Can someone ppoint me to a how to for asterisk business edition? |
| 16:00.10 | vivek | lol she is gonna try from a payphone and ... come over and wack you on your head ... |
| 16:00.25 | mrichmanM | i installed it but it does not start up |
| 16:00.37 | syzygybsd | make it simple codestr0m 'find /var/lib/asterisk/sounds/|grep -i discon' |
| 16:00.44 | mrichmanM | the quickstart guide has been completed |
| 16:00.48 | tamp4x | when i reload asteirsk i get a segfault...anyone have anyideas why? |
| 16:00.49 | codestr0m | vivek: she's in another state.. and soon I'll be back in Sweden. :P |
| 16:01.11 | tamp4x | when i reload asterisk i get a segfault...anyone have any ideas why? the last app to reload is queues.cong |
| 16:01.52 | tamp4x | ok my queus.conf is fucked nm |
| 16:01.55 | syzygybsd | tamp4x: normally there is a little more information then just segfault... what level of verbosity do you have |
| 16:02.05 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 16:02.25 | vivek | is anyone running asterisk and openvpn on openwrt at the same time ? Will wrt54gl be able to handle this ? |
| 16:02.46 | SplasPood | it'll handle it presuming you're not using meetme, and not doing any transcoding |
| 16:02.55 | SplasPood | and even then it won't handle much... (afaik) |
| 16:03.16 | CunningPike | mrichmanM: Contact digium - you paid for support when you bought it |
| 16:03.26 | vivek | SplasPood: that's ok i just need to pass one call between two wrt54gl's via vpn ... |
| 16:03.31 | vivek | no more ... no less |
| 16:03.48 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 16:03.51 | *** join/#asterisk overshard (n=isaac@nc-205-240-45-138.sta.embarqhsd.net) |
| 16:03.53 | SplasPood | presuming you're using the same codec all the way through, it should work.. |
| 16:04.01 | overshard | Anyone know how to fix... Running ztcfg: ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 16:04.01 | overshard | <PROTECTED> |
| 16:04.14 | vivek | SplasPood: and end up on the ata's ... spa2k and spa3k |
| 16:04.16 | vader-- | so do the cisco 7940G phones not support sip hints? |
| 16:04.20 | vivek | cool tnnx |
| 16:04.43 | [TK]D-Fender | vader-- : Apparently not. So time for Plan-"B" |
| 16:04.56 | vivek | i will just use g723a thoughout ... |
| 16:05.00 | *** join/#asterisk ToyMan (n=stuq@74-32-9-135.dsl1.mdl.ny.frontiernet.net) |
| 16:05.27 | rob0 | vivek: Consider the limitations of CPU and RAM. A more robust setup might be to have the openvpn and * NAT'ed from the openwrt. |
| 16:05.28 | CunningPike | overshard: Sounds like a possible error in your config files - can you pastebin zaptel.conf? |
| 16:05.31 | CunningPike | ~pb |
| 16:05.32 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 16:05.46 | overshard | CunningPike, yes give me a min |
| 16:05.47 | CunningPike | Use pastebin.ca - pastebin.com is fscked |
| 16:06.03 | *** part/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 16:06.08 | syzygybsd | is there a good way to watch the asterisk console from 4 or more machines other then having a seperate connection for each? |
| 16:06.30 | *** part/#asterisk codestr0m (n=asura@ns2.netsyncro.com) |
| 16:06.39 | Nugget | syzygybsd: use screen |
| 16:06.45 | [TK]D-Fender | syzygybsd : many 1 connection and then have a VNC party on it :) |
| 16:07.06 | [TK]D-Fender | make* |
| 16:07.16 | syzygybsd | oh, hmm.. I meant 4 asterisk consoles viewing on 1 machine... |
| 16:07.24 | syzygybsd | didn't describe that well, sorry |
| 16:07.36 | Nugget | buy a bigger monitor. |
| 16:07.52 | syzygybsd | lol.. will get my boss on that, already have 2... |
| 16:08.03 | overshard | CunningPike, here it is http://pastebin.ca/80582 |
| 16:09.08 | vivek | rob0: what's * NAT' ed ? sorry i am new to all of this ... are you suggesting sipexpress or something similar ? |
| 16:09.18 | *** join/#asterisk boch (n=root@201.216.241.97) |
| 16:09.48 | CunningPike | overshard: zaptel, not zapata - I think it's zaptel we need to look at, as zaptel's not even loading |
| 16:10.07 | overshard | CunningPike, oh yes sorry where is the zaptel again? |
| 16:10.18 | CunningPike | overshard: /etc |
| 16:10.19 | [TK]D-Fender | AMP yay! |
| 16:10.51 | vivek | i was blissfully unaware of voip a week ago ;) and using my cellphone with wild abandon ... i will miss that ... |
| 16:13.04 | overshard | CunningPike, got the right one this time http://pastebin.ca/80585 |
| 16:13.44 | boch | gentlemen, my asterisk is not reading some DTMF codes, is this a common issue ? |
| 16:14.29 | CunningPike | overshard: Is this A@H/Trixbox or whatever? |
| 16:14.40 | overshard | CunningPike, it is AAH |
| 16:14.46 | smackus | Any thoughts on my call time exceeded message |
| 16:15.02 | CunningPike | overshard: |
| 16:15.05 | CunningPike | ~A@H |
| 16:15.20 | *** join/#asterisk ck12 (n=chatzill@141.214.234.28) |
| 16:15.22 | CunningPike | ~AAH |
| 16:15.25 | jbot | somebody said aah was AMP and amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp) |
| 16:15.37 | overshard | CunningPike: A@H = AAH BOTH THE SAME |
| 16:15.39 | esculapio__ | hola quien habla espanol, estoy buscando la forma de conectar mi asterisk ivr base de datos |
| 16:15.49 | overshard | CunningPike, sorry caps |
| 16:16.14 | CunningPike | overshard: No problem - but you need to go to #freepbx for AAH support |
| 16:16.19 | esculapio__ | How to Asterisk IVR on accessing database |
| 16:16.28 | *** join/#asterisk my_name (n=j@nathan.epi.usf.edu) |
| 16:16.45 | overshard | CunningPike, er ok i'll try that then... i figured that since it was a digium card problem it would fall in this catagory not theirs |
| 16:17.13 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 16:17.18 | my_name | is it possible to use asterisk with dialogic cards? |
| 16:17.24 | CunningPike | overshard: It might - but your zaptel.conf is auto-generated, and no-one here likes making manual changes that could make matters worse |
| 16:17.26 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 16:17.45 | boch | esculapio__: yo quizas te pueda ayudar |
| 16:17.46 | overshard | CunningPike, ok then i will try there |
| 16:17.59 | *** part/#asterisk overshard (n=isaac@nc-205-240-45-138.sta.embarqhsd.net) |
| 16:18.25 | esculapio__ | boch, tengo un asterisk y quiero conectarlo a una base de dato para que proporcione autentificacion y |
| 16:19.04 | esculapio__ | boch, busco la forma pero lo q encuentro no me ayuda mucho, |
| 16:19.15 | boch | esculapio__: que funcion usas para autenticar? yo te recomiendo armar algun script AGI |
| 16:19.37 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:20.03 | esculapio__ | boch, pero estoy leyendo sobre eso, pero no doy, lo que quiero es conectarme a la base de dato y que consurte |
| 16:20.47 | boch | que metodo usas para la autenticacion? la haces directamente desde extensions.conf ? |
| 16:21.08 | vivek | . |
| 16:21.33 | *** join/#asterisk pdtoffice (n=ptinsley@209.12.249.243) |
| 16:22.06 | esculapio__ | boch, en este momento estoy trabajando con ello pero no tengo ninguna configuracion para la base de dato, todo esta en la extensions.conf pero sin nada para db |
| 16:22.57 | esculapio__ | boch, estoy un poco perdido, y solo leeo y leo pero no encuentro el camino. |
| 16:23.01 | boch | esculapio__: bueno, lee un poco mas acerca de AGI y implementalo de esa manera, va a ser mucho mas sensillo |
| 16:23.33 | esculapio__ | boch, tengo un agi que realice pero no me funciona |
| 16:23.46 | esculapio__ | boch, si quiere te lo muestro |
| 16:24.21 | esculapio__ | si |
| 16:24.38 | esculapio__ | tengo un correo desde el 96 :) pero no muy frecuente |
| 16:25.04 | esculapio__ | boch, el correo es elvinelintrepidio@hotmail.com |
| 16:25.14 | rob0 | Mi espanol es muy mal |
| 16:26.07 | esculapio__ | ok ya esta |
| 16:27.29 | dlynes_office | rob0: pourquoi est-ce francais? |
| 16:27.43 | rob0 | Merci beaucoup! |
| 16:28.17 | *** join/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 16:28.26 | trixter | does anyone know offhand what an average E1 cost is from BT? |
| 16:28.27 | dlynes_office | good morning, trixter |
| 16:28.32 | trixter | hi |
| 16:30.24 | *** join/#asterisk boch (n=root@201.216.241.97) |
| 16:33.08 | *** join/#asterisk Krush (n=Krush@p5092342B.dip0.t-ipconnect.de) |
| 16:33.12 | Krush | good evening |
| 16:37.05 | dlynes_office | so active in here this morning |
| 16:37.28 | dlynes_office | trixter: btw...people in britland are probably eating supper right now :) |
| 16:44.27 | *** join/#asterisk hi365 (n=FreePBX7@bzq-88-154-14-8.red.bezeqint.net) |
| 16:44.51 | hi365 | Hi! can someone help with a sangoma card? |
| 16:44.58 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:45.05 | Qwell[] | hi365: Hi! can you ask a real question? |
| 16:45.33 | salviadud | lol |
| 16:45.40 | salviadud | yeah, ask a real one |
| 16:45.43 | *** join/#asterisk s0lid (n=s0lid@61.28.161.133) |
| 16:45.43 | salviadud | haha |
| 16:45.59 | syzygybsd | I got one of those cards in a machine I am load testing right now... |
| 16:46.04 | [TK]D-Fender | </Qwell[] mute="on" bile="off"> |
| 16:46.07 | hi365 | ;-) sure. im tring to install a sangoma a200 (4xFxo),i tried to folow the instructions on their wiki, but its not working |
| 16:46.10 | syzygybsd | too bad popcorn hangs up after 1 minute |
| 16:46.15 | Qwell[] | [TK]D-Fender: Did you just end tag me? |
| 16:46.29 | Qwell[] | :p |
| 16:47.27 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 16:47.34 | ph|ber | anyone have an idea why the *8 feature is not working? |
| 16:47.51 | ph|ber | i added this to my features.conf |
| 16:47.53 | ph|ber | pickupexten = *8 |
| 16:48.22 | Qwell[] | ph|ber: Did you add it to the proper dialplan var? |
| 16:48.27 | [TK]D-Fender | ph|ber : Defined your pickup groups in all deviceds? |
| 16:48.45 | hi365 | im tring to install a sangoma a200 (4xFxo),i tried to folow the instructions on their wiki, but its not working |
| 16:49.37 | [TK]D-Fender | hi365 : And please don't spam your informationless complains over and over.... |
| 16:49.45 | [TK]D-Fender | <bile> |
| 16:50.09 | syzygybsd | hmm.. so asterisk can't handle spawning 96 calls in 1 second on this box.... |
| 16:50.14 | ph|ber | yes. it does not work. |
| 16:50.53 | hi365 | <[TK]D-Fender>: just thought maybe it got swoloed up between all the other jiberish |
| 16:51.06 | ph|ber | group=0 |
| 16:51.10 | ph|ber | callgroup=0 |
| 16:51.10 | ph|ber | pickupgroup=0 |
| 16:51.14 | Qwell[] | hi365: swoloed? |
| 16:51.37 | hi365 | swallowed |
| 16:52.01 | Qwell[] | jibberish? like other people getting help? |
| 16:52.38 | [TK]D-Fender | hi365 : How about you actually DESCRIBE whats not working about it. Pastebin CLI output, error messages, etc.... |
| 16:52.41 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 16:54.08 | hi365 | considering this is the only error ive gotten' ill start here: http://pastebin.ca/80612 |
| 16:54.25 | myiagy | i have a h extension that runs deadagi(script.agi).. i had 30 calls running and hung up all of them at the same time.. |
| 16:54.36 | myiagy | eventually some of them give me: |
| 16:54.45 | [TK]D-Fender | ~centosbug |
| 16:54.46 | jbot | it has been said that centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 16:54.55 | [TK]D-Fender | hi365 : Follow the instructions |
| 16:54.56 | myiagy | res_agi.c:1921 run_agi: No channel, no fd? |
| 16:56.09 | *** join/#asterisk xpato (n=pato@114-225-246-201.adsl.terra.cl) |
| 16:56.46 | myiagy | any idea why this happens? for 28 of the channels, the script ran just fine.. |
| 16:57.00 | hi365 | <[TK]D-Fender> did. now where should i continue from? |
| 16:59.02 | *** join/#asterisk knarfly (n=bwatson@12.42.132.26) |
| 16:59.16 | hi365 | im recompiling zap |
| 16:59.24 | ph|ber | any other ideas? |
| 16:59.34 | Dr-Linux | Qwell[]: hi |
| 17:01.08 | xpato | i have this scenarios, an alactel central with 8 lines, behind it is an asterisk ivr with two option: 1. redirect to phones attached to the Alcatel central, and 2. Redirect trought the Alcatel to the PSTN |
| 17:01.12 | xpato | is this possible? |
| 17:01.49 | Dr-Linux | [TK]D-Fender: i have downloaded latest firmware of SCCP for 7935 as .bin but my phone doesn't grab it from the TFTP |
| 17:01.49 | *** part/#asterisk cybergypsy (n=mark@APoitiers-156-1-47-63.w86-213.abo.wanadoo.fr) |
| 17:02.07 | dlynes_office | xpato: does the alcatel central use analog phones, or digital phones? |
| 17:02.53 | Dr-Linux | however everything is good, but still have the same problem i.e. after 2 time offhook, phone doesn't end the call. |
| 17:02.59 | xpato | analog phones, the asterisk has 8 FOX ports connected tho the Alcatel |
| 17:03.34 | dlynes_office | xpato: then, yes |
| 17:04.37 | xpato | dlynes_office: how many concurrents calls could it handle? |
| 17:04.59 | dlynes_office | xpato: obviously eight, because you've only got eight fxo ports hooked up to the alcatel |
| 17:06.50 | [TK]D-Fender | Dr-Linux : No idea. Never really used Cisco's before |
| 17:06.59 | xpato | dlynes_office: to finnish, any links i could read about a similar configuration? |
| 17:07.15 | dlynes_office | what does Finnish have to do with a phone system? |
| 17:07.31 | Qwell[] | dlynes_office: because a Finn created Linux, which asterisk runs on |
| 17:07.34 | Qwell[] | DUH! |
| 17:07.36 | dlynes_office | heh |
| 17:07.55 | Qwell[] | sheesh, get with it :p |
| 17:08.00 | dlynes_office | Qwell[]: Hauschta nappa? |
| 17:08.05 | Qwell[] | ja |
| 17:08.45 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:08.45 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:09.25 | dlynes_office | xpato: I don't know what to recommend to you on that...it's a pretty straight forward setup |
| 17:09.41 | dlynes_office | xpato: nothing special about it |
| 17:09.45 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:09.45 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:09.57 | *** join/#asterisk mosty (n=mostynm@60-241-198-194.static.tpgi.com.au) |
| 17:09.58 | dlynes_office | xpato: Just ignore the fact that there's an alcatel box there |
| 17:10.20 | dlynes_office | xpato: any time you dial a number, the alcatel box will act like a co |
| 17:10.37 | hi365 | <[TK]D-Fender>: i recompiled and tried to restart asterisk but got an this status: asterisk dead but subsys locked here is my config files: http://pastebin.ca/80618 |
| 17:10.55 | mosty | i setup an asterisk box and had it sent to my clients' site, and they're complaining that the volume is too low. is there a way to raise the volume on all channels with asterisk? or is this a client-side phone setting? |
| 17:10.59 | dlynes_office | xpato: so whenever you dial a number on the asterisk box, just send it unmodified to the alcatel, and have the alcatel interpret the number to determine if it's a pstn call, or an extension call |
| 17:13.10 | [TK]D-Fender | hi365 : Pastebin the actual error from the moment you try to execute it. |
| 17:13.35 | dlynes_office | xpato: getting existing alcatel extensions to call asterisk extensions is another matter, however |
| 17:14.09 | justinu|laptop | you need a DID or Tie line configured |
| 17:14.12 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.238.243.Dial1.SanJose1.Level3.net) |
| 17:15.04 | hi365 | <[TK]D-Fender>: i think this is it: http://pastebin.ca/80622 |
| 17:15.09 | Krush | i have got a 8port bernonet card using 4 ports as NT and 4 as TE. We have connected an isdn phone to one of the ports. while bridging is started, the mISDN stacks completly breaks with kernel trace, |
| 17:15.59 | xpato | dlynes_office: there is none extensions on asterisk, the idea is that asterisk acts only as a IVR/Routing |
| 17:16.35 | _problem_ | Krush: i m running beronet card without problem define ur problem in detail..logs etc. |
| 17:16.50 | xpato | the way i think it should work is, define the Alcatel as the "default gateway" for the Asterisk. That should do the trick |
| 17:16.53 | dlynes_office | xpato: ummmm...then...what I was talking about has no relevance...you had told me that the asterisk box was configured as eight analog extensions |
| 17:17.29 | dlynes_office | xpato: wait...one second you're saying asterisk is the gateway; the next second you're saying the alcatel is the gateway...make up your mind |
| 17:17.42 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 17:17.57 | dlynes_office | xpato: which is it? is asterisk connected to your analog phone lines, or is alcatel? |
| 17:17.59 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 17:18.25 | Dr-Linux | [TK]D-Fender: why Wiki or google doesn't have any help for asterisk vs Cisco 7935? :S |
| 17:18.52 | dlynes_office | Dr-Linux: because cisco wants you to use the cisco support channel, #asterisk |
| 17:19.06 | xpato | dlynes_office: the PSTN goes to the Alcatel via an E1 (8 lines) |
| 17:19.18 | xpato | the Alcates has all the phone extentions |
| 17:19.29 | dlynes_office | xpato: and asterisk is connected via eight analog extensions to the alcatel? |
| 17:19.34 | xpato | yes |
| 17:19.56 | dlynes_office | xpato: then see justinu|laptop's response above on getting asterisk extensions to talk to the alcatel if you need that functionality |
| 17:20.34 | dlynes_office | xpato: his solution will work with Nortel MICS for sure, and any other pbx that has did functionality |
| 17:20.35 | xpato | "you need a DID or Tie line configured" ? |
| 17:20.45 | Dr-Linux | dlynes_office: i asked the question in the #cisco but didn't get any answer |
| 17:20.55 | dlynes_office | xpato: yes |
| 17:20.58 | Dr-Linux | dlynes_office: i'm thinking to call Cisco support. |
| 17:21.14 | dlynes_office | Dr-Linux: what a novel idea |
| 17:21.28 | xpato | ok |
| 17:21.35 | xpato | reading about this |
| 17:21.49 | Dr-Linux | novel |
| 17:22.04 | *** join/#asterisk s0lid (n=s0lid@61.28.161.133) |
| 17:22.06 | dlynes_office | ~dict novel |
| 17:22.15 | dlynes_office | stupid jbot |
| 17:22.18 | justinu|laptop | wrong definition |
| 17:22.22 | dlynes_office | that's not the right definition |
| 17:22.32 | dlynes_office | novel == unique |
| 17:22.38 | dlynes_office | or revolutionary |
| 17:22.45 | dlynes_office | groundbreaking |
| 17:22.47 | dlynes_office | earthshattering |
| 17:22.51 | dlynes_office | etc. |
| 17:23.13 | Dr-Linux | dlynes_office: ooooooo ic, you mean "what a fuckin idea" |
| 17:23.17 | Dr-Linux | right? |
| 17:23.22 | justinu|laptop | yes |
| 17:23.23 | dlynes_office | pretty much |
| 17:23.38 | justinu|laptop | cisco will just tell you to try it with call manager |
| 17:23.53 | hi365 | <[TK]D-Fender>: same thing agian http://pastebin.ca/80631 |
| 17:23.56 | dlynes_office | I don't understand why people come on here trying to get support for hardware and software products other than asterisk and digium |
| 17:24.27 | dlynes_office | justinu|laptop: what's call manager? sccp? |
| 17:24.31 | justinu|laptop | yeah |
| 17:24.34 | coppice | I don't understand why people come on here trying to get support for hardware and software products like asterisk and digium |
| 17:24.35 | justinu|laptop | cisco's softswitch product |
| 17:24.36 | Dr-Linux | justinu|laptop: yeah, i know that, they will goaway after hearing asterisk name |
| 17:24.41 | dlynes_office | coppice: heh |
| 17:24.51 | Dr-Linux | bcoz i believe i'm the first one who is using this phone with asterisk :) |
| 17:25.46 | Krush | _problem_, http://pastebin.ca/80634 |
| 17:25.46 | Dr-Linux | justinu|laptop: but i'll ask them how can i upgrade firmware on the phone. |
| 17:25.56 | hi365 | all: asterisk wont start. error: http://pastebin.ca/80631 |
| 17:26.13 | Dr-Linux | as i have already downloaded latest firmware for this phone with my CCO login |
| 17:26.17 | justinu|laptop | Dr-Linux: worth a try, at least |
| 17:26.27 | [TK]D-Fender | hi365 : looks like you didn't modeprobe/ztcfg the card |
| 17:26.57 | hi365 | r u sure my zapata*** files are correct? |
| 17:26.58 | Dr-Linux | justinu|laptop: actually i don't wanna waste this phone. bcoz if it doesn't work with asterisk, then it's useless for me, |
| 17:27.10 | Dr-Linux | justinu|laptop: also i have a cisco 7920 ip phone |
| 17:27.36 | [TK]D-Fender | hi365 : Go check them yourself... you tell me... |
| 17:27.53 | [TK]D-Fender | hi365 : keeping in mind you're running AMP and I shouldn't even be talking to you about this. |
| 17:28.01 | *** join/#asterisk rue_mohr (n=not@bdr2.fieldrd.scrd.ca) |
| 17:28.03 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 17:28.27 | rue_mohr | hello, is there anything that shoud be special for calls that are outgoing fax as opposed to voice? |
| 17:28.33 | phigwork | dr-linux: is the 7920 working? |
| 17:28.55 | hi365 | <[TK]D-Fender> thanks for being _additional kind! can u please tell me how to modeprob? |
| 17:29.20 | rob0 | hi365: spell it right ... "man modprobe" |
| 17:29.21 | Dr-Linux | phigwork: didn't check that one, but that will work easily that's already checked |
| 17:29.28 | Dr-Linux | but 7935 is new |
| 17:29.40 | phigwork | dr-linux: I have a 7910 i am trying to get working |
| 17:29.51 | phigwork | dr-linux: think you can grab the fw for that for me with your login? :) |
| 17:30.01 | [TK]D-Fender | hi365 : depends on your card. What are you using? |
| 17:30.02 | dlynes_office | Dr-Linux: why buy a whole crapload of phones without even checking to see if they're compatible first? |
| 17:30.26 | [TK]D-Fender | dlynes_home : His boss is a moron and "buy first, demand functionality later" |
| 17:30.30 | hi365 | sangoma a200 |
| 17:30.33 | justinu|laptop | lol |
| 17:30.34 | dlynes_office | [TK]D-Fender: heh |
| 17:30.46 | [TK]D-Fender | hi365 : AH. Not even zaptel. |
| 17:30.59 | [TK]D-Fender | hi365 : Did you start up wanrouter? |
| 17:31.09 | Dr-Linux | dlynes_office: not sure, but my boss want to buy eveything for me to play with them. |
| 17:31.13 | hi365 | but dont they use the zaptel drivers? |
| 17:31.20 | Dr-Linux | phigwork: hhm.... |
| 17:31.24 | dlynes_office | Dr-Linux: ah...your boss is the opposite of mine |
| 17:31.27 | [TK]D-Fender | Dr-Linux : Tell him to stop buying things and let YOU advise him. |
| 17:31.34 | dlynes_office | Dr-Linux: I can't get him to spend any money :p |
| 17:31.57 | dlynes_office | hi365: yes, they do |
| 17:31.58 | [TK]D-Fender | hi365 : Go verify your wanrouter status, then start zaptel. |
| 17:32.02 | Dr-Linux | heh |
| 17:32.04 | hi365 | <[TK]D-Fender> yes |
| 17:32.15 | hi365 | its up ans running |
| 17:32.16 | [TK]D-Fender | hi365 (not digium is what I meant to say so some steps don't apply_ |
| 17:32.17 | dlynes_office | hi365: wanrouter start will automatically load zaptel |
| 17:32.20 | *** join/#asterisk adorah (n=Administ@87.69.72.228) |
| 17:32.23 | [TK]D-Fender | hi365 : pastebin some backup.... |
| 17:32.30 | dlynes_office | hi365: assuming you chose the correct configuration options |
| 17:32.33 | rue_mohr | Is there anything that shoud be special for calls that are outgoing fax as opposed to voice? |
| 17:32.50 | dlynes_office | rue_mohr: lots of horseshoes |
| 17:32.56 | dlynes_office | rue_mohr: and lucky charms |
| 17:33.01 | rue_mohr | hmm |
| 17:33.07 | rue_mohr | lucky charms eh? |
| 17:33.13 | rue_mohr | were not running compression |
| 17:33.13 | hi365 | <[TK]D-Fender> : http://pastebin.ca/80643 |
| 17:33.31 | rue_mohr | idea |
| 17:34.01 | xpato | dlynes_office: are another way without DID or Tie Lines? |
| 17:34.53 | syzygybsd | is there a good document anywhere showing how much CPU power you need for how many zap channels? call spawns per x seconds etc? |
| 17:35.00 | dlynes_office | You only need the dids and/or tie lines if you want alcatel extensions to be able to call asterisk extensions, unless your analog lines on your asterisk box have 1:1 mappings with asterisk extensions |
| 17:35.03 | [TK]D-Fender | hi365 : ok, run "ztcfg -vvvv" |
| 17:35.25 | [TK]D-Fender | hi365 : if everything looks alright then star "asterisk -gvvvvc" |
| 17:36.00 | dlynes_office | xpato: You only need the dids and/or tie lines if you want alcatel extensions to be able to call asterisk extensions, unless your analog lines on your asterisk box have 1:1 mappings with asterisk extensions |
| 17:36.40 | *** join/#asterisk Toerkeium (n=oo@201.216.206.221) |
| 17:36.42 | Toerkeium | hello guys |
| 17:37.18 | Toerkeium | Do I need a customized kernel to install asterisk ? I ask this because I was wondering if I can start my tests installing asterisk in a Linux VPS |
| 17:38.08 | phigwork | sounds like no |
| 17:38.28 | *** join/#asterisk linlin (n=linlin@c-67-184-159-30.hsd1.il.comcast.net) |
| 17:38.30 | Toerkeium | excellent, gonna try it on a vps |
| 17:38.55 | hi365 | <[TK]D-Fender> LOVE U MAN!!! |
| 17:39.04 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 17:39.07 | hi365 | thanks a ton! + _aditional!!!! |
| 17:40.56 | xpato | dlynes_office: the aren't asterisk extensions, only alcatel extensions, all the call from the PSTN gets to the ALCATEL and pass directly to the Asterisk, in the Asterisk the idea is if they press one, tha call goes to an Alcatel phone extension, if they press 2 the call has to goe to a phone over the PSTN |
| 17:41.56 | [TK]D-Fender | hi365 : ywc |
| 17:41.58 | Toerkeium | probably a stupid question: I have some (4) sip line. And I want to install asterisk to manage my IVR settings and control this lines. No need for hardware there right? |
| 17:42.41 | [TK]D-Fender | Toerkeium : For pure VoIP w/o MeetMe conferences you don't need any special hardware or kernel compiling. |
| 17:42.50 | dlynes_office | xpato: ah...yeah...anyways...it should work just fine |
| 17:43.04 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
| 17:43.18 | r_evolution | how many of us have been.... |
| 17:43.19 | dlynes_office | xpato: Just treat it like an asterisk system with 8 phone lines |
| 17:43.20 | r_evolution | friends. |
| 17:43.21 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 17:43.23 | r_evolution | ones you can depend on... |
| 17:43.24 | r_evolution | friends. |
| 17:43.30 | justinu|laptop | r_evolution: ! |
| 17:43.35 | r_evolution | justin: ! |
| 17:43.35 | dlynes_office | xpato: just that some phone numbers will be the same length as alcatel extension numbers |
| 17:43.40 | r_evolution | i have a new toy to play with justin :-D |
| 17:43.45 | r_evolution | it's annoying me tho :-\ |
| 17:43.47 | r_evolution | small ATA |
| 17:43.52 | r_evolution | keeps throwing a Got SIP response 400 "Bad Request" back from |
| 17:43.54 | r_evolution | error |
| 17:43.55 | r_evolution | rar. |
| 17:43.57 | dlynes_office | xpato: and outbound calls need a '9' in front of them if that's how your alcatel is set up |
| 17:44.21 | Toerkeium | thansk [TK]D-Fender, what do you mean by "w/o MeetMe" ? |
| 17:44.40 | justinu|laptop | r_evolution: pastebin the dialog |
| 17:44.49 | r_evolution | no dialog to it homie... |
| 17:44.50 | r_evolution | just randomly |
| 17:44.51 | r_evolution | Got SIP response 400 "Bad Request" back from |
| 17:44.57 | r_evolution | Got SIP response 400 "Bad Request" back from <IP HERE> |
| 17:45.07 | justinu|laptop | well, you gotta turn on sip debug to see what it doesn't like |
| 17:45.12 | justinu|laptop | could be options ping |
| 17:45.20 | r_evolution | yeah i know... but that also means i have to register it to a different server ;) |
| 17:45.21 | [TK]D-Fender | Toerkeium : MeetMe is the * conferencing app. |
| 17:45.41 | Toerkeium | thanks a lot,gonna try it just right now |
| 17:45.45 | r_evolution | SOOO give a fellow a moment |
| 17:45.55 | r_evolution | hey toerk... make sure you loaded ztdummy |
| 17:46.05 | Toerkeium | what's that r_evolution? |
| 17:46.07 | r_evolution | if you dont have a zaptel module already loaded :) |
| 17:46.12 | [TK]D-Fender | r_evolution : In a VPS I doubt he'll have access.... |
| 17:46.17 | r_evolution | ooooo |
| 17:46.23 | Toerkeium | no access to touch kernel |
| 17:46.29 | [TK]D-Fender | r_evolution : And give a fellow a moment, and he'll take an eternity... |
| 17:46.30 | r_evolution | :( |
| 17:46.36 | r_evolution | hahaha @ TK |
| 17:46.36 | Toerkeium | I just need to manage my SIP lines as I want, with my own IVR |
| 17:46.49 | r_evolution | in that case listen to TK... but make sure you call him a damn canadian every so often ;) |
| 17:47.23 | Toerkeium | why you say such thing ? :) |
| 17:47.49 | Toerkeium | based on your exprience.. from 1 to 10, how complex is asterisk to make it work? |
| 17:47.51 | justinu|laptop | terrance and phillip are super sweet |
| 17:47.54 | r_evolution | i calls em like i see's em |
| 17:48.01 | docelmo | OI! |
| 17:48.16 | r_evolution | docel! ;) |
| 17:48.20 | mishehu | justinu|laptop: fart fart fart |
| 17:48.22 | mut | arrrrr |
| 17:48.23 | Toerkeium | docel? |
| 17:48.23 | docelmo | whadup! |
| 17:48.31 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
| 17:48.46 | phigwork | I keep getting these over and over again |
| 17:48.47 | Toerkeium | any good how-to for a beginner? |
| 17:48.48 | phigwork | Jul 6 10:03:00 WARNING[4338]: chan_sip.c:9770 handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 17:48.52 | phigwork | what's that mean? |
| 17:49.11 | [TK]D-Fender | Toerkeium : here.... |
| 17:49.12 | [TK]D-Fender | ~book |
| 17:49.19 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:49.19 | [TK]D-Fender | ~docs |
| 17:49.20 | jbot | hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:49.25 | Toerkeium | cool |
| 17:49.37 | r_evolution | there are many good how-tos :) |
| 17:49.49 | Toerkeium | excellent, thanks a lot gays |
| 17:49.52 | Toerkeium | I mean, guys |
| 17:49.54 | r_evolution | er. |
| 17:49.55 | justinu|laptop | lol |
| 17:49.57 | Toerkeium | heh |
| 17:50.01 | r_evolution | <---likes tits and clits |
| 17:50.06 | justinu|laptop | freudian slip |
| 17:50.21 | [TK]D-Fender | justinu|laptop: Always in fashion..... |
| 17:50.28 | r_evolution | yeah but is he thinking we're gay... or hoping? |
| 17:50.34 | mishehu | freudian slit? |
| 17:50.35 | Toerkeium | hahahaha |
| 17:50.43 | justinu|laptop | r_evolution: you never can tell these days |
| 17:51.20 | mishehu | r_evolution: becareful that it's not like the shower wall in Porky's. |
| 17:51.31 | r_evolution | no glory holes here :) |
| 17:52.36 | Dr-Linux | r_evolution: i guess you have been working on cisco 7935 ip phone? :P |
| 17:52.51 | r_evolution | nah... it's a random ATA... from like Askey Computer Corp? |
| 17:52.58 | r_evolution | some Taiwan guys |
| 17:53.11 | mishehu | AssKey Computer Corp? |
| 17:53.14 | r_evolution | you got a suggestion? Couplea things seem to suggest an issue with the auth header |
| 17:53.16 | r_evolution | yessss |
| 17:53.17 | r_evolution | asskey |
| 17:53.26 | r_evolution | i.e. what happens when you get drunk and sit on your keyboard |
| 17:53.27 | Dr-Linux | ass key |
| 17:53.43 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 17:53.52 | r_evolution | either that or hella gay baby talk |
| 17:54.04 | r_evolution | does widdle mishey-wishy want to askey me a question? |
| 17:54.06 | Dr-Linux | <r_evolution> i.e. what happens when you get drunk and sit on your keyboard >> ofcos, keyboard will broke up! |
| 17:54.19 | r_evolution | yep! and you'll have a key in your ass |
| 17:54.21 | r_evolution | thus |
| 17:54.22 | r_evolution | an asskey |
| 17:56.16 | dlynes_office | ummm |
| 17:56.23 | my_name | hello what is the diff between exten => 2,1,Dial,Zap/1 and exten => 2,1,Dial(Zap/1) |
| 17:56.31 | dlynes_office | when i went afk, did this suddenly become #assterisk? |
| 17:56.31 | justinu|laptop | nothing |
| 17:57.27 | r_evolution | yes |
| 17:57.28 | r_evolution | yes it did |
| 17:57.40 | r_evolution | fuck *... we talk about bitches! |
| 17:57.43 | r_evolution | ;) |
| 17:58.46 | Dr-Linux | question, does this effect on voice quality of the calls goes via 3 asterisk trunks instead of direct call? |
| 17:59.25 | dlynes_office | Dr-Linux: is there any transcoding happening? |
| 17:59.35 | justinu|laptop | the more hops, the more likely quality will suffer |
| 18:00.13 | Dr-Linux | dlynes_office: all the hops have G729 |
| 18:00.19 | Dr-Linux | and trunked with SIP |
| 18:00.25 | dlynes_office | Dr-Linux: are they all in close proximity? |
| 18:00.55 | Dr-Linux | dlynes_home: they are in different areas |
| 18:01.07 | dlynes_office | you didn't answer my question |
| 18:01.12 | dlynes_office | are they in close proximity? |
| 18:01.28 | Dr-Linux | dlynes_office: sorry friend, i didn't understand your question |
| 18:01.38 | Dr-Linux | ~dict proximity |
| 18:01.44 | dlynes_office | i.e. are they all close, network wise? |
| 18:02.10 | dlynes_office | they can be 1000km away, but still be close, when it comes to the network |
| 18:02.11 | Dr-Linux | dlynes_office: i think my answer was right, they are in different countries |
| 18:02.14 | dlynes_office | ah |
| 18:02.23 | dlynes_office | then yeah, your quality will definitely suffer |
| 18:02.24 | Dr-Linux | 2 in US |
| 18:02.32 | Dr-Linux | but have different stats |
| 18:02.38 | Dr-Linux | states* |
| 18:03.02 | dlynes_office | if they're close in proximity, and not many hops between them, the more likely quality will suffer, as justinu|laptop stated |
| 18:03.32 | dlynes_office | because with the more servers relaying the request, the more likely you'll get latency |
| 18:04.19 | Dr-Linux | dlynes_office: actually my case is something like: i have a support repr. he provide support for 2 different servers, mean recieve calls from two different product and different servers |
| 18:04.29 | *** join/#asterisk implicit (n=implicit@ip68-4-84-39.oc.oc.cox.net) |
| 18:04.33 | Dr-Linux | dlynes_office: so that repr. registered with 2 different asterisk server |
| 18:04.56 | Dr-Linux | dlynes_office: i was thinking if i registered him with only one asterisk server .. and |
| 18:05.28 | Dr-Linux | second asterisk queue route to the server he is registered with |
| 18:05.29 | phigwork | dr-lin: that was a no? :) |
| 18:05.32 | *** join/#asterisk firejon (n=firejon@206-169-48-226.static.twtelecom.net) |
| 18:06.56 | dlynes_office | Dr-Linux: it doesn't matter how you do that, your call quality is going to suffer |
| 18:06.59 | Dr-Linux | phigwork: what you need? |
| 18:07.05 | phigwork | the fw for 7910 |
| 18:07.28 | dlynes_office | phigwork: try #ciscowarez |
| 18:07.44 | Dr-Linux | phigwork: i'll give you tomorrow, |
| 18:07.54 | Dr-Linux | phigwork: but i was treated bad in this case |
| 18:08.06 | phigwork | bad? |
| 18:08.30 | Dr-Linux | when i had no CCO login, i asked many guys here, but no one felt to give me any firmware.. |
| 18:08.45 | Dr-Linux | phigwork: but i'll give you tomorrow |
| 18:09.00 | Dr-Linux | right now i'm not on my work system |
| 18:09.00 | phigwork | ah, so now you gotta pay em back by doin it to someone else :) |
| 18:09.07 | phigwork | and the world was right again |
| 18:09.10 | *** join/#asterisk s0lid (n=s0lid@61.28.161.133) |
| 18:09.13 | phigwork | cool |
| 18:10.30 | Dr-Linux | phigwork: not really, but i thought it consider bad in US/CA/Europ to give someone those kind of stuff |
| 18:10.49 | *** join/#asterisk TeePOG (n=1234@dsl-145-151-26.telkomadsl.co.za) |
| 18:11.19 | dlynes_office | Dr-Linux: some people like phigwork don't mind using illegal software though :) |
| 18:11.43 | Dr-Linux | dlynes_office: do you mind? |
| 18:11.44 | justinu|laptop | i thought the term was pay it forward |
| 18:11.53 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 18:12.06 | dlynes_office | Dr-Linux: well, i mind because i'm software developer |
| 18:12.12 | dlynes_office | Dr-Linux: that's how i earn my living |
| 18:12.37 | phigwork | illegal? |
| 18:12.40 | phigwork | do they sell it? |
| 18:12.47 | dlynes_office | phigwork: yes, they do |
| 18:13.04 | Dr-Linux | dlynes_office: well, but in my area 80% peoples can't afford to buy any software.. |
| 18:13.05 | dlynes_office | phigwork: you buy a support plan from them, and the firmware upgrades come with the support plan |
| 18:13.12 | dlynes_office | Dr-Linux: yeah, i'm aware of that |
| 18:13.27 | dlynes_office | Dr-Linux: I don't condone it, but i can understand it in certain situations |
| 18:13.29 | phigwork | so I can call up cisco and say "Hi, I'm some random Joe Shmoe, I want to purchase -just- the latest firmware for the 7910" and they'll be like "Sure thing, that'll be $19.95." |
| 18:13.30 | phigwork | ? |
| 18:13.41 | Dr-Linux | dlynes_office: you guys sell your software according to your countries money, but never think, that what we peoples will do to buy them |
| 18:13.50 | Nugget | phigwork: in a manner of speaking, yes. you buy the support contract. |
| 18:13.54 | rene- | hey, i am trying to distribute calls evenly among 4 sip devices,, i am trying to do it in the dial plan like this: http://pastebin.ca/80671 |
| 18:13.57 | phigwork | what do I need a support plan for, I'm some random joe shmoe, not a corporation |
| 18:14.04 | Nugget | so you can get the firmware. |
| 18:14.05 | dlynes_office | phigwork: no...you say i'm some random joe schmoe and i want to buy a yearly support contract, and they say sure thing, that'll be $250,000USD |
| 18:14.07 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 18:14.08 | rene- | anyone want to take a look at it? |
| 18:14.22 | rene- | i am kindof loosing calls |
| 18:14.37 | phigwork | dlynes: that makes a lot of sense for someone that make $0 a year to pay $250,000 a year so he can use his ip phone |
| 18:14.38 | dlynes_office | and that's precisely why i don't use cisco :p |
| 18:14.42 | Dr-Linux | phigwork: i have support contract with Cisco |
| 18:14.43 | Nugget | the support contract for a cisco phone is $10 a year. |
| 18:14.43 | phigwork | makes |
| 18:14.44 | rene- | but i dont know if thats due my devices loosing registration or my dial plan |
| 18:14.53 | phigwork | Dr-Linux: you can probably use it |
| 18:15.11 | Nugget | it's hard to find a vendor who will sell you just one, but it can be done. lots of us have. |
| 18:15.26 | dlynes_office | phigwork: i was exaggerating, based on the amount I've seen some people say is their yearly support contract priciing |
| 18:15.37 | Dr-Linux | well, we have many cisco devices |
| 18:15.40 | Nugget | the alternative is to not buy a fucking cisco phone and use a different type that doesn't have this requirement. |
| 18:15.48 | Nugget | the alternative is not to just steal the firmware |
| 18:15.55 | phigwork | Cisco phone was free |
| 18:15.57 | Juggie | Nugget, cisco phones suck |
| 18:16.00 | Juggie | find something else |
| 18:16.01 | phigwork | Another phone is $120+ |
| 18:16.07 | Nugget | all the more reason to use a different type |
| 18:16.11 | phigwork | hmmm free... or $120+.... hmmmm |
| 18:16.23 | *** join/#asterisk XARiUS (n=bdarcy@66-146-191-242.skyriver.net) |
| 18:16.27 | Qwell[] | It's only free if your time is worth nothing. |
| 18:16.44 | phigwork | I'm a student. That's how I roll. |
| 18:16.51 | XARiUS | hey guys, anyone with IP501's.. does the soundpointipwelcome.wav play on the 501's, or only the higher models?? |
| 18:17.07 | justinu|laptop | it plays |
| 18:17.17 | XARiUS | hrm.. damned if I've yet to hear it lol |
| 18:17.17 | dlynes_office | Qwell[]: maybe you could convince phigwork to write the sccp support for the 7910 :) |
| 18:17.30 | Qwell[] | chan_skinny already supports 7910 :p |
| 18:17.41 | Qwell[] | somebody was kind enough to give me one, to ensure that |
| 18:17.45 | Dr-Linux | phigwork: i'll give you .. i think, i learnt alot stuff from you guys .. i can't pay anything back to you gusy, but yes that i can do |
| 18:17.47 | dlynes_office | ah...then why the hell is he screwing around with sip firmware? |
| 18:17.51 | Qwell[] | got me |
| 18:18.12 | phigwork | 7910 isn't sip |
| 18:18.20 | phigwork | it needs to tftp boot |
| 18:18.27 | dlynes_office | phigwork: so what's wrong with chan_skinny? |
| 18:18.28 | Dr-Linux | 7920 is not a sip |
| 18:18.30 | Juggie | sip tftp boots too |
| 18:18.42 | Dr-Linux | neither 7935/7936 |
| 18:18.54 | Qwell[] | Dr-Linux: You've gotta send me one of those, to get working :p |
| 18:18.56 | phigwork | dlynes_office: nothing, as long as I can feed the phone the right scripts and firmware, which i don't have |
| 18:18.57 | Juggie | just get a mitel 5215 and be happy! :) |
| 18:18.59 | Juggie | no more retardedness. |
| 18:19.04 | Dr-Linux | chan_skinny doesn't work .. but SCCP does good, |
| 18:19.11 | Qwell[] | Dr-Linux: see above ;) |
| 18:19.15 | Dr-Linux | but asterisk do not have SCCP module |
| 18:19.25 | dlynes_office | Juggie: i'd rather go with the Nortel i2007 : |
| 18:19.33 | Qwell[] | chan_sccp won't work when 1.4 is released... |
| 18:19.36 | Qwell[] | yay Sergio |
| 18:19.46 | dlynes_office | heh |
| 18:19.47 | Dr-Linux | Qwell[]: send you what? |
| 18:19.52 | Qwell[] | I could fix it for him, but he's a dick, so...NAH |
| 18:19.57 | Qwell[] | Dr-Linux: a 7935/7936 ;) |
| 18:20.13 | Qwell[] | dlynes_office: latest release was in April. That makes me so happy :p |
| 18:20.44 | dlynes_office | why would it make you happy that he stopped development on chan_sccp? |
| 18:20.46 | Juggie | get yourself this phone -> http://www.mitel.com/DocController?documentId=17406&c=9512&sc=9517 and be happy! :) |
| 18:20.48 | Juggie | no more lameness. |
| 18:20.55 | Dr-Linux | Qwell[]: i have only one 7935 and that works fine for me, but have a little problem, after 3 times offhook it doesn't provide me "endcall" option on the screen. |
| 18:20.57 | Qwell[] | dlynes_office: because he's a dumbass. :) |
| 18:21.03 | dlynes_office | or because you figure he thinks that your channel driver's gone far enough that his is irrelevant now? |
| 18:21.05 | Qwell[] | Dr-Linux: yeah, that's a bug in chan_sccp |
| 18:21.10 | Qwell[] | dlynes_office: nah |
| 18:21.18 | Juggie | whats the difference in SCCP/SKINNY? |
| 18:21.21 | Qwell[] | dlynes_office: he just...stopped caring, after I called him out on a few things |
| 18:21.25 | Qwell[] | Juggie: absolutely nothing |
| 18:21.26 | dlynes_office | Juggie: nothing |
| 18:21.30 | dlynes_office | Juggie: only authors |
| 18:21.34 | Qwell[] | skinny client control protocol |
| 18:21.35 | Juggie | i guess we dont need both then eh :) |
| 18:21.43 | Qwell[] | Juggie: most definitely not |
| 18:21.49 | Juggie | which one is in the tree? |
| 18:21.54 | Qwell[] | skinny |
| 18:21.57 | dlynes_office | Juggie: Qwell's (skinny) |
| 18:21.58 | Dr-Linux | Qwell[]: how can i fix that bug? |
| 18:22.07 | Dr-Linux | Qwell[]: is there any patch or something? |
| 18:22.09 | Qwell[] | Dr-Linux: by using chan_skinny...and sending me a phone :p |
| 18:22.12 | XARiUS | justinu: at what point *should* it play, once the phone has completed loading and is usable? |
| 18:22.22 | MikeJ[Laptop] | Qwell is skinny? |
| 18:22.29 | Qwell[] | Dr-Linux: I couldn't tell ya...I can't look at his code |
| 18:22.30 | Juggie | Qwell, you wanna impress me make a chan_mitel :) |
| 18:22.32 | Qwell[] | MikeJ[Laptop]: heh, ass |
| 18:22.33 | dlynes_office | MikeJ[Laptop]: no...he's a fat cow |
| 18:22.37 | Dr-Linux | Qwell[]: but you are the one who advised me to compile chan_sccp and use it??? |
| 18:22.44 | justinu|laptop | XARiUS: yeah... i've noticed it won't play when the phone "can't contact boot server" |
| 18:22.47 | Qwell[] | Dr-Linux: yeah...because it "works" |
| 18:23.03 | dlynes_office | Dr-Linux: the thing about opensource is that unless you support it, it doesn't work |
| 18:23.04 | Qwell[] | Dr-Linux: Did you send a message to the list? They'll have an answer for you |
| 18:23.08 | XARiUS | justinu: should work fine over ftp mass deployment setup yeah? All the phones load their configs properly. |
| 18:23.17 | justinu|laptop | XARiUS: it should yeah |
| 18:23.37 | XARiUS | justinu|laptop: okie thanks, I'll just keep dicking with it... cool feature, wanna put in something custom for my clients when I deliver the phones :) |
| 18:23.44 | dlynes_office | Dr-Linux: in order for opensource authors to support various expensive hardware, they need someone to either donate that hardware to them, or donate some cash to them so they can afford to buy the hardware |
| 18:23.45 | Dr-Linux | Qwell[]: i have sent 3 messages on the post, but they havn't got any answer from them |
| 18:23.53 | my_name | is there a tutorial on how to setup channel for a dialogic d/41jct-ls card |
| 18:23.59 | Qwell[] | Dr-Linux: I haven't seen any...are you sure it went through? |
| 18:24.22 | Qwell[] | dlynes_office: s/donate/loan/ |
| 18:24.31 | dlynes_office | or loan, yeah |
| 18:24.43 | dlynes_office | and get them to pay return freight as well :) |
| 18:24.46 | Dr-Linux | Qwell[]: where can i forward that mail to you? |
| 18:25.05 | Qwell[] | Dr-Linux: if it was sent to the list, it would have gotten to me |
| 18:25.09 | Qwell[] | dlynes_office: indeed |
| 18:25.20 | Qwell[] | dlynes_office: somebody loaned me a 7920 recently...got it working in fairly short order |
| 18:25.32 | dlynes_office | yeah...I remember |
| 18:25.41 | firejon | does anyone know if the h extension works properly for the hangup event when using queues & AgentCallbackLogin? |
| 18:25.53 | *** join/#asterisk Tako-san (n=Tako-san@24.108.162.254) |
| 18:26.08 | Qwell[] | I need to get my bosses 7970 back here to test...heh |
| 18:26.38 | dlynes_office | man |
| 18:26.40 | dlynes_office | wtf? |
| 18:26.51 | Dr-Linux | Qwell[]: let me send send again at chan-sccp-users@lists.berlios.de |
| 18:26.58 | Qwell[] | Dr-Linux: did you subscribe? |
| 18:27.04 | dlynes_office | my phones won't upgrade their firmware on my home network, but they will on my office network |
| 18:27.06 | Dr-Linux | yes |
| 18:27.20 | dlynes_office | and they don't grab dhcp on a hub for whatever reason; only on a switch |
| 18:27.21 | Qwell[] | Dr-Linux: are you getting your own messages? |
| 18:27.40 | justinu|laptop | XARiUS: is your dhcp server sending the phones a "tftp-boot-server" option? |
| 18:28.05 | Dr-Linux | Qwell[]: no |
| 18:28.14 | Qwell[] | Dr-Linux: sounds like a problem...heh |
| 18:28.17 | Juggie | dlynes_home, i have a problem with them working across subnets... they dont seem in my case to want to work when the tftp server is on a different subnet. |
| 18:28.23 | Juggie | i gave up on them all and junked them. |
| 18:28.28 | Dr-Linux | Qwell[]: let me send again |
| 18:28.28 | Juggie | i have one polycom and all mitel phones now. |
| 18:28.59 | Juggie | mitel phones do config/firmware over tftp/http |
| 18:29.17 | XARiUS | justinu: nope, I'm just setting the ftp info up manually, only 5 phones, going in a very small office with no server, hosted pbx solution. |
| 18:29.22 | Juggie | so using http and mod rewrite you can generate config files on the fly. |
| 18:29.52 | XARiUS | my server does host the FTP, however, remote to the phones. |
| 18:30.03 | Dr-Linux | Qwell[]: what was that? your pvt |
| 18:30.04 | justinu|laptop | XARiUS: i think you need that option... before custom ringtones, or startup sounds will play |
| 18:30.09 | justinu|laptop | XARiUS: at least in my experience |
| 18:30.15 | XARiUS | ahh. |
| 18:30.19 | Qwell[] | Dr-Linux: a welcome message |
| 18:30.20 | XARiUS | that would explain it then. |
| 18:30.44 | justinu|laptop | XARiUS: maybe a WRT54GL running openwrt or dd-wrt |
| 18:30.51 | Dr-Linux | Qwell[]: does it mean, it's done? and i'm able to request |
| 18:30.52 | justinu|laptop | could be your custom DHCP server |
| 18:30.57 | Qwell[] | Dr-Linux: try it.. |
| 18:31.07 | Dr-Linux | ok |
| 18:31.08 | jbalcomb | Anyone know what the 'tftp-hpa' package is? is it better/different than just 'tftp'? |
| 18:31.09 | phigwork | Hm, this is weird. In the verbose console, I see that * is dialing Zap/1/18008887600 |
| 18:31.10 | Qwell[] | (again..) |
| 18:31.22 | phigwork | but i get the message "Remember, y ou must first dial a 1 when .." blah blah |
| 18:31.28 | phigwork | whats up with that |
| 18:31.32 | justinu|laptop | insert a pause |
| 18:31.37 | justinu|laptop | dial w18008887600 |
| 18:31.43 | phigwork | ah cool |
| 18:31.49 | phigwork | i was told there wasn't a pause character :) |
| 18:31.50 | XARiUS | justinu: yeah, I'd been running older wrt54g's with sveasoft.. never got the dang tftp boot options working on those though. |
| 18:31.59 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 18:32.13 | jbalcomb | My TFTP server is not producing any log messages, my polycom says it can't access the boot server (my gxp-2000 can), and when I try to u/l a file it says 'not found' |
| 18:32.13 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 18:32.18 | Qwell[] | jbalcomb: -vv |
| 18:32.22 | phigwork | justinu: thanks |
| 18:33.34 | syzygybsd | what are the most limiting factors on sound quality on an asterisk box. i notice lack of traffic on zap when I hit 35 calls, but static when I hit blocks more then 5. Are there any other inticators that I sould watch to see issues |
| 18:34.16 | stack_ | OT: I have some Polycom 301 & 501 phones. If you accidentally dial 911 (which is pretty easy to do), the phone automatically dials out, without having to press send. I can't find a way to disable this. Any ideas? |
| 18:34.32 | tzanger | stack_: it's in the phone's dialplan |
| 18:34.39 | justinu|laptop | redirect 911 to an IVR that asks them if they really want to dial 911, stay on the line |
| 18:34.46 | justinu|laptop | otherwise, hangup now |
| 18:34.48 | CunningPike | stack_: THere is an emergency contact setup somewhere - search sip.cfg for 'emergency' |
| 18:35.16 | tzanger | yeah I do that in my * dialplan... playback(youre-gonna-call-911-in-5-seconds) or somesuch |
| 18:36.42 | jbalcomb | Qwell[]: its already -vvvvv |
| 18:37.37 | stack_ | CunningPike: my value line for emergency is empty... |
| 18:38.17 | CunningPike | stack_: Then it is your dialplan - we also Playback() a sound file when people dial 911 - it's a good idea....... |
| 18:38.33 | jbalcomb | tftpd-hpa OPTIONS="-c -l -vvvvv -u nobody -s /asterisk/tftp" |
| 18:38.36 | stack_ | ok, I'll go with that then |
| 18:38.43 | stack_ | thanks |
| 18:39.44 | CunningPike | stack_: The reason we put the recording in in the first place was because of a delay when our SPA-3000s place the call - we played the recording so that people _wouldn't_ hangup - but it has the additional benefit of allowing _to_ hangup if they dial by mistake |
| 18:39.50 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 18:40.47 | Dr-Linux | Qwell[]: i just sent an email to chan-sccp-users@lists.berlios.de |
| 18:41.11 | trixter | does anyone know offhand what an average E1 cost is from BT? |
| 18:43.35 | Qwell[] | Dr-Linux: there it is |
| 18:44.13 | [TK]D-Fender | jbalcomb : setting Option 66 for tftp and told Polycom that was the method to use? By default its FTP IIRC. |
| 18:46.46 | jbalcomb | [TK]D-Fender: Hrmm.. I was hoping to be able to get the polycom to set itself up once i plugged it in.. |
| 18:46.53 | jbalcomb | [TK]D-Fender: I do have option 66 set though |
| 18:46.59 | *** part/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 18:47.02 | [TK]D-Fender | jbalcomb : vERIFY ITS PROVISIONING METHOD IN THE BOOT MENU |
| 18:47.14 | jbalcomb | [TK]D-Fender: Any chance theres a vendor specific option to tell the phone which to use? |
| 18:47.34 | justinu|laptop | also, verify capslock is not set :P |
| 18:47.45 | jbalcomb | [TK]D-Fender: feck. Boot Server Type: FTP |
| 18:47.48 | r_evolution | pwnd. |
| 18:47.50 | [TK]D-Fender | jbalcomb : there may be a DCHP flag for that.... |
| 18:47.50 | r_evolution | pwnd. |
| 18:47.51 | r_evolution | just pwnd. |
| 18:48.18 | r_evolution | TK = polycom crusader |
| 18:48.31 | jbalcomb | [TK]D-Fender: Welp, I'm not sure it matters. I'd be just as happy if it auto-provisioned from FTP as TFTP |
| 18:48.39 | [TK]D-Fender | r_evolution : So far they have succeeded in "not royally sucking like the rest" |
| 18:48.50 | r_evolution | fuck im glad you reminded me of that jbal... i gotta check for provis on this new ATA |
| 18:49.02 | r_evolution | yeah... i already told you the next hard-phones we got were going to be polycom |
| 18:49.05 | *** join/#asterisk s0lid (n=s0lid@124.6.176.98) |
| 18:49.14 | r_evolution | after the whole sipura-941 cant make a 3-way using 729 |
| 18:49.16 | r_evolution | thing |
| 18:49.45 | *** join/#asterisk cron (n=cron@blk-224-156-15.eastlink.ca) |
| 18:51.17 | [TK]D-Fender | r_evolution : I don't know for certain that Polycom CAN.... |
| 18:51.23 | [TK]D-Fender | r_evolution : worth testing. |
| 18:51.43 | [TK]D-Fender | r_evolution : IIRC Cisco's dont (/didn't?) |
| 18:52.00 | r_evolution | teh sux. |
| 18:52.03 | r_evolution | the thing is... |
| 18:52.07 | r_evolution | the Sipura ATAs CAN! |
| 18:52.13 | r_evolution | the $60 ATA can do it |
| 18:52.17 | r_evolution | but the $120 hardphone cant |
| 18:52.20 | r_evolution | wtf is THAT about? |
| 18:52.25 | *** join/#asterisk asterisk_newbie (n=a_ti_tu_@bl7-130-1.dsl.telepac.pt) |
| 18:52.54 | asterisk_newbie | hi all |
| 18:52.58 | asterisk_newbie | hi l-fy |
| 18:53.11 | asterisk_newbie | greetings from Portugal :) |
| 18:53.30 | *** join/#asterisk JonR800 (i=jon@p1mp.org) |
| 18:54.14 | [TK]D-Fender | r_evolution : not that I've seen... |
| 18:54.26 | *** join/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 18:54.28 | r_evolution | they can |
| 18:54.33 | r_evolution | we use the SPA-2100s here |
| 18:54.38 | [TK]D-Fender | r_evolution : What models have you tested? |
| 18:54.50 | jbalcomb | How about the Linksys 942? Looks really nice in the pictures. |
| 18:54.58 | [TK]D-Fender | r_evolution : Those have a bigger processor IIRC.... thats probably why. The lower ones were too wimpy for it. |
| 18:54.58 | Egonis | I added a music on hold class, but am getting 'no class: blah' in the asterisk console when it tries to start moh for that class, am I missing a step? |
| 18:54.59 | r_evolution | ^ |
| 18:55.18 | r_evolution | i think they MAY have tested a 2002... but i know the 2100 works |
| 18:55.39 | [TK]D-Fender | jbalcomb : ... ICK. Still a funny feel, no presence, and so many other flaws. Only thing on its side is the backlight. |
| 18:56.11 | *** join/#asterisk roche (n=roche@200.122.154.250) |
| 18:57.39 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 18:57.58 | CunningPike | jbalcomb: FTP is preferable over TFTP for Polycoms for a variety of reasons..... |
| 18:58.06 | jbalcomb | CunningPike: such as? |
| 18:58.15 | jbalcomb | [TK]D-Fender gotcha |
| 18:59.02 | CunningPike | jbalcomb: Log file uploads, auto-polling for updates, exception and directory storage....... |
| 18:59.15 | Egonis | Do I need to 'sox' an mp3 before using it for music on hold |
| 18:59.16 | Egonis | ? |
| 18:59.22 | jbalcomb | CunningPike: running tftp with -c allows all that to happen |
| 18:59.46 | CunningPike | jbalcomb: WHich - all of it? |
| 19:00.08 | [TK]D-Fender | Egonis : pASTEBIN YOUR MUSICONHOLD.CONF |
| 19:00.09 | jbalcomb | CunningPike my quandry at this point would be handing out an ftp address just for the polycoms while still giving the gxp2000 the tftp adress via dhcp |
| 19:00.10 | [TK]D-Fender | ~pb |
| 19:00.12 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 19:00.19 | jbalcomb | CunningPike: stuff that invlolves uploading |
| 19:00.42 | *** part/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 19:00.55 | CunningPike | jbalcomb: Ah, I see your quandry - PeeStream still doesn't do FTP provisioning, eh? |
| 19:00.59 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 19:01.44 | jbalcomb | CunningPike: no, ftp is not /better/ for provisioning though.. tftp is faster and simpler. aditionally switches, routers, and other network devices have operated on tftp for years. |
| 19:01.50 | [TK]D-Fender | CunningPike : That's GrandSUCK actually... |
| 19:02.00 | TrixVox | tftp blows |
| 19:02.02 | justinu|laptop | i prefer HTTP |
| 19:02.07 | TrixVox | http rocks |
| 19:02.08 | CunningPike | [TK]D-Fender: lol |
| 19:02.15 | dpryo | My phones uses https .. like wtf |
| 19:02.23 | [TK]D-Fender | justinu|laptop : What about uploads? To boot fine, but when it has to "talk back"? |
| 19:02.24 | TrixVox | https is even better |
| 19:02.37 | justinu|laptop | [TK]D-Fender: mod_put :) |
| 19:02.39 | jbalcomb | umm... http/https is silly |
| 19:02.44 | dpryo | and they use ftp for backups. |
| 19:02.49 | dpryo | Avaya 4620 |
| 19:03.03 | justinu|laptop | http/s works in nearly any environment... try that with tftp/ftp |
| 19:03.05 | TrixVox | do you really need to backup a $50 sipura? |
| 19:03.05 | r_evolution | i dunno... i've had some issues provisioning with TFTP |
| 19:03.17 | CunningPike | jbalcomb: Agreed on http(s) - I find FTP just as easy to set up as TFFTP |
| 19:03.24 | TrixVox | what's to backup, you already have the config it's using |
| 19:03.35 | dpryo | TrixVox: Probably not, my phones have addressbooks and such stored locally. |
| 19:03.36 | CunningPike | jbalcomb: But ymmv - whatever works |
| 19:03.46 | *** join/#asterisk Cresl1n_ (n=matt@gateway.digium.com) |
| 19:03.53 | jbalcomb | TrixVox: some phones upload config diferential files as well as logs |
| 19:05.11 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 19:07.38 | [TK]D-Fender | justinu|laptop : FTP should be just as easy as HTTP. |
| 19:07.46 | XARiUS | I'm cursing tftp myself right now.. :) |
| 19:07.59 | XARiUS | only cause I'm not using it :( |
| 19:08.42 | [TK]D-Fender | The only reason I use TFTP here is because of the 2 craptastic UIP-200's I've got floating around. Wish I could dump them somewhere..... |
| 19:09.12 | XARiUS | man.. I bought one of those when they first came out in lots of about 20.. lol |
| 19:09.22 | XARiUS | I held out hope for them.. but what a POS. |
| 19:09.54 | XARiUS | I've still got one of the original iaxy's, the clear blue square with the wall wart twice the size of the iaxy.. haha |
| 19:10.34 | coppice | Uniden is a japanese company that desperately wants to be a low end chinese one :-) |
| 19:10.38 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 19:10.59 | XARiUS | anyone tried that cordless handset + sip base unit product they have yet? |
| 19:11.01 | blebleble | anyone good with hylafax? i am able to send out faxes from the command line but not from any fax client software |
| 19:11.12 | Krush | hmm. i still got several systrace with misdn, even while reloading the module. profiling kernel support is disabled. any ideas? |
| 19:11.12 | XARiUS | from the pics and specs it looks very cool, but I'm guessing it works about as well as the UIP 200's |
| 19:11.44 | phigwork | What's the syntax for setting PHONE (the phone to ring) to two extensions? PHONE=Zap/3,SIP/101? |
| 19:11.54 | XARiUS | ampersand |
| 19:11.57 | *** join/#asterisk asterisk_newbie (n=a_ti_tu_@bl7-128-21.dsl.telepac.pt) |
| 19:12.02 | phigwork | Zap/3&SIP/101 |
| 19:12.03 | *** join/#asterisk vivek (n=vivek@unaffiliated/tintin) |
| 19:12.06 | XARiUS | SIP/device&SIP/anotherdevice&ZAP/channel |
| 19:12.07 | phigwork | thanks |
| 19:12.26 | jbalcomb | what the hell is 'invalid opcode (2)' |
| 19:12.27 | TrixVox | fyi, sipuras can report back their configs via tftp, http or https |
| 19:12.56 | jbalcomb | yeah, we should all switch to those neato SipYerAs phones. |
| 19:13.18 | *** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com) |
| 19:14.03 | XARiUS | gotta say, very impressed with these ip501's.. I bought 5 on recommendations of folks here. Had never used one before yesterday. |
| 19:14.28 | XARiUS | I'm not doing any of the presence stuff for this client, but I might pick up another one to play with :) |
| 19:14.46 | *** join/#asterisk gravis- (i=jayson-@69.60.244.25) |
| 19:14.49 | justinu|laptop | they are pretty good for IP phones |
| 19:15.01 | Qwell[] | XARiUS: The standard recommendation fee is one polycom per 5 purchased |
| 19:15.03 | XARiUS | blow the crap outta this 7960 I have on my desk for twice the price |
| 19:15.06 | coppice | XARiUS: doesn't "i bought 5 on the recommendation of some people on the internet i've never met" sound a little gullible? :-) |
| 19:16.37 | jbalcomb | XARiUS: I also started buying ip501s on the inchannel recommendations and am quite pleased |
| 19:16.38 | XARiUS | lol coppice, if I hadn't been in this chan (on and off) for the better part of 3 years.. yeah it would :) |
| 19:16.38 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 19:16.38 | XARiUS | hell, I owe most people in here a beer for one reason or another. specially twisted. |
| 19:16.38 | gravis- | I'm trying to flash my Cisco 7961 from SCCP to SIP-8.2, and when it goes to load the image it says "Load Authentication Failed P0S3-08-2-00"? Any ideas? |
| 19:16.38 | *** part/#asterisk dlynes_office (n=dlynes@216.251.149.66) |
| 19:16.39 | Qwell[] | gravis-: wrong firmware? |
| 19:16.39 | Qwell[] | 7960 != 7961 |
| 19:16.39 | *** join/#asterisk [Airwolf] (n=airwolf@84.246.5.160) |
| 19:17.07 | gravis- | Qwell: Its a 7961 |
| 19:17.08 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.197.209) |
| 19:17.21 | Qwell[] | gravis-: yeah, isn't the firmware for the 7961 named differently? |
| 19:17.27 | Qwell[] | That is a 7960 firmware filename |
| 19:17.45 | [Airwolf] | Does anyone know a sip client for Windows Mobile 5.0 ? |
| 19:17.50 | jbalcomb | coppice: is that worse than having unprotected sex with a girl you met at the bar? |
| 19:17.53 | Qwell[] | and why would you bother with sip on those anyhow? |
| 19:17.58 | XARiUS | sjphone supposedly, airwolf |
| 19:17.59 | [Airwolf] | I tried SJphone, but it doesn't work very good |
| 19:18.06 | XARiUS | yeah =/ it sucked for me too |
| 19:18.11 | XARiUS | it crashed continually on my treo700w |
| 19:18.33 | [Airwolf] | Here too and te mobile became slow |
| 19:18.35 | [Airwolf] | the |
| 19:18.42 | gravis- | Qwell: Well i flash all my cisco phones with Sip to work better with asterisk |
| 19:18.46 | [Airwolf] | You havn't found another ? |
| 19:19.04 | Qwell[] | gravis-: well, you'll need to use the proper firmware |
| 19:19.04 | XARiUS | I looked briefly, but gave up after an hour.. so nope =/ |
| 19:19.05 | gravis- | Qwell: ill look for it now |
| 19:19.11 | Qwell[] | those phones, even though they look the same, and have similar models, are completely different |
| 19:19.21 | [Airwolf] | XARiUS, to bad |
| 19:19.26 | XARiUS | whats the diff with the 7961? |
| 19:19.30 | Qwell[] | XARiUS: everything |
| 19:19.32 | XARiUS | haven't even looked at'em yet |
| 19:19.42 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
| 19:19.44 | Qwell[] | but, the only real noticable thing, is the lit line buttons |
| 19:20.04 | XARiUS | I had a cute "feature" on my 7960 yesterday, I made an outbound call, phone spontaneously rebooted, yet continued to ring all thru the boot cycle. |
| 19:20.06 | XARiUS | gave me a chuckle |
| 19:20.24 | Qwell[] | XARiUS: that's what you get for using sip... |
| 19:20.28 | XARiUS | lol |
| 19:20.35 | XARiUS | okay so what should I be using, sccp? :) |
| 19:20.42 | Qwell[] | chan_skinny ;) |
| 19:20.53 | XARiUS | ahh.. see, back when I first got into * (uh, almost 3yrs ago?) |
| 19:20.58 | *** join/#asterisk bmg505 (n=leon@c1-215-6.rndf.isadsl.co.za) |
| 19:21.02 | XARiUS | the skinny implementation sucked, so I never looked back and went with sip |
| 19:21.03 | *** join/#asterisk dacleric (n=dacleric@p548219E3.dip0.t-ipconnect.de) |
| 19:21.09 | Qwell[] | it still sucks |
| 19:21.19 | Qwell[] | but, feel free to donate to the cause |
| 19:21.21 | XARiUS | LOL then why should I use it?? |
| 19:21.29 | Qwell[] | (me being "the cause") |
| 19:21.34 | XARiUS | ahh I see. :) |
| 19:22.38 | *** join/#asterisk TrixV0x (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 19:22.55 | Qb3rt | i am trying to call in my pbx... a voice is telling that the number you have dialed is not in service please check the number and try again i100... in the CLI i dont see anything... the voice look like asterisk voice.... |
| 19:23.08 | XARiUS | I think I'd rather donate to getting this supposed app_followme into the * svn.. |
| 19:23.14 | XARiUS | I can't seem to find it, however. |
| 19:23.20 | XARiUS | been looking for a solid followme app for ages. |
| 19:23.37 | Qwell[] | XARiUS: it's in bwesche's branch |
| 19:23.39 | XARiUS | at least one that doesn't require all sorts of dialplan mumbo-jumbo. |
| 19:23.43 | gravis- | Qwell: you were right cisco 79x1 are tottally different |
| 19:23.44 | CunningPike | Isn't there a variable for the CID of the original caller when a call is transfered to an Asterisk extension from a phone on the other side of a PRI? RDNIS or something...... |
| 19:23.51 | Qwell[] | gravis-: of course I'm right :p |
| 19:24.03 | Qwell[] | for the record...if any of you folks work at cisco... |
| 19:24.07 | Qwell[] | you owe me some commission fees |
| 19:24.12 | XARiUS | lol.. |
| 19:24.20 | gravis- | Qwell: thats why i had no problem with my 7960 |
| 19:24.28 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 19:24.40 | XARiUS | CunningPike: last time I used a pri, it was the standard ${CALLERIDNUM} type stuff. didn't have to do anything special to get it. |
| 19:24.42 | ManxPower | Any ham radio people here? /msg me. |
| 19:24.52 | Qwell[] | ManxPower: well, hi to you too |
| 19:26.25 | *** join/#asterisk diLLec (n=dillec@ns3.wireless4u.de) |
| 19:26.25 | CunningPike | XARiUS: What is happening is that when our receptionist (who is on our legacy Nortel) is transferring calls to an Asterisk extension, we get her CID instead of the original caller's. That doesn't happen on Nortel sets |
| 19:27.07 | gravis- | Qwell: would you happen to be a registered cisco user? haha i need to download the firmware files |
| 19:27.39 | *** join/#asterisk ncefruby (n=cef@38.119.128.203) |
| 19:27.44 | Qwell[] | gravis-: me giving/selling it to you would be incredibly illegal |
| 19:28.11 | XARiUS | CunningPike: Not a clue then, sorry m8. Haven't done any inter-system stuff like that. |
| 19:28.21 | XARiUS | CunningPike: Stick around, I'm sure someone has the answer. |
| 19:28.31 | CunningPike | XARiUS: Thanks - anyone else? :) |
| 19:28.40 | ManxPower | CunningPike, I suspect the Nortel thinks it's a TRUNK interface and so is setting the callerid of the person transfering |
| 19:29.22 | XARiUS | Qwell: I can't believe you'd even consider SELLING it to him. :) |
| 19:29.36 | Qwell[] | XARiUS: give/sell...both just as illegal |
| 19:29.53 | XARiUS | Then again, after the run around I had to go thru to even get support contracts on my phone for the NORMAL 8 dollars.. I think I'd ask for money too lol |
| 19:30.05 | CunningPike | ManxPower: I think you are correct - it's an NI2 tie trunk |
| 19:30.08 | Qwell[] | oh, I don't even have a contract :p |
| 19:30.26 | ManxPower | CunningPike, My advice is to talk to your Nortel guru. |
| 19:30.32 | CunningPike | ManxPower: We are :) |
| 19:30.39 | Qwell[] | ManxPower: "I *am* the nortel guru!" |
| 19:30.47 | *** join/#asterisk jwa (n=jwa@columbia80.com) |
| 19:31.02 | XARiUS | OT: anyone recommend a solid irc client for osx? |
| 19:31.20 | Qwell[] | irssi through ssh? |
| 19:31.25 | jwa | hello. |
| 19:31.36 | XARiUS | eh. |
| 19:31.45 | XARiUS | was looking for something along the lines of lightweight gui :) |
| 19:32.02 | ncefruby | anybody online using latest ooh323? |
| 19:32.27 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
| 19:32.33 | Qwell[] | XARiUS: xterm not gui enough for you? :p |
| 19:32.49 | XARiUS | lol |
| 19:33.03 | ncefruby | i think there is a bug - that is not allowing any ASN.1 encoding through |
| 19:33.13 | ncefruby | wondering if anyone else may have experienced that |
| 19:33.27 | CunningPike | XARiUS: Colloquy |
| 19:33.35 | XARiUS | CunningPike: yeah was just checking that out actually :) |
| 19:33.48 | CunningPike | XARiUS: It's great |
| 19:33.57 | ManxPower | you can look at the pri debug to see what CLID info the nortel is sending you. |
| 19:34.08 | CunningPike | XARiUS: Just make sure that you run stable :) |
| 19:34.20 | XARiUS | ohh: it even comes with "Plug-In Love (tm)" |
| 19:34.28 | jwa | I am experiencing lockups w/ zaptel 1.2.6 -- w the 'zaptel' and 'wcte11xp' modules loaded & without asterisk running, I can reliably lock up the system by doing something to consume lots of memory ( cat bigfile > anotherbigfile, for instance ) |
| 19:34.34 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 19:34.56 | jwa | the system locks hard; no kernel panic/oops. |
| 19:35.10 | CunningPike | ManxPower: Yes - we're only getting the receptionists CID - I was hoping the original caller would be in a header somewhere but it isn't :( |
| 19:35.27 | CunningPike | XARiUS: Plug in love? |
| 19:35.43 | XARiUS | CunningPike: just some technobabble on that irc client website. :) |
| 19:35.53 | CunningPike | XARiUS: Ah |
| 19:35.59 | ManxPower | Yeah, it sounds like a Nortel issue. |
| 19:36.12 | CunningPike | Isn't everything? :S |
| 19:36.16 | pdtoffice | oh ya, IAXModem + Hylafax 20x better than rxfax |
| 19:36.25 | Strom_C | man, this is odd. I've got a system setup here where there is a fax machine on an SPA-2002 connecting across the LAn to an asterisk box hooked to a PRI. Faxes to and from landline fax machines work just fine. faxes to and from something called "efax" fail miserably. |
| 19:36.26 | CunningPike | Anyone lend me a forklift? |
| 19:36.46 | Qwell[] | Strom_C: efax is junk |
| 19:36.52 | XARiUS | CunningPike: How bout a 6 pack and a few sledgehammers instead? |
| 19:36.54 | Strom_C | Qwell[]: that was my suspicion |
| 19:37.17 | [TK]D-Fender | CunningPike : I did that last year :) |
| 19:37.29 | jwa | anyone else experience similar lockups? |
| 19:37.29 | CunningPike | [TK]D-Fender: Felt good, I bet |
| 19:37.36 | phigwork | hm |
| 19:37.45 | phigwork | I'm confused, and trying to get a sip phone working |
| 19:37.54 | Strom_C | Qwell[]: I'm correct in assuming that if faxes to a landline fax work just fine, come through as fast as all get out, and as clear as day, then the problem must be with efax, right? |
| 19:38.03 | rob0 | Am I correct in assuming that Vonage is a ripoff aimed at people who don't know any better? (I know that's a stupid question, sorry.) |
| 19:38.07 | phigwork | to dial a sip phone that's registered to asterisk, you just do SIP/extension, right? |
| 19:38.07 | Qwell[] | Strom_C: indeed |
| 19:38.14 | phigwork | or SIP/ip? |
| 19:38.15 | Qwell[] | rob0: finance.yahoo.com - VG |
| 19:38.25 | Qwell[] | rob0: it speaks volumes |
| 19:38.26 | XARiUS | phigwork: SIP/device-name actually. |
| 19:38.31 | phigwork | ahh |
| 19:38.35 | Strom_C | rob0: be your own telephone company with asterisk ;) |
| 19:38.39 | XARiUS | phigwork: device being the name of the device in sip.conf |
| 19:38.49 | XARiUS | phig: extensions just point at devices. |
| 19:39.06 | pdtoffice | rob0, vonage isn't too bad if you don't want to know anything about it, you just want it to work |
| 19:39.13 | XARiUS | phig: if you named your devices in sip.conf the extension name, like [1001] then yes, SIP/1001 |
| 19:39.21 | [TK]D-Fender | CunningPike : Glorious. * + Sangoma + Polycom = Peace of mind. Now if only I could ditch this GUI......... |
| 19:39.25 | phigwork | yeah, i named it the phone, ip10s |
| 19:39.34 | boch | dudes, what hw do you recommend to convert an e1 to 30 fxs ports? |
| 19:39.39 | CunningPike | [TK]D-Fender: What GUI? |
| 19:39.40 | XARiUS | phig: okie, then SIP/ip10s |
| 19:39.40 | phigwork | sweet, it rings |
| 19:39.42 | Strom_C | boch: channel bank |
| 19:39.43 | Qwell[] | boch: a channelbank? |
| 19:39.44 | phigwork | xarius: thanks |
| 19:39.51 | XARiUS | phigwork: np :) |
| 19:40.07 | pdtoffice | [TK]D-Fender, thats actually the combo we have settled on as well, had a lot of failures with the digium gear and echo is way easier to deal with on the sangoma gear |
| 19:40.09 | XARiUS | IT'S ALIVEEEE!!!! :) |
| 19:40.18 | boch | Qwell[]: but any particular brand or model? |
| 19:40.31 | Qwell[] | boch: rhino, adtran, carrier access |
| 19:40.33 | rob0 | I'm using Asterlink orig/term and several of the gratis services. |
| 19:41.17 | boch | Qwell[]: thanks |
| 19:41.22 | phigwork | now to make it dial out |
| 19:41.30 | XARiUS | phig: gl with that! have fun. |
| 19:41.32 | XARiUS | :) |
| 19:41.39 | phigwork | hehe |
| 19:41.55 | phigwork | it doesn't have many settings, and it seems i've set them all, so i must be missing something in my configs |
| 19:42.36 | XARiUS | ah, thats always the case.. a typo.. a.. wait that doesn't look right.. |
| 19:42.37 | XARiUS | :) |
| 19:42.49 | Qwell[] | wtf is up with those random calls, where it's dead air for a few seconds, then the recording says "Sorry, I've dialed the wrong number" |
| 19:43.01 | XARiUS | Qwell: Operator error? :) |
| 19:43.06 | Qwell[] | it's a recording :p |
| 19:43.13 | Juggie | Qwell, fax machine spam? |
| 19:43.17 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 19:43.22 | Qwell[] | not fax |
| 19:43.23 | phigwork | in the sip.conf where i have the [ip10s], which context= is it looking for? the one it uses for dialing out or the one for receiving calls? |
| 19:43.28 | Qwell[] | no fax tone |
| 19:43.45 | Juggie | maybe its listning for the remote tone first? |
| 19:43.51 | *** join/#asterisk davidPDX (n=davidras@65-102-35-247.ptld.qwest.net) |
| 19:43.55 | XARiUS | phig: context=whatevre is where the phone will land once you dial a number. |
| 19:43.56 | Qwell[] | Juggie: doubtful |
| 19:44.23 | phigwork | xarius: ok. it just confused me with setting up fwd and such, since those contexts are incoming :) |
| 19:44.52 | XARiUS | phig: make good use of contexts in extensions.conf, think of it like a router file, top to bottom, use includes judiciously to keep everything seperated and retain your sanity. |
| 19:45.15 | gandhijee | wheres the option in the kernel config to change the kernel hz |
| 19:45.50 | gandhijee | i don't seem to remember where it is |
| 19:46.06 | phigwork | xarius: yeh, i've been trying to do that. keeping each entry in a seperate file, too |
| 19:46.08 | [TK]D-Fender | CunningPike : ScopServ |
| 19:46.38 | XARiUS | phig: that works, I host multiple companies off one system, each company has their on ext.conf, but if you're dealing with a small first time setup, stick with one file. |
| 19:46.44 | Strom_C | Qwell[]: it's some predictive dialer |
| 19:46.50 | XARiUS | phig: break it out into seperate files after you've got it working, a lot easier that way. |
| 19:46.51 | [TK]D-Fender | CunningPike : Don't get me wrong, as far as GUI's go its the best out there, but its still a GUI... |
| 19:46.57 | phigwork | true |
| 19:47.00 | Qwell[] | Strom_C: yeah, but why would it just hangup? |
| 19:47.01 | Strom_C | Qwell[]: i guess it's the predictive dialer with the appearance of a heart |
| 19:47.09 | phigwork | i've been getting things working one by one tho. |
| 19:47.18 | phigwork | I don't see anything on the command line console coming from the phone |
| 19:47.26 | *** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net) |
| 19:47.32 | XARiUS | phig: did you setup your sip entry as a friend, peer or user? |
| 19:47.40 | phigwork | I can't tell if I've got a config wrong or if its the phone. |
| 19:47.44 | phigwork | friend |
| 19:47.57 | phigwork | should proxy server and outbound proxy be the same thing on the phone? |
| 19:48.06 | XARiUS | what phone? |
| 19:48.13 | phigwork | Swissvoice ip10s |
| 19:48.26 | XARiUS | eh, never touched one, trial by elimination. |
| 19:48.27 | phigwork | I see example configs for mcgs or whatever, but not sip |
| 19:48.41 | Qwell[] | mgcp? ugh |
| 19:48.46 | phigwork | yeah mgcp |
| 19:48.54 | *** join/#asterisk Eggplants (n=none@dsl-216-155-213-228.cascadeaccess.com) |
| 19:48.58 | phigwork | multimedia gateway control protocol? |
| 19:49.43 | XARiUS | phig: how many contexts do you have in your extensions.conf? |
| 19:49.51 | CunningPike | [TK]D-Fender: I'll check it out - would it be useful for help desk types, for MAC? Does it limit what you can do manually like A@H does? |
| 19:50.16 | [TK]D-Fender | CunningPike : MAC? |
| 19:50.31 | CunningPike | Moves, Adds, Changes |
| 19:50.34 | CunningPike | ~mac |
| 19:50.36 | jbot | i guess mac is the best computer ever |
| 19:50.57 | phigwork | xarius: hm, like, 10, or so.. I have local, tollfree, etc seperated and included in one context |
| 19:51.19 | AlexCTI | Hi everyone, I have a * box with board with 4 PRIs attach to a PBX and it was wrking pretty fine but one of the ports, lost the DATA sync, I make some tests, swicth cables and ports and i couldn't mae that it goes back, so my question is there is any tool which fix that? |
| 19:51.23 | [TK]D-Fender | CunningPike : And of course it limits you... its a GUI! But its far less limiting than AMP, and includes TONS more in he "canned" side. I destroys every other canned system lets say. |
| 19:51.23 | CunningPike | jbot, MAC is an abbreviation for Moves, Adds and Changes |
| 19:51.24 | jbot | ...but mac is already something else... |
| 19:51.24 | phigwork | what's weird is, when I set host=<ip of phone>, asterisk complained that it wasn't set to dynamic |
| 19:51.39 | phigwork | why couldn't i specify the ip in host=? |
| 19:51.42 | CunningPike | jbot, MAC is also an abbreviation for Moves, Adds and Changes |
| 19:51.44 | jbot | okay, CunningPike |
| 19:52.00 | XARiUS | phig: you working with your iphone and * on the same lan? |
| 19:52.02 | [TK]D-Fender | ~mac |
| 19:52.03 | jbot | [mac] the best computer ever, or an abbreviation for Moves, Adds and Changes |
| 19:52.08 | phigwork | xarius: yeah |
| 19:52.14 | XARiUS | phig: set it to dynamic. |
| 19:52.21 | *** join/#asterisk QuAtRo[NL] (n=jos@ip54532855.speed.planet.nl) |
| 19:52.25 | phigwork | cool |
| 19:52.50 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 19:52.50 | *** part/#asterisk KranZ (n=user@sme.bestline.net) |
| 19:52.52 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 19:53.01 | QuAtRo[NL] | Does someone have a script or a way to log all calls? |
| 19:53.02 | KranZ | poop |
| 19:53.10 | QuAtRo[NL] | In a custom format? |
| 19:53.13 | Qwell[] | QuAtRo[NL]: like...cdr maybe? |
| 19:53.20 | phigwork | doesn't asterisk log calls to xml or something? |
| 19:53.21 | jbalcomb | ok, how about a symlink from /var/http and /var/ftp to /var/tftp? |
| 19:53.28 | Qwell[] | say, cdr_custom.so perhaps? |
| 19:53.38 | KranZ | anyone done a pri+data over a T1 through a sangoma? |
| 19:53.46 | KranZ | non mlppp |
| 19:54.45 | QuAtRo[NL] | I have already made a script (with phpagi) which determines when a call starts and who calls who... |
| 19:55.18 | QuAtRo[NL] | But i can't determine when a call ends... |
| 19:55.52 | *** part/#asterisk davidPDX (n=davidras@65-102-35-247.ptld.qwest.net) |
| 19:55.59 | jwa | I am experiencing lockups w/ zaptel 1.2.6 -- w the 'zaptel' and 'wcte11xp' modules loaded & without asterisk running, I can reliably lock up the system by doing something to consume lots of memory ( cat bigfile > anotherbigfile, for instance ) .. the box functions ok without the modules loaded, so i'm thinking it's either the modules or the card.. any ideas? |
| 19:56.06 | *** join/#asterisk MrDigital (n=Mr@pool-72-81-31-172.phlapa.east.verizon.net) |
| 19:56.25 | jbalcomb | QuAtRo[NL] how about making a system call in your dialplan that echos a log entry |
| 19:56.57 | jbalcomb | Anyone like to recommend a particular FTP Server for the Linux (Debian)? |
| 19:57.07 | MrDigital | anyone here buy from voipsupply? |
| 19:57.22 | KranZ | yup |
| 19:57.26 | gandhijee | MrDigital: i have |
| 19:57.26 | Nugget | I bought a 7960 from them. No complaints. |
| 19:57.28 | QuAtRo[NL] | jbalcomb: Proftpd is stable |
| 19:57.51 | gandhijee | 2 WIP's took a while to ship, but that wasn't really thier fault |
| 19:57.59 | AlexCTI | Hi everyone, I have a * box with board with 4 PRIs attach to a PBX and it was wrking pretty fine but one of the ports, lost the DATA sync, I make some tests, swicth cables and ports and i couldn't mae that it goes back, so my question is there is any tool which fix that? |
| 19:58.49 | KranZ | reboot |
| 19:58.49 | MrDigital | nugget: how quick did your order ship? |
| 19:58.59 | *** join/#asterisk klav (n=hatred@dD5769EC1.access.telenet.be) |
| 19:59.20 | klav | hi |
| 19:59.30 | Nugget | next day I guess, I don't recall. as quick as I was expecting it to in any event. |
| 20:00.20 | klav | i'm testing asterisk in vmware, and i get tons of stutter on the calls to the demo extensions ; when i call a number that doesnt exist the sounds is fine; any ideas? |
| 20:01.19 | QuAtRo[NL] | jbalcomb: I don't fully understand what you mean |
| 20:01.40 | Qwell[] | klav: don't use vmware |
| 20:01.53 | Qwell[] | klav: and yes...I'm serious |
| 20:02.03 | klav | :) |
| 20:02.32 | klav | hmm |
| 20:02.45 | klav | so there's no way of deploying asterisk on a virtualised environment? |
| 20:02.53 | Strom_C | Qwell[]: we should start a wiki: "The horrible things people do to asterisk" :) |
| 20:03.10 | Qwell[] | Strom_C: There is already a mailing list |
| 20:03.13 | QuAtRo[NL] | klav: It does work fine on Xen |
| 20:03.14 | Qwell[] | Strom_C: -users... |
| 20:03.18 | Strom_C | hahaha |
| 20:03.19 | ManxPower | klav, I strongly doubt a virtualized enviroment will be close enough to a realtime OS |
| 20:03.52 | Strom_C | klav: asterisk operates within fairly strict tolerances |
| 20:04.12 | klav | well it's no point of memory or cpu power as they're both not in the 20%+ range |
| 20:04.13 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 20:05.00 | heison | looking for a media gateway appliance that can talk SIP/IAX with interface with PRI, FXO and FXS... any recommendation other than AudioCodes Mediant ? |
| 20:05.23 | Qwell[] | heison: asterisk |
| 20:05.34 | klav | anyway, thanks for the answers guys |
| 20:05.58 | heison | okay, rephrase... other than Asterisk and AudioCodes Mediant |
| 20:07.16 | QuAtRo[NL] | Isn't there a simple way to call a function when someone hangs up? |
| 20:07.54 | CunningPike | QuAtRo[NL]: Use the 'h' extension |
| 20:07.59 | Strom_C | the h extension? :) |
| 20:08.13 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-20-4.w193-253.abo.wanadoo.fr) |
| 20:08.15 | QuAtRo[NL] | It does exist but i don't know where to place it... |
| 20:08.16 | jhiver | hi lads |
| 20:08.23 | jhiver | have you seen this error before? |
| 20:08.31 | jhiver | ERROR: Could not open H.323 listener port on 1720 |
| 20:08.31 | jhiver | Jul 6 21:09:26 ERROR[2827]: chan_h323.c:2366 load_module: Unable to create H323 listener. |
| 20:08.31 | jhiver | Jul 6 21:09:26 WARNING[2827]: loader.c:414 __load_resource: chan_h323.so: load_module failed, returning -1 |
| 20:08.54 | Qwell[] | jhiver: You have another thing listening on 1720? |
| 20:08.55 | jhiver | I'm trying to use debian sid's asterisk and asterisk-h323 packages |
| 20:09.09 | QuAtRo[NL] | CunningPike: Where to place the h? |
| 20:09.16 | QuAtRo[NL] | Instead of the phonenumer? |
| 20:09.16 | Strom_C | and what version of asterisk is that, god forbid, jhiver? |
| 20:09.20 | [TK]D-Fender | heison : Mediatrix |
| 20:09.26 | Qwell[] | Strom_C: 0.97 |
| 20:09.28 | jhiver | let me check |
| 20:09.30 | Strom_C | haha |
| 20:09.39 | heison | i'm also having difficulty detecting '#' with Perl AGI, it detects [0,9] and * without issues, but '#' seems to appear as an empty string in Perl AGI, any clue? |
| 20:09.45 | *** join/#asterisk Tigersand (n=Sandtige@adsl-212-213-18.bhm.bellsouth.net) |
| 20:11.00 | heison | [TK]D-Fender: I saw their name but don't see a device that could do PRI, FXO & FXS all in one box... |
| 20:11.04 | jhiver | ok, according to lsof -i, nothing listening on port 1720 |
| 20:11.15 | Qwell[] | jhiver: including *? |
| 20:11.23 | Qwell[] | jhiver: ie; you have two h323 drivers |
| 20:11.41 | jhiver | I have apt-get installed the nufone's stack |
| 20:11.50 | jhiver | the other one has performance problems apparently |
| 20:11.55 | [TK]D-Fender | heison : All in one? Never seen ANY that do.... |
| 20:12.06 | Strom_C | jhiver: what version of asterisk though? |
| 20:12.11 | jhiver | Asterisk 1.2.9.1 |
| 20:12.11 | Strom_C | does it have a version number at all? |
| 20:12.26 | Strom_C | 1.2.9.1 from a DEBIAN package?? |
| 20:12.30 | jhiver | yes |
| 20:12.32 | jhiver | debian sid |
| 20:12.32 | heison | [TK]D-Fender: yeah, all in one... there are 2 boxes out there that I know of... Asterisk & AudioCodes Mediant 1000 |
| 20:12.37 | Qwell[] | I refuse to believe that |
| 20:12.46 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 20:12.52 | Strom_C | oh christ, you're running asterisk on a bleeding-edge unstable distro of debian |
| 20:12.56 | [TK]D-Fender | heison : So you can get an FXO, FXS, and PRI card all in 1 chassis with them? |
| 20:12.58 | Strom_C | no wonder you're having problems |
| 20:13.21 | jhiver | you think that's because of that? |
| 20:13.34 | heison | [TK]D-Fender: yes, I can put FXO, FXO and PRI modules in the same box - simultaneously |
| 20:13.35 | Strom_C | that's at least where I might point the appropriate fingers :) |
| 20:13.40 | jhiver | jeez man, sarge is incredibly out of date and sid doesn't work... great :) |
| 20:13.51 | Qwell[] | jhiver: welcome to Debian |
| 20:13.53 | *** join/#asterisk smackus (n=smackus@63.149.122.93) |
| 20:13.55 | droops | lol |
| 20:13.59 | Strom_C | jhiver: what the hell do you need blleding-edge on an asterisk box for? |
| 20:14.14 | droops | he wants pain strom |
| 20:14.21 | Strom_C | jhiver: my standard asterisk install is debian sarge stable and asterisk svn stable branch |
| 20:14.26 | Strom_C | works like a charm |
| 20:14.26 | jhiver | so that the config files I use on my FreeBSD box work out of the box :) |
| 20:14.28 | Qwell[] | droops: That's PainStrom(TM) |
| 20:14.47 | smackus | I am having problems with voicemail stuff... When i record a busy/unavailable message, it creates the files, but when I call the phone, it still plays the default recordings. |
| 20:14.47 | jhiver | asterisk svn stable branch is version what these days? |
| 20:14.55 | Qwell[] | jhiver: svn stable branch.. |
| 20:14.59 | smackus | the files are created and owned by root |
| 20:15.00 | Qwell[] | that is the "version" |
| 20:15.03 | jhiver | lol ok |
| 20:15.18 | Strom_C | smackus: exten => watever,n,Voicemail(u${EXTEN}) |
| 20:15.57 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:15.57 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 20:16.00 | smackus | it plays back the appropriate message, just not the custom recorded one. |
| 20:16.08 | droops | hmmm painstrom(tm) |
| 20:16.14 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:16.21 | smackus | the default one as if I had never gone through the voicemail setup. |
| 20:16.36 | jhiver | right... |
| 20:16.38 | CunningPike | QuAtRo[NL]: exten => h,1,foo() |
| 20:16.41 | heison | has anyone here been able to detect '#' from Perl AGI? |
| 20:16.51 | jhiver | I guess I'll have to downgrade to sarge and see if I have the same issue |
| 20:16.53 | CunningPike | QuAtRo[NL]: It needs to be in the context that the current call is in |
| 20:17.12 | smackus | what are the file permissions supposed to be for voicemail accounts, and who should the owner be? |
| 20:17.19 | r_evolution | hey Qwell... no need to hate on debian ;) |
| 20:17.21 | [TK]D-Fender | heison : Thats a very "specialty" factor I don't see a lot of companies catering to.... |
| 20:17.28 | r_evolution | * is working just fine with sarge here ;x |
| 20:17.35 | Qwell[] | r_evolution: it deserves it every once in a while :p |
| 20:17.37 | smackus | drwxr-xr-x 4 root root 4096 Jun 30 08:22 voicemail |
| 20:17.40 | r_evolution | nevar! |
| 20:17.41 | heison | $AGI-get_data() returns '' (empty string) when '#' is pressed |
| 20:17.42 | r_evolution | HERETIC! |
| 20:17.59 | smackus | -rwx------ 1 root root 330 Jul 6 14:05 busy.gsm |
| 20:18.26 | r_evolution | hey jhiver... what's your issue? |
| 20:18.29 | [TK]D-Fender | smackus : pastebin your voicemail calls in extensions.conf (everything leading to them) |
| 20:18.40 | heison | [TK]D-Fender: well, we're trying to build an Asterisk box but won't want to be tied to much hardware drivers... |
| 20:18.41 | r_evolution | the name smackus makes me laugh. |
| 20:18.46 | smackus | ok... brb |
| 20:18.49 | r_evolution | ILL SMACKUS YOU IN YOUR EYE! |
| 20:18.50 | smackus | glad to amuse |
| 20:18.51 | jhiver | r_evolution, I need a decently recent asterisk on my debian box |
| 20:18.55 | r_evolution | and? |
| 20:18.57 | Qwell[] | jhiver: don |
| 20:18.58 | heison | [TK]D-Fender: that way, we can easily provision a fail over machine |
| 20:18.58 | r_evolution | i use 1.2.9.1 |
| 20:18.59 | Qwell[] | t use packages |
| 20:19.04 | r_evolution | no kidding |
| 20:19.07 | r_evolution | make install anyone? |
| 20:19.20 | jhiver | but I can't be arsed to compile it because i'm too lazy to do it any other way the freebsd "ports" do it :) |
| 20:19.21 | CunningPike | smackus: Are you running as root? |
| 20:19.27 | r_evolution | it's not like it's THAT hard |
| 20:19.33 | r_evolution | or it takes that long |
| 20:19.41 | r_evolution | dude... if you're too lazy to compile... |
| 20:19.42 | heison | without physically moving PSTN lines - unless if the failure happens on the actual media gateway |
| 20:19.45 | r_evolution | i dont know what to tell you :( |
| 20:19.52 | jhiver | well at any rate, I'll move to testing and see how it goes |
| 20:20.02 | smackus | yep |
| 20:20.12 | kay2 | [TK]D-Fender: when I do an "Originate" in the AMI, I can't send any other command as long as the callee hasnt answered ? |
| 20:20.15 | jhiver | well I don't really see the point of having all these wonderful package systems if you can't use them really |
| 20:20.19 | CunningPike | smackus: Did you pastebin? |
| 20:20.24 | r_evolution | eh. seriously. It's working fine here for me... we've got like 250 (?) people on it active |
| 20:20.27 | smackus | just about |
| 20:20.40 | r_evolution | eh... debian is just not updated often b/c people like everything to be secure ;) |
| 20:20.49 | r_evolution | ;) |
| 20:20.49 | smackus | http://pastebin.ca/80776 |
| 20:20.55 | r_evolution | ;x |
| 20:21.03 | *** join/#asterisk SparFux (n=player@e182017087.adsl.alicedsl.de) |
| 20:21.12 | r_evolution | eh I use a lot of the packages for everything else |
| 20:21.26 | r_evolution | i just prefer to make sure some things are as up-to-date as possible |
| 20:21.59 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:22.01 | r_evolution | man... so we have this new ATA i'm playing with... and i'm not sure if it's the BOX itself... or the gateway the box is behind |
| 20:22.08 | r_evolution | but i'm getting some straight shite ping times |
| 20:22.10 | r_evolution | 1005 ms O_O |
| 20:22.40 | CunningPike | r_evolution: Sounds like your packets stopped for a burger and a shake along the way |
| 20:22.49 | r_evolution | no kidding! |
| 20:22.50 | [TK]D-Fender | heison : Just because you don't want to have hardware driver issues with * doesn't mean your external solution has to be all in the same chassis. |
| 20:22.54 | r_evolution | sounds like they stopped and copulated |
| 20:23.06 | smackus | r_evolution: what distro are you using? |
| 20:23.06 | jhiver | and I /have/ to use sarge because I can't install the g729 codec on FreeBSD and Digium told me "it's not supported - go away" |
| 20:23.09 | r_evolution | yay more packets! er waite |
| 20:23.11 | [TK]D-Fender | heison : You can have 1U for PRI, 1U for FXS, and 1U for FXO for example. |
| 20:23.17 | r_evolution | i'm using sarge smackus |
| 20:23.19 | jhiver | after shelling out 300 bucks, it kind of hurts a bit |
| 20:23.22 | r_evolution | haha jhiver |
| 20:23.24 | r_evolution | no kidding! |
| 20:23.24 | [TK]D-Fender | kay2 : no idea |
| 20:23.31 | r_evolution | i'd throw rocks at them |
| 20:23.33 | CunningPike | smackus: Are you getting an error? Or what? :S |
| 20:23.34 | kay2 | [TK]D-Fender: thx |
| 20:23.34 | jhiver | seriously man |
| 20:23.42 | r_evolution | "we don't support it now FUX OFF" |
| 20:23.52 | jhiver | it's their ./register utility which doesn't detect any NIC cards |
| 20:23.57 | [TK]D-Fender | jhiver : You don't HAVE to use Sarge.. you jsut can't use BSD ;) |
| 20:23.57 | heison | [TK]D-Fender: yes, that would work technically, but no, coz there would be more places to fail... plus a lot more rack space |
| 20:24.06 | [TK]D-Fender | jhiver : Use Slackware then for that BSD feel ;) |
| 20:24.09 | jhiver | so it can't compute the stupid "id" they need |
| 20:24.18 | harryvv | Have there been any new patches in asterisk that will allow call waiting to work? |
| 20:24.37 | r_evolution | yeah that shite does get on my nerves... |
| 20:24.39 | [TK]D-Fender | heison : More space true, but your's has a single point of failure... whcih is worse I ask you? ;) |
| 20:24.41 | r_evolution | i added a freaking NIC |
| 20:24.45 | jhiver | Digium does not provide support for FreeBSD or the FreeBSD G.729 |
| 20:24.46 | r_evolution | and had to call Digium |
| 20:24.46 | jhiver | register utility, either freely or commercially. The path you |
| 20:24.46 | jhiver | downloaded the FreeBSD G.729 register utility is listed as unsupported. |
| 20:24.50 | r_evolution | i was like... you cunts. |
| 20:24.54 | jhiver | that's what they told me :) |
| 20:25.03 | [TK]D-Fender | harryvv : NOT happening.... |
| 20:25.05 | r_evolution | jerks. |
| 20:25.14 | jhiver | but the thing is THERE IS A NIC IN THE BOX :) |
| 20:25.20 | r_evolution | but seriously man... it's working fine on sarge |
| 20:25.23 | r_evolution | on i don't doubt it |
| 20:25.26 | r_evolution | oh* |
| 20:25.30 | r_evolution | not on |
| 20:25.39 | r_evolution | else wise you prolly couldn't DL it directly to the box ;) |
| 20:25.49 | r_evolution | unless they've come up with a new technology that allows one to DL simply by thought |
| 20:25.51 | jhiver | ok, I'll reinstall sarge and when it doesn't work I'll come and moan to you ok :) |
| 20:25.54 | harryvv | TK, okay I guess then the ONLY way around this is to find some obscure SIP provider in the greater Vancouver area that will port our number. |
| 20:26.04 | r_evolution | haha... i'd rather you come and moan to me BEFORE you re-install |
| 20:26.06 | smackus | CunningPike: no errors... just when we go through the voicemail setup and record all of the custom greetings... ie: this is smackus, leave me a message... and then call back into voicemail, I get whats her names default recording saying please leave a message. not the personalized messages I recorded. |
| 20:26.09 | r_evolution | make both our lives easier |
| 20:26.16 | r_evolution | PM me |
| 20:26.17 | jhiver | ok: MOAN MOAN |
| 20:26.20 | r_evolution | O_O |
| 20:26.23 | jhiver | now I reinstall a sarge: ) |
| 20:26.24 | r_evolution | that sounds dirty |
| 20:26.25 | *** join/#asterisk Trazz (n=traderz@65.114.86.29) |
| 20:26.30 | r_evolution | i think my girl might get jealous O_o |
| 20:26.37 | CunningPike | smackus: Do the files actually exist in the right place? |
| 20:26.48 | *** part/#asterisk Trazz (n=traderz@65.114.86.29) |
| 20:27.14 | smackus | "/var/spool/asterisk/voicemail/progrexion/156/" which is where the files were created when we recorded them to the phone. |
| 20:27.38 | *** join/#asterisk apocn (n=apo@201.229.137.218) |
| 20:27.53 | [TK]D-Fender | smackus : pastebin your voicemail calls.... |
| 20:28.06 | smackus | ok, the actuall ls -l of the directory? |
| 20:28.31 | [TK]D-Fender | smackus : No, I wanna see your dial-plan |
| 20:28.33 | apocn | Hello, I'm logged to the Asterisk Management Interface, but when I send "Action: SIPpeers" the response message is: Access Denied. why is that? |
| 20:28.49 | apocn | "Response: Error | Message: Permission denied" |
| 20:28.52 | [TK]D-Fender | apocn : Didn't set your rights up for that login properly |
| 20:28.57 | heison | [TK]D-Fender: with the all in one box solution, it's easy to buy 2 of those boxes... |
| 20:29.33 | apocn | [TK]D-Fender, when I login I get: "Response: Success | Message: Authentication accepted" |
| 20:29.41 | smackus | http://pastebin.ca/80776 |
| 20:30.47 | CunningPike | smackus: Shouldn't it be exten => s,3,VoiceMail(${ARG1}@${ARG4}) ? |
| 20:30.57 | apocn | [TK]D-Fender, I just noticed something: I sent the "Action: ListCommands" , and there I see: IAXpeers but SIPpeers is not in the list... |
| 20:31.00 | CunningPike | The 4th argument is your voicemail context |
| 20:31.19 | smackus | CunningPike: lemme look |
| 20:31.21 | *** part/#asterisk jwa (n=jwa@columbia80.com) |
| 20:31.26 | [TK]D-Fender | apocn : just because you are allowed to CONNECT doesn't give you the RIGHTS to do EVERYTHING. |
| 20:31.27 | apocn | I can see the Queues, etc, etc... (List agents, etc..) but can't do SIPpeers, why? |
| 20:31.33 | apocn | hehe, ok |
| 20:32.01 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 20:32.02 | CunningPike | smackus: Sorry - brain fart |
| 20:32.05 | apocn | I have read/write access to system,call,log,verbose,command,agent,user |
| 20:32.07 | smackus | exten => 156,1,Macro(progrexion_exten,201,201,progrexion) |
| 20:32.11 | [TK]D-Fender | smackus : You need to prefix your boxes with "b" for the busy msg, or "u" for the unavailable one.... |
| 20:32.14 | smackus | yeah... the first is the macro name, right? |
| 20:32.19 | *** join/#asterisk Katty (n=aisaacs@64.82.232.54) |
| 20:32.23 | smackus | so it is still arg 3, right? |
| 20:32.24 | [TK]D-Fender | smackus : everywhere you call Voicemail. |
| 20:32.26 | apocn | what else should I add there to have access to SIP? |
| 20:32.36 | smackus | ok. |
| 20:32.42 | [TK]D-Fender | apocn : Couldn't say. Go read the WIKI on manager.conf setup |
| 20:32.45 | Katty | afternoon. |
| 20:32.58 | apocn | [TK]D-Fender, okidoki... not enough documentation about AMI, but ok. thanks |
| 20:33.17 | [TK]D-Fender | apocn : the security options are pretty obvious. |
| 20:33.22 | [TK]D-Fender | Katty : Mew. |
| 20:34.37 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:35.56 | *** join/#asterisk monkey13 (n=monkee13@71.194.25.60) |
| 20:36.29 | smackus | [TK]D-Fender: can you verify me? http://pastebin.ca/80797 |
| 20:36.38 | Katty | hey mister fender (= |
| 20:36.49 | [TK]D-Fender | smackus : NO. |
| 20:37.33 | *** join/#asterisk TeePOG (n=1234@196.207.45.253) |
| 20:37.41 | [TK]D-Fender | smackus : As in, yes I checked, and NO, thats just wrong... go re-read the STDEXTEN sample and rebuild that before you start getting people leaving 2 VM's per call with a differen reason each time. |
| 20:38.13 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 20:38.31 | [TK]D-Fender | heading home, back in a bit. |
| 20:41.00 | *** join/#asterisk clive- (n=pirch@dsl-145-20-15.telkomadsl.co.za) |
| 20:43.40 | clive- | my 1750 is whistling so loudly here,,,,, |
| 20:44.02 | file | use a hammer... |
| 20:44.56 | clive- | lol |
| 20:44.58 | CunningPike | smackus: When you recorded your greeting, which option did you choose? Unavailable, or Busy? |
| 20:45.00 | *** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com) |
| 20:45.15 | Katty | file, deary. |
| 20:45.25 | file | Kitty Katty! |
| 20:46.00 | Katty | will i be seeing you in chicago? |
| 20:46.39 | file | nooo :( |
| 20:46.54 | r_evolution | mmmmm hammers. |
| 20:47.00 | r_evolution | TIN CAN ON A STRING!!!!! |
| 20:47.41 | Katty | )= |
| 20:47.58 | Katty | that makes me all sad inside. |
| 20:50.29 | file | awwww |
| 20:53.50 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 20:53.54 | *** part/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 20:54.19 | anthm | Katty you said you like the cure right? here's 2 live from last summer http://www.cluecon.com/mp3/ |
| 20:55.14 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.146) |
| 20:55.32 | r_evolution | good song that |
| 20:55.35 | r_evolution | just like heaven ;) |
| 20:59.22 | smackus | CunningPike: i have recorded all of them. |
| 20:59.48 | smackus | so the unavailable should be the one used by default if I do not specify which one to use in the dial plan, (the way I understood it) |
| 21:00.18 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 21:00.24 | *** part/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 21:01.29 | *** part/#asterisk apocn (n=apo@201.229.137.218) |
| 21:02.04 | CunningPike | smackus: Yes, it should...... So, you went and selected 1 for unavailable message and recorded it, and then went back and selected 2 for busy message? |
| 21:02.23 | CunningPike | smackus: Confirm that you have two files in your voicemail folder - unavail.wav and busy.wav. |
| 21:02.24 | *** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com) |
| 21:02.47 | TeePOG | exit |
| 21:02.51 | TeePOG | erm |
| 21:03.03 | TeePOG | missed the / |
| 21:03.39 | *** join/#asterisk marv0997 (i=marv0997@190.4.2.86) |
| 21:04.40 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 21:12.14 | *** join/#asterisk riddlebox (n=blah@24-207-167-238.dhcp.stls.mo.charter.com) |
| 21:12.16 | riddlebox | j #ubuntu |
| 21:12.18 | riddlebox | oops |
| 21:12.40 | MrDigital | hey CunningPike whats happenin |
| 21:12.51 | wunderkin | j #openpbx |
| 21:12.59 | CunningPike | Hey, MrDigital. What's up |
| 21:13.07 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:13.18 | MrDigital | chillin |
| 21:13.19 | MrDigital | can i pm you |
| 21:13.31 | r_evolution | he's gonna hit on you Pike! RUN! |
| 21:13.38 | jhiver | hey lads... now compiling asterisk from svn trunk 1.2... which repository do I get the nufone's h323 stuff from? |
| 21:13.38 | MrDigital | shut up r0d3nt |
| 21:13.40 | MrDigital | errrrrrr |
| 21:13.42 | wunderkin | asl? |
| 21:13.43 | MrDigital | r_evolution |
| 21:13.44 | wunderkin | ;D |
| 21:13.48 | r_evolution | heeh |
| 21:13.54 | r_evolution | im no r0d3nt |
| 21:13.56 | MrDigital | can i pm you then r_ |
| 21:14.02 | r_evolution | why? |
| 21:14.07 | MrDigital | i need ur help |
| 21:14.26 | r_evolution | i mean you can PM me all you like... it doesnt mean i'll be able to help... but you'd better off asking in here first |
| 21:14.34 | justinu|laptop | lol |
| 21:14.50 | MrDigital | i need help writing a option 1 to lookup a field in a mysql db |
| 21:15.23 | *** join/#asterisk Powerkill (n=PoWeRKiL@84.205.154.179) |
| 21:15.31 | r_evolution | eh i might be able to help |
| 21:15.35 | Powerkill | !seen kram |
| 21:15.47 | r_evolution | !seen Powerkill |
| 21:16.01 | jhiver | ok I see h323 is supposed to be in the asterisk source... |
| 21:16.09 | justinu|laptop | ~seen r_evolution |
| 21:16.27 | jbot | r_evolution is currently on #asterisk (3h 33m 23s). Has said a total of 124 messages. Is idling for 40s, last said: '!seen Powerkill'. |
| 21:16.27 | jhiver | don't know if its compilation is enabled by default though |
| 21:16.41 | r_evolution | d'oh |
| 21:16.44 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 21:16.45 | r_evolution | YOU"RE A LIAR JBOT! |
| 21:17.10 | Netgeeks | Mr. Digital: if you still need help with the mysqld db lookup, just hollar, I can give you a few pointers |
| 21:17.28 | Toerkeium | asteriskTFOT |
| 21:17.31 | Toerkeium | what a nice guide!! |
| 21:17.37 | justinu|laptop | ~book |
| 21:17.39 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:17.39 | *** part/#asterisk heison (n=heison@ns.somanetworks.com) |
| 21:17.47 | Toerkeium | book, sorry |
| 21:18.19 | Toerkeium | easy to read/understand and cover the basics you need to move on |
| 21:18.39 | riddlebox | I am having problems compiling asterisk from source, I get this error: http://pastebin.ca/80832 |
| 21:19.08 | jhiver | guys... I've compiled asterisk from source, but no chan_h323.so |
| 21:19.11 | riddlebox | can someone help me please? |
| 21:19.13 | syle | means you need to dl latest zaptel source and install that first |
| 21:19.15 | jhiver | how do I compile it? |
| 21:19.57 | r_evolution | ~book |
| 21:19.59 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:20.05 | Toerkeium | and errors in compile time are very descriptive |
| 21:20.20 | Toerkeium | riddlebox: read that book, your error is listed there |
| 21:20.30 | *** join/#asterisk nexstar (n=nexstar@adsl-67-112-181-25.dsl.lsan03.pacbell.net) |
| 21:20.56 | Toerkeium | I imagine a hard work to write that book, make the effort of that people worth :) |
| 21:21.15 | jhiver | guys... no idea on how to get / compile / install nufone's chan_h323 with svn asterisk? |
| 21:21.31 | justinu|laptop | you'll need to know C |
| 21:21.36 | justinu|laptop | and figure out which APIs changed |
| 21:21.46 | jhiver | ? |
| 21:21.49 | jhiver | it's that bad? |
| 21:21.54 | justinu|laptop | yeah |
| 21:21.55 | *** join/#asterisk evisu (n=hIRC@bzq-88-152-238-38.red.bezeqint.net) |
| 21:21.56 | justinu|laptop | probably |
| 21:21.59 | justinu|laptop | lots of stuff changes |
| 21:22.11 | nexstar | Error: Missing Dependency: zaptel = 1.2.5 is needed by package zaptel-modules-2.6.9-34.EL |
| 21:22.15 | jhiver | ok... is there any decent h323 stack I can use? |
| 21:22.21 | nexstar | trying to yum -y update and i get that error |
| 21:22.23 | TrixV0x | you can ask jerjer now that nufone doesn't get any attention |
| 21:22.30 | justinu|laptop | not sure, i'm not familiar with 323 |
| 21:22.39 | TrixV0x | he should have lots of time to rewrite h323 |
| 21:24.31 | justinu|laptop | there's much love for jerjer here |
| 21:24.35 | nexstar | Error: Missing Dependency: zaptel = 1.2.5 is needed by package zaptel-modules-2.6.9-34.EL |
| 21:24.45 | nexstar | anyone to help? |
| 21:24.46 | phigwork | Anyone know why I keep getting this: Jul 6 14:13:20 WARNING[6674]: chan_sip.c:9770 handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 21:24.47 | phigwork | ? |
| 21:25.02 | phigwork | every 14-15 minutes it seems |
| 21:25.17 | justinu|laptop | it's probably safe to ignore |
| 21:25.24 | TrixV0x | yeah, i can hear the crowds cheering for jerjer |
| 21:25.28 | phigwork | cool |
| 21:27.08 | riddlebox | Toerkeium,that error is not in the book? |
| 21:27.10 | jhiver | shit, I need to compile pwlib and openh323 from source before I can build h323 |
| 21:27.36 | jhiver | dependencies hell start here |
| 21:27.55 | r_evolution | sorry jhiver... i told you i dont use h323 :( |
| 21:28.15 | jhiver | oh sure |
| 21:29.03 | tehmaze | would it be possible to compress (mp3/wav) streams on the fly trough * ? |
| 21:29.10 | tehmaze | to gsm, and dump them on the call |
| 21:30.29 | jhiver | ok, trying to compile pwlib now... |
| 21:32.23 | Toerkeium | people, I have installed asterisk already, and I think it's working. I have a SIP line.. could I just try to connect my asterisk to this SIP line and connect to my asterisk with a softphone? would that be a good test to know if my asterisk is working? |
| 21:32.47 | jhiver | Toerkeium, you could start connecting your phone to asterisk |
| 21:32.53 | jhiver | and then use the Echo() application |
| 21:32.54 | eBody | Toerkeium, download X-lite and just test it out |
| 21:32.59 | jhiver | to hear yourself speaking |
| 21:33.07 | jhiver | then once that's done, try to connect the other leg |
| 21:33.33 | Toerkeium | hmm interesting, readed about the Echo(), it replicates anything I say on the phone, right? |
| 21:33.40 | jhiver | that's right |
| 21:33.42 | Toerkeium | to check bidirectional comunications? |
| 21:33.43 | Toerkeium | ok |
| 21:33.45 | Toerkeium | gonna check that |
| 21:34.00 | jhiver | well it allows you to sort one problem at a time |
| 21:34.22 | jhiver | once you know that there are no issues between your * server and your phone, you can worry about your SIP trunk |
| 21:34.46 | Toerkeium | yes, it's a good behavior :)) |
| 21:34.49 | smackus | CunningPike: sorry, i got pulled away on another issue... you asked me to confirm if i had busy and unavail.wav in my voicemail folder. yes i do. |
| 21:34.54 | jhiver | jeez this pwlib takes forever to compile |
| 21:35.19 | Toerkeium | jhiver: what's the parameter to configure a username and password in the sip.conf file? |
| 21:35.27 | Toerkeium | so I can register with the softphone |
| 21:35.37 | jhiver | hang on, I'll fetch u a sample conf |
| 21:35.48 | Toerkeium | thank you very much |
| 21:37.33 | jhiver | I'll msg to you |
| 21:37.40 | Toerkeium | thank you |
| 21:38.05 | jhiver | there |
| 21:40.41 | *** join/#asterisk s0lid (n=s0lid@61.28.161.132) |
| 21:40.44 | Toerkeium | it worked !! |
| 21:40.55 | *** join/#asterisk RoyK[at] (n=roy@chello080109196173.3.graz.surfer.at) |
| 21:40.57 | Toerkeium | at least it registered |
| 21:42.47 | *** join/#asterisk Lord_Drachenblut (n=Lord@74.129.228.28) |
| 21:42.51 | Lord_Drachenblut | hello |
| 21:43.12 | smackus | hi |
| 21:44.47 | RoyK[at] | <PROTECTED> |
| 21:48.21 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:48.27 | Dr-Linux | hi all |
| 21:51.20 | *** join/#asterisk ariel_ (n=ariel_@74.8.35.2) |
| 21:53.34 | *** part/#asterisk rue_mohr (n=not@bdr2.fieldrd.scrd.ca) |
| 21:54.49 | Dr-Linux | ~dict bacon |
| 21:55.10 | Dr-Linux | RoyK[at]: thanks :) |
| 21:55.47 | mrtwister | hi who still not sleep :). is asterisk have something like STRLEN in extensions ? i.e. exten => _X.,n,GoToIf(STRLEN(${CALLERIDNUM}= |
| 21:56.28 | ManxPower | mrtwister, you didn't read /path/to/src/asterisk/docs/README.variables did you? |
| 21:57.32 | RoyK[at] | Dr-Linux: not really muslim food, is it? :) |
| 21:57.33 | mrtwister | just not remember well |
| 21:57.34 | mrtwister | thanks |
| 21:58.51 | Dr-Linux | RoyK[at]: i don't know what it is, but you gave me me with good touch, so happy to accept :) |
| 21:59.08 | Dr-Linux | human feelings matter |
| 21:59.50 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 22:00.31 | justinu|laptop | pork |
| 22:05.36 | *** join/#asterisk Kokey (n=jramirez@201.123.192.227) |
| 22:14.23 | *** join/#asterisk apocn (n=apo@201.229.137.218) |
| 22:14.59 | *** join/#asterisk kpettit (n=keith@69.15.155.154) |
| 22:15.00 | *** join/#asterisk wunderkin (i=kev@0-1pool199-49.nas26.tempe1.az.us.da.qwest.net) |
| 22:15.05 | kpettit | lo |
| 22:15.14 | kpettit | I'm tring to setcallerid in macro in extensions.conf |
| 22:15.23 | apocn | Hello, to connect to my asterisk from my softphone I first need to connect to the VPN. When I try to connect I get the message: failed for '10.0.0.2' - Not a local SIP domain |
| 22:15.39 | kpettit | what's the proper way to append two varialbes? |
| 22:15.45 | kpettit | <PROTECTED> |
| 22:15.58 | apocn | err: chan_sip.c:11043 handle_request_register: Registration from 'Apocn <sip:apo@192.168.0.253>' failed for '10.0.0.2' - Not a local SIP domain |
| 22:16.12 | *** part/#asterisk eBody (n=ehernand@207.71.51.162) |
| 22:16.30 | kpettit | I have a variable I declare earlier TCSCID101 = 5551234567 and I can't seem to set that var correctly in my set callerid |
| 22:17.07 | *** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 22:19.13 | kpettit | so basicall how can I combine $arg1 and TCSCID into $TCSCID$arg1 so I can use that in setcallerid? |
| 22:19.15 | *** part/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 22:19.47 | wunderkin | kpettit, is the var set in the macro? the var may not be available where you want it |
| 22:20.04 | Toerkeium | shit! |
| 22:20.09 | Toerkeium | this thing is amazing |
| 22:20.19 | kpettit | in [general] I have a bunch of TCSCID101 = "8015551234" |
| 22:20.39 | kpettit | wunderkin, 101 in that is the extension and sip callerid |
| 22:20.54 | apocn | When I try to connect to asterisk using my softphone I get the error: chan_sip.c:11043 handle_request_register: Registration from 'Apocn <sip:apo@192.168.0.253>' failed for '10.0.0.2' - Not a local SIP domain |
| 22:20.58 | wunderkin | oh.. i see |
| 22:21.22 | wunderkin | yeah thats not how... dont remember.. |
| 22:21.33 | kpettit | so I'm making so when they dial interoffice it shows the 3 digit extension, but when they dial out (using the macro) it will show the full callerid I want to set |
| 22:21.35 | *** join/#asterisk XARiUS (n=bdarcy@66-146-191-242.skyriver.net) |
| 22:21.55 | kpettit | it used to work on older systems, but I think something is different or depreciated in this newver version |
| 22:22.09 | kpettit | I was running 1.2.7 but just upgraded to 1.2.9 to see if it helped but its' the same. |
| 22:23.01 | wunderkin | you are asking for the variable tcscid and arg1 not tcsid"arg1" |
| 22:23.10 | kpettit | So the just is creating a new variable out of two existing variables |
| 22:24.08 | kpettit | I want to create $TCSCID101 out of TCSCID$callerid(num) |
| 22:24.28 | Qwell[] | kpettit: Set(TCSCID${CALLERID(num)} = blah) |
| 22:25.27 | kpettit | Qwell, when I do that I get "TCSCID100" set for my caller id |
| 22:25.47 | Qwell[] | no |
| 22:25.56 | Qwell[] | You get the variable TCSCID100 set to blah |
| 22:26.14 | kpettit | ah got ya |
| 22:26.24 | Qwell[] | Is that not what you want? |
| 22:26.38 | XARiUS | Qwell: you mentioned that app_followme earlier, you know where that project is located so I can read up on it? |
| 22:27.03 | Strom_C | Qwell[]: i solved that fax problem |
| 22:27.03 | Qwell[] | XARiUS: bweschke's branch in svn |
| 22:27.06 | Qwell[] | Strom_C: oh? |
| 22:27.10 | Strom_C | Qwell[]: it turned out to be the fax machine |
| 22:27.13 | Qwell[] | wtf |
| 22:27.25 | Strom_C | yeah. swapped out the machine and it worked |
| 22:27.29 | Qwell[] | heh |
| 22:27.42 | Qwell[] | I still blame efax |
| 22:28.04 | kpettit | Qwell[], it shows as a blank var |
| 22:28.11 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 22:28.18 | Qwell[] | kpettit: then you did it wrong |
| 22:28.19 | Strom_C | Qwell[]: I do too |
| 22:28.39 | kpettit | In extensins.conf I have TCSCID101 = "5551234567" going from 101 to 110 or so |
| 22:28.43 | *** join/#asterisk Un1x (i=seann@CPE00016c29e15a-CM00080d40ee4c.cpe.net.cable.rogers.com) |
| 22:28.51 | Un1x | anyone know any voip providers like broadvoicE? |
| 22:29.06 | Un1x | or someone with similar plans that at least allow unlimted calls to Canada.... |
| 22:29.34 | XARiUS | don't know of many offering all you can eat packages, sorry. |
| 22:30.04 | Un1x | lol not all you can eat, but at least unlimted calling within nort america.... |
| 22:30.09 | Un1x | or at least USA+ canada |
| 22:31.12 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 22:31.37 | XARiUS | yeah thats what I meant, for those in North America, I don't know of any really. |
| 22:34.15 | jhiver | AAARGH, compiling h323 is a pain in the butt :( |
| 22:35.03 | jhiver | Un1x, why "unlimited" when in most cases using a decent termination provider will come cheaper? |
| 22:35.21 | jhiver | I mean you can get USA & Canada for sub cent rates these days |
| 22:35.38 | Un1x | yea but why go fotr subcent rates when u can try find unlimted.... |
| 22:35.50 | Un1x | ok |
| 22:35.55 | Un1x | well jhiver tell me few then |
| 22:36.13 | jhiver | well, if you make the line hot 7/7 you will get cut off anyway |
| 22:36.28 | jhiver | but I think broadvoice and voicepulse have these kind of plans |
| 22:36.36 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:36.36 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:36.38 | rob0 | Isn't Voipjet like 1.3 cents/minute? |
| 22:36.54 | jhiver | yeah but VoIPjet rates aren't that good really |
| 22:37.15 | rob0 | what do you use? |
| 22:37.19 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 22:37.22 | XARiUS | I personally haven't found any 1c/subcent providers with decent call quality. |
| 22:37.46 | XARiUS | though I haven't actively searched in the last 6 months either, maybe times have changed. |
| 22:38.22 | jhiver | I use plug&tel but we exchange traffic so... maybe I'm having a special deal :) |
| 22:38.27 | rob0 | ah |
| 22:38.43 | jhiver | let me check how much I pay for these destinations |
| 22:38.46 | XARiUS | 1.9/2.9 seems to be the going rate for us mere mortals. :) |
| 22:39.11 | *** join/#asterisk burnproof (n=burnproo@210.213.244.61) |
| 22:39.16 | jhiver | for USA I pay 0.0092 |
| 22:39.39 | jhiver | and 0.0057 for canada |
| 22:39.39 | XARiUS | yeah thats carrier level ratecenter pricing. |
| 22:39.59 | jhiver | I had to give them 500 euro cash advance, then I pay post pay |
| 22:40.25 | TrixV0x | VoicePulse offers carrier level rate center pricing to all customers with no minimums |
| 22:40.27 | rob0 | I've got Asterlink, was comparing them with Vonage earlier ... Vonage's $15 plan would cost about $4.45 at Asterlink. |
| 22:40.32 | *** part/#asterisk ariel_ (n=ariel_@74.8.35.2) |
| 22:40.39 | XARiUS | I could goto RNK and get the same, if not cheaper rates, billed on a weekly basis. |
| 22:40.58 | rob0 | 500 minutes ... I'd have trouble talking that much :) |
| 22:41.01 | TrixV0x | obviously they are slightly higher than a carrier for vp's margin, but you'd never get anything that low without a commitment |
| 22:41.26 | *** join/#asterisk Eggplant (n=none@dsl-216-155-212-105.cascadeaccess.com) |
| 22:41.30 | jhiver | Exactly, and it's ridiculously cheap already anyway |
| 22:41.33 | TrixV0x | i make about 80% RBOC calls and most of them are 0.007 or less |
| 22:42.05 | TrixV0x | i just send all the >1c through a flat-rate like voipjet, but the voice quality blows compared to vp |
| 22:42.11 | *** join/#asterisk KillahKrew (n=None@83.216.96.44) |
| 22:42.28 | *** part/#asterisk KillahKrew (n=None@83.216.96.44) |
| 22:42.41 | Strom_C | <PROTECTED> |
| 22:42.41 | Strom_C | <PROTECTED> |
| 22:42.43 | TrixV0x | if the customers complain, i swing it all over to vp, it's still a 20-30% savings |
| 22:42.46 | Strom_C | er |
| 22:42.48 | Strom_C | wtf |
| 22:42.58 | Strom_C | stupid client |
| 22:43.24 | Strom_C | exit |
| 22:43.28 | XARiUS | lol.. |
| 22:43.29 | Strom_C | ok, i fail at life |
| 22:43.43 | XARiUS | least you got that one right. :) |
| 22:45.19 | vader-- | hmmm |
| 22:45.35 | vader-- | why can't the cisco phones handle the hint/prescence function of asterisk |
| 22:45.42 | vader-- | or for that matter the sip standard |
| 22:46.52 | file | moo? |
| 22:50.27 | XARiUS | yeah, I agree, it's a moo point. |
| 22:50.28 | XARiUS | ;) |
| 22:50.49 | jhiver | vader--, I've never played with cisco :( |
| 22:51.01 | jhiver | what is the hint / presence function supposed to do? enlighten me |
| 22:51.08 | XARiUS | vader: I'm gonna toy with that on one of my polycom's.. never tried it myself. |
| 22:53.37 | rpm | what is the key sequence again to reboot Polycom IP-601's? its like Menu, Handset and something else |
| 22:54.16 | XARiUS | ehm.. |
| 22:54.42 | XARiUS | volume+/volume-/mute/messges |
| 22:55.00 | XARiUS | or messages, your preference. |
| 22:55.01 | XARiUS | :) |
| 22:55.15 | rpm | thanks :) |
| 22:55.18 | Nivex | to hear people in here talk, preferences are something like this: polycom < cisco SCCP < cisco SIP |
| 22:55.22 | XARiUS | sometimes easier to just yank the plug. |
| 22:55.41 | rpm | its a pain in the ass going through the menus or pulling the ethernet out, but my fingers are too fat. |
| 22:55.55 | XARiUS | Nivex: sccp above sip, really? |
| 22:56.11 | XARiUS | might have to donate to qwell's cause afterall =/ |
| 22:56.21 | Nivex | that's kinda what I'm wondering. Noone has much nice to say about the Cisco SIP firmware |
| 22:56.37 | XARiUS | to be honest, I've been using it for... 2 years. |
| 22:56.49 | XARiUS | I have hte occasionally spontaneous reboot (like once a month) |
| 22:57.18 | XARiUS | other than that, it's treated me well, I did a rollout of 22 7960's which are still truckin today on sip 7.1 |
| 22:57.18 | Nivex | I've not seen that, but right now two of my three lines show 'X' on them from when my asterisk box rebooted a week and a half ago |
| 22:57.58 | XARiUS | yeah I ran into that as well, it seems like if it crosses some threshold of like.. an hour, they just stop trying to register completely |
| 22:58.50 | Nivex | one thing I miss from the Cisco system I had at my last job were the little call features like DND |
| 22:59.04 | Nivex | SIP firmware has no such thing (that I can find) |
| 22:59.15 | XARiUS | nah it does, it's just burried. |
| 22:59.27 | XARiUS | settings button, 6, 1. |
| 22:59.45 | smackus | does it matter if I use sendmail or procmail for voicemail to send to email account? |
| 22:59.46 | Nivex | It will not toggle :( |
| 22:59.50 | smackus | admin took sendmail off |
| 22:59.57 | smackus | now voicemail does not work to email |
| 22:59.58 | XARiUS | Niv: odd dude. |
| 23:00.14 | XARiUS | what phone? 7940/60 or 7961? |
| 23:00.47 | smackus | Nivex: are you saying that you cannot toggle DND on your sip phone? |
| 23:01.00 | Nivex | smackus: correct |
| 23:01.09 | smackus | worked out of the box on my polycoms |
| 23:01.13 | smackus | what are you using? |
| 23:01.29 | XARiUS | in your SIP< |
| 23:01.30 | XARiUS | errr |
| 23:01.35 | XARiUS | in your sip<mac>.cfg file |
| 23:01.41 | XARiUS | dnd_control: 0 ; Default 0 (Do Not Disturb feature is off) |
| 23:01.51 | Agrajag- | gday. using the Echo command i hear myself about 0.5 - 0.75 seconds after i talk. this is on a local lan, ping time to the box running asterisk is about 0.140ms using iaxcomm and xten softphones, tried gsm and ulaw. i also have an spa-3000 and the phone connected to the fxs on that doesn't have the problem - echo is pretty much real time. it's driving me nuts, i can't figure out what's causing it |
| 23:01.57 | XARiUS | (0-off, 1-on, 2-off with no user control, 3-on with no user control) |
| 23:02.02 | Nivex | smackus: Cisco 7960 |
| 23:02.11 | XARiUS | Nivex: probably a config file setting (see above) |
| 23:02.22 | smackus | go with XARiUS's answer |
| 23:02.31 | Nivex | XARiUS: ah ha... I have it set to '2' and apparently meant '3' |
| 23:02.45 | XARiUS | so you want the phone permanantly on dnd? |
| 23:03.32 | vader-- | the hint feature basically tells you if a channel in is use |
| 23:03.38 | vader-- | so say i have a secretary and a boss |
| 23:03.46 | Nivex | XARiUS++ |
| 23:03.47 | vader-- | the secretary has her own extensino plus the boss's |
| 23:03.59 | vader-- | if the boss is on the phone it will show on her phone that he/she is on |
| 23:04.07 | XARiUS | vader: you using a polycom? |
| 23:04.08 | Nivex | now if it weren't buried... |
| 23:04.12 | vader-- | cisco 7940s |
| 23:04.17 | vader-- | 60 of them |
| 23:04.27 | Qwell[] | running sccp, I hope |
| 23:04.27 | XARiUS | oh, nice, that cost some bank. |
| 23:04.28 | XARiUS | :) |
| 23:04.32 | XARiUS | lol qwell |
| 23:04.37 | vader-- | qwell na sip |
| 23:04.41 | Qwell[] | pfft |
| 23:04.50 | vader-- | when i was configuring them everyone said run sip image with asterisk |
| 23:05.00 | Qwell[] | sip image sucks |
| 23:05.01 | XARiUS | <Qwell[]> sccp sucks!!! You all should use it, down with sip!! |
| 23:05.05 | smackus | what is sip image? |
| 23:05.06 | vader-- | i got these phones brand new 206$ a peice |
| 23:05.10 | Qwell[] | XARiUS: exactly! |
| 23:05.26 | XARiUS | not bad then vader, volume discount I'm sure at 60 :) |
| 23:05.31 | vader-- | plus educational |
| 23:05.32 | vader-- | :) |
| 23:05.49 | Qwell[] | only $206?.. |
| 23:05.51 | XARiUS | Who has setup presence on a ip501 with asterisk and wants to show me how?? :) |
| 23:05.52 | Qwell[] | erm |
| 23:05.57 | Qwell[] | that's a lot |
| 23:06.01 | Qwell[] | especially with discounts :p |
| 23:06.06 | vader-- | qwell for brand new 7940Gs? |
| 23:06.12 | vader-- | that is cheap |
| 23:06.17 | Strom_C | yeah, i was quoted $199 for 15 |
| 23:06.28 | Strom_C | so $206 for 60 seems pricey |
| 23:06.39 | Qwell[] | 60 + educ |
| 23:06.48 | Strom_C | yeah |
| 23:07.00 | XARiUS | okay that must be no one :( |
| 23:07.32 | vader-- | is there a list of features that are in sccp and sip? |
| 23:07.35 | XARiUS | I like the 7940's, but personally $200 for a phone capable of only 2 registrations and NO speed dials.. sucks. |
| 23:07.39 | vader-- | on the cisco 7940/60 phones |
| 23:07.43 | vader-- | like a comparison |
| 23:07.52 | Strom_C | pfft, speed dial |
| 23:08.01 | XARiUS | dudeeee, users WANT speed dials. |
| 23:08.04 | XARiUS | sadly. |
| 23:08.07 | Qwell[] | speeddial is easy :p |
| 23:08.32 | Strom_C | speed dial was important in 1964 when everyone had rotary phones |
| 23:08.37 | XARiUS | uh, 7940 only has 2 buttons.. gonna glue some more on there qwell? :) |
| 23:08.47 | Qwell[] | CP-7940G= Cisco IP Phone 7940G, Global, Spare 1 USD 172.00 |
| 23:08.58 | vader-- | qwell used or new? |
| 23:09.00 | Qwell[] | wholesale list |
| 23:09.27 | smackus | speed dial for us is TACI |
| 23:09.45 | smackus | http://www.voip-info.org/wiki-Asterisk+click+to+call |
| 23:10.30 | XARiUS | yeah.. I deal mainly in small office environments.. <10 phones. Of course, these fancy voip phones replace cheap $50 desk phones that have 20 configurable speed dials.. |
| 23:10.41 | XARiUS | so, I always take flack for that. |
| 23:10.42 | Qwell[] | XARiUS: It has 6 softkeys, with a "more" button... |
| 23:10.45 | Qwell[] | speeddial is easy |
| 23:10.50 | XARiUS | the 7960 does, yes. |
| 23:10.53 | XARiUS | the 7940 has 2 |
| 23:10.55 | Qwell[] | no, both do |
| 23:10.55 | Strom_C | so tell your users to learn to dial faster |
| 23:11.04 | Qwell[] | or was it 4? whichver |
| 23:11.07 | Qwell[] | whichever |
| 23:11.13 | XARiUS | I've got a 7940G right next to me, it has 2 buttons on it. |
| 23:11.18 | Qwell[] | softkeys |
| 23:11.28 | XARiUS | we're talking SIP here. |
| 23:11.29 | vader-- | qwell is there a list that compares the asterisk sccp functions/features to sip's? |
| 23:11.29 | XARiUS | remember. :) |
| 23:11.50 | Qwell[] | XARiUS: No, you're talking SIP ;) |
| 23:11.52 | vader-- | the 7940G has 2 buttons and the 7960G has 6 buttons |
| 23:11.55 | Qwell[] | I talk non-shit firmware |
| 23:11.57 | XARiUS | lol touche. |
| 23:12.01 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 23:12.33 | XARiUS | okay dude, so then please enlighten me.. if the sccp implementation sucks ass... and the sip firmware sucks ass... |
| 23:12.38 | XARiUS | what are the high and low points of going totally with sccp? |
| 23:12.48 | XARiUS | (seriously) |
| 23:12.55 | Qwell[] | You can do basically whatever the hell you want, with sccp |
| 23:12.58 | XARiUS | I have a spare 7960g here I can throw sccp on, if it's worth the experience :) |
| 23:13.14 | Qwell[] | If you can code...have a blast with it |
| 23:13.24 | vader-- | i wanna figure out how to stop this from happening |
| 23:13.24 | vader-- | http://www.thepokerhq.net/cisco.jpg |
| 23:13.39 | Qwell[] | vader--: don't use sip firmware |
| 23:13.40 | Qwell[] | :p |
| 23:13.44 | Qwell[] | or, at least, not the 8.x |
| 23:13.49 | XARiUS | need to learn how to hold a camera, it's all crooked. |
| 23:13.54 | CunningPike | Put your phone back right way up |
| 23:13.54 | vader-- | where the extensino shows up on the caller id i keep getting extensino or caller id number @172.13.0.2 my asterisk box's ip address |
| 23:14.05 | XARiUS | lol |
| 23:14.09 | Lord_Drachenblut | someone told me there is a cheap modem that can be used as a fxo or fxs card with asterisk anyone know about this card |
| 23:14.14 | XARiUS | obviously the problem, phone must be counterbalanced :) |
| 23:14.14 | Qwell[] | vader--: known "bug" in 8.x |
| 23:14.18 | Strom_C | Lord_Drachenblut: it sucks |
| 23:14.21 | vader-- | and that crap at the bottom |
| 23:14.22 | Strom_C | Lord_Drachenblut: dont waste your time |
| 23:14.27 | vader-- | where is shows (out 180) |
| 23:14.31 | vader-- | wtf is up with that |
| 23:14.38 | Qwell[] | vader--: sip code 180? |
| 23:14.41 | vader-- | ya |
| 23:14.41 | Lord_Drachenblut | Strom_C, any good cheap altetenatives |
| 23:14.45 | Qwell[] | you don't |
| 23:14.51 | vader-- | i can't stop that from showing up |
| 23:14.58 | Qwell[] | see above |
| 23:14.58 | XARiUS | Qwell, if I'm trying to cookie cut small business phone systems.. and don't want to spend my profit time coding away at sccp, should I still use sccp? :) |
| 23:14.59 | Strom_C | Lord_Drachenblut: Sipura SPA-3000 or get a TDM400P |
| 23:15.08 | vader-- | hmmm |
| 23:15.12 | Qwell[] | XARiUS: If they want features that no other phones can provide |
| 23:15.16 | vader-- | i already have 8.0 loaded on alot of phones |
| 23:15.24 | vader-- | i don't know how easy it would be to drop back to 7.1 |
| 23:15.27 | XARiUS | Features such as...? |
| 23:15.28 | Qwell[] | very |
| 23:15.32 | Qwell[] | XARiUS: whatever you can think of |
| 23:15.41 | XARiUS | Okay well, whats one you've done that you're proud of? |
| 23:15.47 | Dr-Linux | Qwell[]: did you got my email on Chan_SCCP maling list? |
| 23:15.53 | Qwell[] | Dr-Linux: yes, you got a response |
| 23:15.56 | Lord_Drachenblut | Strom_C, what's so bad bout it |
| 23:16.19 | Dr-Linux | Qwell[]: wow great, but i can't check my official mail from home :( |
| 23:16.23 | Strom_C | Lord_Drachenblut: echo, poor quality, bugs, hard to get playing well with other cards |
| 23:16.28 | Qwell[] | Dr-Linux: You can look at the list archives |
| 23:16.41 | Lord_Drachenblut | Strom_C, i see well thanks for the info |
| 23:16.43 | Dr-Linux | Qwell[]: how? |
| 23:16.47 | Strom_C | Lord_Drachenblut: more people come in here with problems with that card than with any other card |
| 23:16.50 | Dr-Linux | Qwell[]: link please |
| 23:16.55 | Qwell[] | http://lists.berlios.de/pipermail/chan-sccp-users/ |
| 23:17.00 | Lord_Drachenblut | Strom_C, thanks |
| 23:17.10 | Lord_Drachenblut | Strom_C, and how you been by the way |
| 23:17.15 | Strom_C | Lord_Drachenblut: been good |
| 23:17.17 | Strom_C | but busy |
| 23:17.42 | droops | busy is good |
| 23:17.42 | Qwell[] | XARiUS: how about a softkey that orders pizza? |
| 23:17.42 | Qwell[] | ...literally |
| 23:17.42 | vader-- | SIP Image 8.3 fixes the problem im talking about |
| 23:17.42 | Lord_Drachenblut | Strom_C, kewl deal |
| 23:17.44 | *** join/#asterisk Jimerson (n=Jim@c-71-200-9-249.hsd1.md.comcast.net) |
| 23:17.52 | Qwell[] | XARiUS: like I said...anything you can think of |
| 23:17.56 | XARiUS | Qwell: If it can bring hookers, and pizza also, I'm totally down for supporting your cause. |
| 23:17.57 | XARiUS | ;) |
| 23:18.43 | XARiUS | OKie, i'm gonna see if I can get this presence stuff working on this 501. |
| 23:18.45 | XARiUS | I'm intrigued. |
| 23:19.00 | Lord_Drachenblut | Strom_C, is it fxo or fxs i want to connecto to pots |
| 23:19.05 | Strom_C | fxo |
| 23:19.08 | Qwell[] | fxq |
| 23:19.14 | Strom_C | fxx |
| 23:19.40 | Lord_Drachenblut | Strom_C, any recommendations for a single port fxo card low price or is it still the 2 you said earlier |
| 23:19.49 | Strom_C | Lord_Drachenblut: Sipura SPA-3000 or get a TDM400P |
| 23:19.52 | Dr-Linux | Qwell[]: :( |
| 23:20.23 | Dr-Linux | Qwell[]: who responded me? |
| 23:20.23 | Dr-Linux | you? |
| 23:20.24 | Qwell[] | no |
| 23:20.29 | Qwell[] | Ryan Amos? |
| 23:20.30 | Dr-Linux | Qwell[]: so that means, there is no solution? |
| 23:20.49 | Qwell[] | not unless Sergio gets off his ass, I guess... |
| 23:21.14 | Dr-Linux | Qwell[]: who the fuck is Sergio? |
| 23:21.18 | *** join/#asterisk philth|work (n=ceac2822@d38-179-126.home1.cgocable.net) |
| 23:21.22 | Qwell[] | the guy who pretends the maintain chan_sccp |
| 23:21.26 | Qwell[] | s/the/to/ |
| 23:21.42 | Strom_C | haha |
| 23:22.05 | philth|work | <PROTECTED> |
| 23:22.09 | Qwell[] | notice that he hasn't sent a single message to the list since May 28th |
| 23:22.34 | Dr-Linux | Qwell[]: i hope he read the mailing list, i'll mail him |
| 23:22.37 | XARiUS | Qwell: I'm starting to understand where your.. excitment/sarcasm eminates from. :) |
| 23:22.43 | Qwell[] | XARiUS: ;) |
| 23:22.48 | Qwell[] | XARiUS: You have no idea |
| 23:22.50 | XARiUS | lol |
| 23:23.05 | Qwell[] | He's a fool, plain and simple |
| 23:23.20 | Dr-Linux | Qwell[]: what if i change the chan_SCCP version? does that help? |
| 23:23.26 | Dr-Linux | if not what you suggest? |
| 23:23.52 | Qwell[] | Dr-Linux: I don't know...under normal circumstances, I'd fix it for you, but... |
| 23:24.28 | Dr-Linux | lol |
| 23:25.34 | XARiUS | whats wrong, paypal account maxed out? :) |
| 23:25.38 | XARiUS | afk |
| 23:26.03 | Qwell[] | XARiUS: nah, I don't touch chan_sccp |
| 23:26.44 | *** join/#asterisk [Airwolf] (n=airwolf@84.246.5.160) |
| 23:29.09 | *** join/#asterisk monkey13 (n=monkee13@71.194.25.60) |
| 23:29.16 | XARiUS | @()$*$ |
| 23:30.06 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 23:30.20 | [Airwolf] | XARiUS, i found another softphone |
| 23:30.24 | [Airwolf] | For windows mobile |
| 23:30.47 | [Airwolf] | But it's IAX2 and it's a demo version with a 2 minute limit for a call |
| 23:31.24 | XARiUS | I'll take iax2 over sip for portability and unknown routers/firewalls anyday |
| 23:40.24 | XARiUS | gah |
| 23:40.55 | XARiUS | "hint" doesn't even seem to be in my asterisk build =/ |
| 23:41.24 | *** join/#asterisk Coeus (n=Coeus@ip24-255-125-43.dc.dc.cox.net) |
| 23:41.35 | *** join/#asterisk anthm (n=anthm@000-423-595.area4.spcsdns.net) |
| 23:41.35 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:42.12 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
| 23:44.02 | r_evolution | wake up clowns |
| 23:44.34 | *** join/#asterisk korihor (n=humberto@200.35.213.46) |
| 23:44.51 | Qwell[] | XARiUS: got something released within the last 8 years? |
| 23:45.11 | *** join/#asterisk eKo1 (n=eKo1@65.169.159.98) |
| 23:45.16 | r_evolution | Qwell... are you bashing debian again? |
| 23:45.19 | r_evolution | dirty heretic. |
| 23:45.24 | XARiUS | yeah lol last week actually |
| 23:45.46 | XARiUS | SVN-branch-1.2-r36377 |
| 23:48.29 | [Airwolf] | XARiUS, I don't mind it's iax, but the time limit and no information on how to purchase that application. |
| 23:48.52 | jhiver | aaah, fuck, compiling h323 is a total bitch |
| 23:48.55 | luke-jr_ | Anyone know a company with servers available on colocation with very low latency? |
| 23:49.17 | luke-jr_ | jhiver: nah, just USE=h323 emerge asterisk =p |
| 23:49.18 | XARiUS | airwolf sounds like they stopped working on it then :( |
| 23:49.29 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 23:49.52 | jhiver | luke-jr_, it's that easy on gentoo? too bad I can't install that distro on my server |
| 23:50.19 | jhiver | ~seen jerjer |
| 23:50.31 | jbot | jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #asterisk, 75d 4h 54m 39s ago, saying: 'but that's cvs code from like pre-1.2'. |
| 23:50.32 | luke-jr_ | jhiver: well, the compiling/installing part-- I haven't bothered to configure it |
| 23:50.32 | jhiver | help! |
| 23:50.32 | jhiver | :) |
| 23:50.32 | *** part/#asterisk n3^ (n=n3@63-253-43-58.ip.mcleodusa.net) |
| 23:50.42 | jhiver | oh well that answers it :l) |
| 23:51.15 | rene- | hi, how do i tell asterisk makefile to create a startup script for me? |
| 23:51.25 | luke-jr_ | probably wouldn't be too hard-- just stick a few lines in h323.conf or whatever |
| 23:52.07 | Dr-Linux | rene-: make config |
| 23:52.15 | r_evolution | still on that h323 jhiver? |
| 23:52.19 | rene- | cool |
| 23:52.37 | *** part/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 23:53.27 | jhiver | r_evolution, I've given up compiling h323 |
| 23:53.33 | jhiver | it's just totally fucked |
| 23:53.51 | r_evolution | :( |
| 23:54.05 | r_evolution | WOW it's been that long since jerjer was in here? |
| 23:54.12 | r_evolution | I'm sure he's still on MIRC tho |
| 23:54.15 | r_evolution | try PMing him |
| 23:54.25 | jhiver | dah |
| 23:54.29 | jhiver | forget it |
| 23:54.35 | r_evolution | i've talked to him since 75d ago |
| 23:56.00 | jhiver | yeah well I'll try something else |
| 23:56.11 | r_evolution | tell em to migrate to sip |
| 23:56.13 | r_evolution | or die :) |
| 23:56.34 | smackus | can anyone tell me why when i go to voicemail at the extension I am dialing, it still goes to "/var/spool/asterisk/voicemail/progrexion/156/tmp"? |
| 23:56.40 | smackus | what am I still doing wrong? |
