| 00:00.13 | wgroh | no calls to any females at least |
| 00:00.15 | tzanger | the rev2 cards all have hardware hdlc and dtmf detection, anda coouple other things IIRC |
| 00:00.20 | tzanger | wgroh: yep |
| 00:00.24 | Qwell[] | oh |
| 00:00.31 | tzanger | you disable the hardware dtmf support and that goes away |
| 00:00.50 | wgroh | vpmdtmfsupport |
| 00:00.52 | wgroh | I saw that somewhere |
| 00:01.20 | wgroh | cool, I'll have to try that instead of an IVR before all calls |
| 00:01.34 | wgroh | that tells people they cannot call women or michael jackson |
| 00:02.13 | tzanger | wgroh: haha |
| 00:02.18 | tzanger | I never tried that :-) |
| 00:02.31 | wgroh | ;) |
| 00:02.47 | Hmmhesays | i don't know if I can ever play this solo |
| 00:02.55 | tzanger | play what solo |
| 00:03.00 | Hmmhesays | picking 32nd notes is a bitch |
| 00:03.08 | tzanger | hahaha |
| 00:03.13 | Hmmhesays | "bat country" by avenged sevenfold |
| 00:04.03 | tzanger | ok that's funny |
| 00:04.21 | tzanger | Rick Mercer's got the Environment Minister drilling holes and driving spiles into maple trees |
| 00:05.06 | tzanger | she's from alberta... he asks her "do you do this in alberta?" and she says "yeah but something else comes out of the trees there" |
| 00:05.23 | brif8 | X-Rob: any ideas why this chan_iax2.c: Max retries exceeded to host 71.41.50.162 on IAX2/lecanto-16384 (type = 6, |
| 00:08.03 | Cybertoy | websae, I'm using voipdiscount... |
| 00:08.07 | Cybertoy | ... you asked earlier. |
| 00:11.12 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
| 00:11.48 | melange8272 | anyone using sipphone for conf? |
| 00:12.06 | Hmmhesays | who wants to buy me a new guitar |
| 00:12.23 | Cybertoy | Fender? |
| 00:12.55 | Hmmhesays | i'd rather have a hamer or an ibanez |
| 00:13.29 | melange8272 | Gibson all the way |
| 00:13.47 | Hmmhesays | i don't really care for my les paul |
| 00:14.09 | Cybertoy | can it do SIP? |
| 00:14.17 | Shaun2222 | i remember theri was a key i could press on teh 7960 phones to reset them to factory defaults, anybody remember what that was |
| 00:14.22 | melange8272 | new pickups.. |
| 00:14.24 | Shaun2222 | you held a key while power cycling it |
| 00:14.29 | Cybertoy | yeah ... # key |
| 00:14.33 | Hmmhesays | yeah hold down # |
| 00:14.37 | Cybertoy | and then 123456789*0# |
| 00:14.47 | Shaun2222 | thats right thanks |
| 00:15.16 | Cybertoy | shaun, do you have daylight savings time on your phones? |
| 00:15.31 | Shaun2222 | uhh |
| 00:15.32 | Cybertoy | my 7970 still shows "regular" eastern time |
| 00:15.36 | Shaun2222 | i dont know |
| 00:15.44 | Shaun2222 | my phone doesnt even show the time right now |
| 00:15.52 | Shaun2222 | i just got them. |
| 00:16.02 | Shaun2222 | it would be nice if it showed the time/date though :) |
| 00:18.15 | eric_hill | Anyone know how to do queue auto-logoff with dynamic queue members? |
| 00:19.27 | Shaun2222 | holding # doesnt seam to be doing it... |
| 00:19.34 | Shaun2222 | this phone has a older sip firmware |
| 00:19.38 | Shaun2222 | 6 somthing |
| 00:19.49 | Shaun2222 | i know i've done this on version 8.2 |
| 00:21.19 | *** part/#asterisk melange8272 (n=melange8@ool-4576ab1f.dyn.optonline.net) |
| 00:23.15 | *** join/#asterisk rahool (n=Rahool@ppp-70-226-84-69.dsl.klmzmi.ameritech.net) |
| 00:24.48 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 00:26.01 | *** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju) |
| 00:28.20 | *** join/#asterisk plecebo (n=larry@D-128-208-215-137.dhcp4.washington.edu) |
| 00:44.00 | Hmmhesays | whats the deal, with my brain, why am I so obviously insane |
| 00:44.17 | skyboy | curious question...does AMP work with the * or just aah |
| 00:45.36 | skyboy | Hmmmhesays: likely because with asterisk there is plenty of risk and you end with very little ass ;) |
| 00:49.57 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
| 00:52.30 | lokkju | ok, I am not getting any sound, when I use Playback(). When I use Festival, I can hear it perfectly. There are no error messages regarding it in the full log. the only references in the full log are: "Apr 11 12:46:22 DEBUG[383] channel.c: Scheduling timer at 160 sample intervals" and "Apr 11 12:46:22 VERBOSE[383] logger.c: -- Playing 'vm-theperson' (language 'en')". Any suggestions? |
| 00:52.53 | *** join/#asterisk Vyeperman^2 (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
| 00:53.11 | Az_au | i had a problem the opposite way around where festival wouldn't play anything (even tho it said it did) but playback was fine |
| 00:53.20 | [hC] | do the polycom ip501's do PoE? |
| 00:53.21 | Az_au | never solved it tho.. wasn't a major goal :P |
| 00:53.22 | lokkju | rofl |
| 00:53.29 | lokkju | damn |
| 00:53.31 | lokkju | :( |
| 00:53.42 | Az_au | i'd say the two conflict somehow |
| 00:53.48 | lokkju | I would rather have that problem... |
| 00:53.54 | lokkju | eh, seriously? |
| 00:53.59 | lokkju | I could disable festival |
| 00:54.31 | hypa7ia | hey [hC] |
| 00:54.33 | Az_au | another guy i know has it working fine with both but i don't have access to his config atm |
| 00:54.45 | lokkju | hmm |
| 00:55.11 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 00:55.25 | [hC] | heyy hypa7ia |
| 00:56.12 | [av]bani | ~phones |
| 00:56.13 | jbot | [phones] at http://bani.anime.net/phones/ |
| 00:56.31 | [av]bani | the answer is ... no |
| 00:56.54 | lokkju | Az_au, no luck |
| 00:57.20 | [hC] | hm, i thought the 501 had some sort of PoE cable adapter. |
| 00:57.33 | [av]bani | ip501 does not "do poe" |
| 00:57.43 | [av]bani | you can buy an injector, but it does not do poe natively |
| 00:57.53 | [av]bani | "do poe" means the phone itself does poe, |
| 00:57.54 | [hC] | nod, that was a stupid question now that i think about it. |
| 00:57.55 | [hC] | heh. |
| 00:57.56 | lokkju | [hC], there is a diff between "doing PoE" and supporting PoE adaptors |
| 00:57.59 | [hC] | yes yes |
| 00:58.00 | [hC] | i get it |
| 00:58.03 | [av]bani | otherwise you could say that any phone in the universe "does poe" |
| 00:58.06 | [hC] | thank you for rubbing my nose in it |
| 00:58.08 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
| 00:58.10 | [av]bani | no problem |
| 00:58.12 | [hC] | :) |
| 00:58.22 | lokkju | anything that is 5V or less will support PoE adaptors, unless it is very power thirst |
| 00:58.24 | [av]bani | you're on a roll today |
| 00:58.28 | [hC] | OK!! |
| 00:58.29 | [av]bani | first rfc2833, now poe |
| 00:58.40 | lokkju | hmm |
| 00:58.42 | [hC] | this is the first day in like, 2 months, that i got to work at 9am |
| 00:58.44 | [av]bani | what next? |
| 00:58.48 | [hC] | im having neural misfires today |
| 00:58.52 | lokkju | that list does not have wifi voip phones :( |
| 00:59.07 | [hC] | i just ordered my first set of linksys wip300's |
| 00:59.07 | [av]bani | lokkju: omg wifi phone with poe! |
| 00:59.21 | [av]bani | :)) |
| 00:59.31 | lokkju | [av]bani, uh, sorry, was that list only for which supported PoE? |
| 00:59.33 | [hC] | hahahaha |
| 01:02.16 | key2 | PoE = Power Over Ethernet ? |
| 01:02.20 | lokkju | this is so agrivating, not even error messages to help me |
| 01:02.26 | lokkju | key2, yes |
| 01:03.20 | *** part/#asterisk rva (n=rafa@200.210.51.130) |
| 01:04.25 | asterboy | Is anyone in here Chinese? |
| 01:05.34 | asterboy | Does anyone know what the 2 lines of 3 logograms on the GXP-2000 represent? |
| 01:07.54 | asterboy | This is the first time I have been in possesion of electronics with Chinese writing embedded in the crystal display. (it |
| 01:08.15 | timscott | O_O |
| 01:08.18 | asterboy | 's on the right hand side clearly visible with the backlight on. |
| 01:08.23 | timscott | O_o |
| 01:08.26 | timscott | I always wondered that myself |
| 01:08.42 | timscott | Probably says, "America - Owned by china." or something equally hilarious |
| 01:08.51 | asterboy | LolL |
| 01:09.04 | asterboy | That is so true though. |
| 01:09.22 | asterboy | They have a false economy, built on only importation. |
| 01:09.59 | asterboy | Bush keeps asking for more money, they keep lending it to him and taxes are going to need to be raised. |
| 01:10.41 | asterboy | but Iraq will be a good tit to suck for a while. |
| 01:12.02 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 01:13.29 | [hC] | soo.. looking at the ip501's spec sheet and install guide |
| 01:13.40 | [hC] | it looks like the polycom ip501 does actually do PoE |
| 01:13.51 | [hC] | but that the phone doesnt negotiate anything like 802.3af or CDP |
| 01:14.00 | [hC] | it can just accept power on the wire, like if you were to used an injector |
| 01:14.13 | [hC] | so their special cable has 802.3af capabilities built on a chip which is inline |
| 01:14.16 | [hC] | and then sends power to your phone |
| 01:17.04 | [av]bani | 'tis a silly phone |
| 01:17.55 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 01:17.59 | timscott | Subdolus! |
| 01:18.02 | timscott | Fancy seeing you here. |
| 01:18.09 | war_ | Hello there. |
| 01:18.11 | subdolus | well hello war! |
| 01:18.15 | subdolus | :) |
| 01:18.15 | timscott | Hello. :) |
| 01:18.21 | subdolus | how's things |
| 01:18.23 | timscott | I never was able to get through to that conference |
| 01:18.26 | timscott | things are good, busy busy |
| 01:18.29 | timscott | yourself? |
| 01:18.38 | subdolus | much the same, much the same |
| 01:18.40 | subdolus | having a day off :) |
| 01:18.43 | timscott | :) |
| 01:19.04 | timscott | i'm still at work myself. :S |
| 01:19.07 | timscott | worked late tonight |
| 01:19.17 | subdolus | haha yikes |
| 01:21.02 | *** join/#asterisk shmooz (n=shmooz@H142.C72.B0.tor.eicat.ca) |
| 01:24.30 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
| 01:28.10 | *** join/#asterisk MacDome (n=eseidel@A17-255-104-111.apple.com) |
| 01:28.21 | lokkju | grr, anyone know off the top of their head how to choose the sip proxy you want to dial from with x-lite? |
| 01:28.31 | asterboy | no instructions at all for for the GXP-2000 |
| 01:28.33 | lokkju | like if I have 4 configured |
| 01:28.39 | asterboy | Good thing I have Internet |
| 01:28.39 | *** join/#asterisk vopi (n=kkk@202.139.210.17) |
| 01:31.09 | *** join/#asterisk Darkhalf (n=darkhalf@cpe-72-130-156-112.san.res.rr.com) |
| 01:38.02 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 01:38.32 | key2 | is there any quad GSM card working with asterisk ? |
| 01:42.33 | Az_au | lokkju: on eyebeam you dial #1 or #2 etc to choose which account to use |
| 01:43.03 | Az_au | asterboy: there is some doco you can download from the grandstream site but its not too comprehensive |
| 01:45.12 | *** join/#asterisk Utah_Dave (n=boucha@12.118.109.86) |
| 01:48.58 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 01:49.59 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 01:57.24 | *** join/#asterisk angom_h (n=angom@red-corp-201.130.165.94.telnor.net) |
| 02:00.32 | *** join/#asterisk trbldwine (i=trbldwin@71.194.161.170) |
| 02:03.15 | *** join/#asterisk tengulre11 (n=tengulre@61.185.224.66) |
| 02:04.26 | *** join/#asterisk cced (n=dev2003@222.33.36.205) |
| 02:05.53 | Hmmhesays | so i'm going out with this girl again tonight |
| 02:05.54 | Hmmhesays | crazy |
| 02:06.50 | Nugget | I presume chan_sccp2 is the channel module I want to be using, correct? Also, can sccp traverse nat (where server is public but phone is behind nat) |
| 02:07.27 | Qwell | Nugget: chan-sccp.berlios.de, BUT |
| 02:07.40 | Qwell | You must try chan_skinny first, from 6859 |
| 02:07.54 | Nugget | whyfor? |
| 02:08.09 | Qwell | to test it |
| 02:08.13 | Nugget | oic |
| 02:08.24 | wunderkin | it is part qwell's bitch now |
| 02:08.47 | Nugget | I've got the phone connecting via sccp, but I'm getting one-way audio. and the display is all horked with buttons that say <p>  and stuff. |
| 02:09.04 | Qwell | Nugget: nice... |
| 02:09.04 | Nugget | presumably my tftp server doesn't contain everything it needs |
| 02:09.09 | wunderkin | that sounds cool |
| 02:09.11 | Qwell | To fix that, you need to update the locales or something |
| 02:11.07 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 02:12.44 | Qwell | wunderkin: You got that backwards. |
| 02:12.49 | Qwell | chan_skinny is making me it's bitch |
| 02:12.57 | Qwell | Nugget: pfft, that is awful |
| 02:13.00 | Nugget | if I can't get this nat issue figured out, though, I'm back to sip. |
| 02:13.03 | Qwell | mine was so much better :p |
| 02:13.10 | wunderkin | forced to submission |
| 02:13.18 | Qwell | I never sent him that patch though...SOB |
| 02:15.40 | *** join/#asterisk Leob (n=chatzill@w2kvpn-22.media.mit.edu) |
| 02:15.46 | Cherebrum | Qwell: Are you using a Cisco MickeyMouse Disneyland POS 79XX? |
| 02:16.00 | Qwell | eh? |
| 02:16.08 | Cherebrum | a 7960 or 7940? |
| 02:16.15 | timscott | hahahaha |
| 02:16.22 | timscott | "Cisco MickeyMouse Disneyland POS" |
| 02:16.25 | timscott | Very slick. :) |
| 02:16.30 | Cherebrum | heh heh |
| 02:16.31 | Qwell | It's P0S |
| 02:16.35 | Qwell | get it right |
| 02:17.16 | Nugget | heh |
| 02:19.54 | Qwell | note to self: cisco firmware humor doesn't go over well |
| 02:22.17 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-174-51.lsanca.fios.verizon.net) |
| 02:24.53 | Leob | guys 'odbc connect' keeps crashing my system, even though odbc settings seem to be fine... any ideas of what is going on? |
| 02:29.34 | *** join/#asterisk awannabe (n=gti@ip24-251-150-76.ph.ph.cox.net) |
| 02:30.13 | awannabe | hello all, i may sound really dumb, but i cant find any docs on howto setup a auto attendant!! can anyone help me out? |
| 02:30.46 | Jaxxan | ~wiki |
| 02:30.56 | Az_au | tried http://www.voip-info.org/wiki/view/Asterisk+tips+autoattendant ? |
| 02:31.38 | awannabe | i saw that, but thats all i really found, that seems to use a DB backend |
| 02:33.04 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
| 02:33.32 | Az_au | what exactily are you trying to achieve? |
| 02:35.30 | awannabe | well, just setup a autoattendant, right now im just toying, wanted to setup a test multilevel AA |
| 02:35.51 | awannabe | i just dont have asterisk with DB support yet, i need to do it, but wanted to learn all the basics first, then go from tehre |
| 02:37.22 | Az_au | try looking up DigitTimeout ResponseTimeout Background commands, they are used for such things |
| 02:37.47 | Az_au | there should be examples for them that give you what you are after |
| 02:38.16 | awannabe | ok great, ill check that out |
| 02:38.19 | awannabe | thanks :) |
| 02:41.07 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 02:41.39 | brodiem | Anyone use AsterFax? Wondering if it's reliable, or if there's a better email-to-fax gateway |
| 02:43.54 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 02:44.10 | awannabe | i guess in realitiy, a AA just plays a greating, and then waits for a option to be answered |
| 02:44.12 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
| 02:45.51 | gammacoder | brodiem: not sure yet, I'm rolling out a test asterfax this week |
| 02:47.07 | brodiem | gammacoder, I guess I will be too |
| 02:49.27 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 02:53.15 | asterboy | hylafax works well |
| 02:55.04 | brodiem | asterboy, is it more for connecting physical fax machines? |
| 02:56.24 | asterboy | why do you need asterfax then? |
| 02:56.43 | asterboy | just to convert to email? |
| 02:56.43 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
| 02:58.55 | asterboy | hylafax does a good job of that. |
| 02:59.00 | asterboy | converts to pdf too. |
| 03:00.16 | asterboy | I wouldn't mind trying AsterFax as long as it does not interfere with my voice communications. |
| 03:03.45 | *** join/#asterisk Eggplant (i=No@dsl-469.cascadeaccess.com) |
| 03:06.09 | *** join/#asterisk Thock (i=Landon@ip70-162-89-46.ph.ph.cox.net) |
| 03:06.13 | Thock | Howdy everyone |
| 03:06.36 | Thock | Anyone mind giving a VoIP newbie some pointers? |
| 03:06.36 | asterboy | made in china |
| 03:06.49 | Strom_M | RIGHT THERE |
| 03:06.49 | asterboy | sure |
| 03:06.54 | Strom_M | AND THERE |
| 03:07.03 | Thock | NOT THERE, THERE. |
| 03:07.05 | asterboy | and HERE! |
| 03:07.08 | asterboy | ~docs |
| 03:07.10 | jbot | docs is probably 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 |
| 03:07.10 | Thock | Ha haa, Pointers. |
| 03:07.11 | brookshire | NOT HERE |
| 03:07.47 | asterboy | Read over there and there. |
| 03:07.53 | asterboy | and over there. |
| 03:07.53 | Thock | jbot: I've looked through the wiki and the oreilly book. I just need a bit of clarifcation on a few things. |
| 03:08.15 | asterboy | oh oh, someone is talking to jbot |
| 03:08.22 | Thock | just stating :( |
| 03:08.31 | asterboy | you don't want to piss off jbot |
| 03:09.15 | Thock | They are more of hardware questions than anything. Really luser stuff. :/ |
| 03:09.20 | Thock | -ly |
| 03:09.44 | brookshire | ~thwap thock |
| 03:09.46 | jbot | ACTION thwaps thock on the nose with a 2 by 4 |
| 03:09.51 | Thock | :( |
| 03:09.51 | brookshire | :D |
| 03:10.05 | brookshire | jbot is a bot |
| 03:10.10 | Thock | I realized. |
| 03:10.17 | Strom_M | digium: where it's always a party |
| 03:10.18 | brookshire | k.. just checking :D |
| 03:10.24 | asterboy | my gxp-2000 came with no instructions...they don't want you knowing how to use their hardware. |
| 03:10.26 | brookshire | (in strom's pants) |
| 03:10.30 | Strom_M | hahahahaha |
| 03:10.49 | asterboy | ewwww |
| 03:10.59 | [TK]D-Fender | Strom_M, asterboy : Geez man... let the poor fellow embarrass himself before you start the reaming, ok?! |
| 03:11.23 | Strom_M | hey, I'll gladly take the reaming |
| 03:11.27 | [TK]D-Fender | Thock : Ok, what do you want to know? |
| 03:11.36 | brookshire | and by reaming.. he means rimming |
| 03:11.41 | Strom_M | ahahahhha |
| 03:11.41 | [TK]D-Fender | Strom_M : Well you sem too busy GIVING it... |
| 03:11.49 | asterboy | thock, it's kinda like a buffet in here...you need to serve your self or place a specific order |
| 03:12.03 | Thock | the boss wants 20 voip lines, and around 4-6 POTS lines. Configuration of asterisk isn't really the problem, its figuring out the hardware and how it all networks together. I've been working on getting some sort of diagram so HE will understand it, but first i've gotta figure it out. :/ |
| 03:12.14 | *** join/#asterisk snoopjohn (n=jscott@gateway.digium.com) |
| 03:12.20 | Strom_M | POTS lines: TDM2400P |
| 03:12.24 | brookshire | snoopjohn! |
| 03:12.26 | Thock | I'm just trying to figure out what sort of hardware i'll need to peice together. |
| 03:12.26 | Strom_M | voip lines: ethernet card |
| 03:12.31 | [TK]D-Fender | but 20 voip lines are you reffering to PHONES, or lines support incoming/outgoing calls to you PBX? |
| 03:12.35 | snoopjohn | brookshire! |
| 03:12.36 | asterboy | thock, get the Bible! |
| 03:12.43 | Thock | [TK]D-Fender: actual Lines, not phones. |
| 03:12.51 | asterboy | thock, read the Bible! |
| 03:12.57 | [TK]D-Fender | Thock : Where are you located? |
| 03:13.05 | Thock | Phoenix, Az |
| 03:13.07 | asterboy | ~bible |
| 03:13.09 | jbot | methinks bible is see 'thebook' |
| 03:13.09 | brookshire | omg |
| 03:13.18 | asterboy | ~thebook |
| 03:13.19 | jbot | somebody said thebook was Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 03:13.35 | Thock | already read through the book, i'm still a little unclear |
| 03:13.39 | file[laptop] | !!?!! |
| 03:13.44 | [TK]D-Fender | Thock : Ok, well that is a LOT of VoIP channels, and bandwidth may very likely be a concern. What kind of connection would you be running it on? |
| 03:13.58 | brookshire | thock: how can we help you sir? |
| 03:14.05 | Strom_M | 20 voip channels? aggregate a pair of T1s or get a DS3 |
| 03:14.06 | brookshire | (or ma'am) |
| 03:14.16 | Thock | two T1's. Our current setup is two T1's, through Quest. One for long distance and one for local. |
| 03:14.22 | Thock | around 7 lines of POTS |
| 03:14.24 | [TK]D-Fender | Strom_M : a single dedicated T1 would do.... |
| 03:14.30 | Strom_M | dontcha mean Qwest? |
| 03:14.34 | asterboy | better yet, order it and keep it by the toilet so whenever you have to clean out your days worth of existance, you can fill up your mind at the same time with knowledge that is * |
| 03:14.34 | Strom_M | [TK]D-Fender, not if they want ulaw |
| 03:14.35 | Thock | Right. That. |
| 03:14.43 | brodiem | asterboy, no email to fax |
| 03:14.53 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 03:14.56 | asterboy | with what, asterfax? |
| 03:15.00 | Thock | Our main concerns are Confrencing (around 8 total, maximum) |
| 03:15.03 | asterboy | hylafax does it |
| 03:15.25 | [TK]D-Fender | Thock : I'm a little unclear about how you use your current 2 T1... |
| 03:15.48 | asterglide | having good fax? |
| 03:15.52 | [TK]D-Fender | [hC] : So how's the MB stuff working out for you? |
| 03:15.53 | *** part/#asterisk asterglide (n=jscott@gateway.digium.com) |
| 03:16.10 | Thock | [TK]D-Fender: As i am. I was given this as a task yesterday due thursday, and i've tried to do as much reading as possible, the wiki, the book, etc. I'm just trying to figure out the costs of hardware. What's reccommended by the community, etc. |
| 03:16.12 | brodiem | has anyone compared hylafax vs asterfax? |
| 03:16.30 | [hC] | just got home, just about to try it out now, since i left my 601 at hoje |
| 03:16.32 | [hC] | home |
| 03:16.39 | *** join/#asterisk Cherebrum (n=jgarland@ares.v2business.com) |
| 03:16.50 | asterboy | I heard asterfax is great when and if yo can get it running. |
| 03:17.03 | Thock | [TK]D-Fender: From what i understand, one t1 is for data/voice, and the other is strictly for long distance. |
| 03:17.05 | key2 | !seen kram |
| 03:17.10 | [TK]D-Fender | Thock : lets say $2500 for the 4 port T1 card w/ HWEC, and whatever you feel like using as a server. You didn't mention qty / type of phones... |
| 03:17.29 | vopi | Hello , I have 3 sip account from same provider in my asterisk server , I registered it all |
| 03:17.30 | vopi | . I tested from sip fone to my server , look like it use only one sip account all time ,I test 2 sip client in same time the first one work well , but the secound failed 603 . anyone have idea for random thats sip account ? |
| 03:17.30 | [hC] | mark's probably too busy for irc these days :) |
| 03:17.34 | [TK]D-Fender | Thock : so taht means you have 2 T1's carrying voice in mixed capacities. |
| 03:17.41 | brodiem | asterboy, I just don't like the fact that it's a beta release, but I know other "stable" software shouldn't be labeled as so either.. |
| 03:18.08 | Thock | [TK]D-Fender: I was hoping to gleam a bit of info how it works, on the network side. The T1 hooks up to what, how are the connections to the phones made, can you use straight twisted pair copper for the endpoint phones or is it ethernet only |
| 03:18.18 | asterboy | then go with hylafax |
| 03:18.24 | Thock | If you can point me to material that covers that sort of thing, please do and i'll be silent, but so far i haven't been able to find anything |
| 03:18.27 | asterboy | very matour |
| 03:18.38 | asterboy | :P |
| 03:18.40 | Strom_M | Thock, if you use one of the TDM cards, you can do twisted pair to analog phones |
| 03:18.54 | Strom_M | or you can use IP phones and plug them into your data network |
| 03:19.08 | [TK]D-Fender | Thock : you can use all sorts of different kinds of equipment with *. The question is what do you WANT to do? buy all IP phones to replace what you've got? Try to reusie old equipement / single pair wiring, etc... |
| 03:19.18 | brodiem | asterboy, do you run it on the same box as *? |
| 03:19.18 | asterboy | or you can use ATAs |
| 03:19.23 | asterboy | yes |
| 03:19.27 | brodiem | asterboy, seems to reference that it wants to be external |
| 03:19.34 | Thock | [TK]D-Fender: I'm sure we'll eventually go full VoIP, phones and all, but the boss wants to keep around 4-6 POTS lines for legacy sake |
| 03:19.37 | Strom_M | Thock, also, your T1 thing doesnt make much sense unless your local and long distance traffic are equally balanced |
| 03:19.42 | asterboy | no need to be external |
| 03:19.58 | [hC] | Thock: you can keep pots lines and use voip phones, you know.. |
| 03:20.09 | Thock | Strom_M: i didn't set up this network, dude. I'm just trying to salvage and set this up as best as i can. Heh. |
| 03:20.14 | Strom_M | heh |
| 03:20.24 | Thock | Seriously |
| 03:20.27 | Strom_M | it's digium field trip to the bar time! later |
| 03:20.39 | Thock | my boss walked into my office, said "go to www.asterisk.org and tell me if we should use this instead of intertel for 30 grand" |
| 03:20.53 | asterboy | I use a splitter before * though and turn on the modem for receive otherwise you need to dedicate a line or have * listen for data and call a port |
| 03:20.59 | asterboy | err...channel |
| 03:21.08 | Strom_M | thock: read the o'reilly book |
| 03:21.10 | Strom_M | ~thebook |
| 03:21.11 | jbot | extra, extra, read all about it, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 03:21.14 | Thock | Strom_M: did. |
| 03:21.17 | Strom_M | seriously, it will help you understand a lot |
| 03:21.23 | [TK]D-Fender | Intertel is a pretty rare digital system my OWN head office was considering before looking at the Avaya IP offic solution funny... |
| 03:21.37 | Thock | I have read it. I just have a few specifics, that's all. |
| 03:21.47 | asterboy | lol Strom, that's exectly what he needs to do. |
| 03:22.07 | asterboy | buy the book...read the book...pray to the book |
| 03:22.37 | brodiem | asterboy, what type of interface are you splitting before *? |
| 03:22.38 | asterboy | you have a few specifics that needs a bunch of specifics to answer |
| 03:22.51 | asterboy | VOIP line |
| 03:23.44 | asterboy | VOIP ----> ATA |-----> Modem |
| 03:23.58 | asterboy | <PROTECTED> |
| 03:24.30 | brodiem | so you're passing fax through SIP/IAX? |
| 03:24.35 | brodiem | and it works? |
| 03:24.41 | asterboy | SIP |
| 03:25.07 | brodiem | I would avoid this whole thing if I could get that to work |
| 03:25.08 | asterboy | and it works if you turn the modem to 9600 baud and XON/XOFF Flow Control |
| 03:25.13 | Az_au | what codec? |
| 03:25.20 | asterboy | ulaw |
| 03:25.33 | Az_au | i went for the iaxmodem/hylafax solution myself |
| 03:25.49 | brodiem | * -> (SIP/g711u) -> SPA-1001 -> fax machine |
| 03:25.53 | asterboy | I really like the power of hylafax...good compatibility |
| 03:25.56 | brodiem | It fails about 90% of the time |
| 03:26.04 | Az_au | ya.. and the ability to fax from your workstation |
| 03:26.13 | brodiem | it always makes a handshake and then dies |
| 03:26.16 | asterboy | ya mine fails about 90% too |
| 03:26.26 | brodiem | asterboy, lol so it doesn't really work then |
| 03:26.33 | *** join/#asterisk Red_Dragon_X (n=Red_Drag@24.137.153.163) |
| 03:26.36 | Az_au | i have had no failures with hylafax |
| 03:26.38 | asterboy | it works good enough. |
| 03:26.48 | Red_Dragon_X | Whats up everyone ! |
| 03:26.56 | asterboy | It will retry and usually push on through. |
| 03:27.03 | brodiem | Az_au, do you have a physical fax machine in the mix though? |
| 03:27.13 | Az_au | not at this stage |
| 03:27.25 | Az_au | if i did i'd prolly plug it into a tdm400p |
| 03:27.27 | asterboy | physical fax works always |
| 03:27.32 | Red_Dragon_X | Anyone here working with 2 digium cards in one server ?? |
| 03:27.36 | asterboy | my failures are from Sportster |
| 03:27.58 | asterboy | yep, well...3 digium cards in 1 server |
| 03:28.03 | Red_Dragon_X | hmmm |
| 03:28.06 | brodiem | asterboy, your physical fax is also pluged into an ATA with SIP/g711? |
| 03:28.09 | asterboy | 3 X100Ps |
| 03:28.16 | asterboy | yes |
| 03:28.19 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-187-145.dsl.chcgil.sbcglobal.net) |
| 03:28.22 | Red_Dragon_X | hmm it might not be the same thing then |
| 03:28.25 | asterboy | I need a Courier modem. |
| 03:28.31 | *** join/#asterisk hansin321 (n=chatzill@c-67-174-182-21.hsd1.co.comcast.net) |
| 03:28.33 | brodiem | what ATA |
| 03:28.36 | asterboy | those don't have the hangups Sportster has |
| 03:28.45 | asterboy | Dlink |
| 03:28.47 | Red_Dragon_X | i have a TE210 and a TDM40 and they just wont work together ... |
| 03:28.58 | asterboy | can't remember the model, I'm upstairs. |
| 03:29.22 | asterboy | Red, had you moved them around the pci slots. |
| 03:29.23 | asterboy | ? |
| 03:29.36 | Red_Dragon_X | the TE card is my main phone line so that is priority, but i have the TDM so we can start faxing on the PRI lines for a WAY cheaper rate |
| 03:29.41 | Red_Dragon_X | hmmm |
| 03:29.44 | Red_Dragon_X | no i havent |
| 03:29.57 | Red_Dragon_X | there is only one other slot to try |
| 03:30.00 | asterboy | use the info in /proc |
| 03:30.07 | asterboy | those files tell you everything. |
| 03:30.35 | asterboy | look at interrupts and make sure there is NO sharing going on with the Digium cards |
| 03:30.36 | Red_Dragon_X | .. /proc ?? |
| 03:30.42 | asterboy | cd /proc |
| 03:30.43 | Az_au | proc filesystem |
| 03:30.45 | asterboy | cat interrupts |
| 03:30.57 | asterboy | it's not really a filesystem |
| 03:31.01 | Qwell | two processes? What are you, nuts? |
| 03:31.03 | Qwell | and yes, it is |
| 03:31.13 | Az_au | :D |
| 03:31.25 | Qwell | none /proc proc defaults 0 0 |
| 03:31.33 | Qwell | proc is the filesystem |
| 03:31.41 | asterboy | not in the sense that you will want to save data to it |
| 03:31.51 | brodiem | its procfs |
| 03:31.51 | Az_au | well actually |
| 03:31.52 | Az_au | you can |
| 03:31.53 | Qwell | So, dev isn't a filesystem either? :p |
| 03:32.13 | Qwell | and tmpfs isn't a filesystem? |
| 03:32.16 | asterboy | not in the sense that you will want to PERMANENTLY save something to it |
| 03:32.21 | Qwell | like tmpfs |
| 03:32.37 | Qwell | or fat32 |
| 03:33.11 | Qwell | asterboy: quit while you're behind :p |
| 03:33.59 | Az_au | haha |
| 03:34.05 | asterboy | it's not a filesystem |
| 03:34.18 | *** part/#asterisk freat (n=ron@h-72-244-84-43.chcgilgm.covad.net) |
| 03:34.23 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:34.33 | Az_au | http://www.google.com.au/search?q=proc%20filesystem would disagree |
| 03:34.38 | brodiem | asterboy, proc is a filesystem |
| 03:34.52 | brodiem | asterboy, everything in it is still populated by inodes like a normal filesystem |
| 03:35.03 | Qwell | CONFIG_PROC_FS=y |
| 03:35.03 | Qwell | CONFIG_SYSFS=y |
| 03:35.03 | Qwell | CONFIG_TMPFS=y |
| 03:35.08 | Qwell | They're all filesystems :P |
| 03:35.37 | asterboy | nah |
| 03:35.40 | brodiem | lol |
| 03:35.49 | asterboy | that was fun though |
| 03:35.52 | *** join/#asterisk sevard (i=sev@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 03:36.00 | asterboy | just playin with ya |
| 03:36.42 | *** join/#asterisk kokos1978 (n=kokos@HSE-London-ppp3510549.sympatico.ca) |
| 03:36.44 | brodiem | ..out |
| 03:38.51 | Red_Dragon_X | can some one please take a look at my zaptel and zapata to make sure i have this set up right ? i cant get this TDM to work! http://generalhan.pastebin.ca/49120 |
| 03:41.16 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
| 03:45.04 | *** join/#asterisk Foxtro (i=foxtro@123-78-246-201.adsl.terra.cl) |
| 03:45.09 | Foxtro | hola |
| 03:45.13 | Foxtro | alguien que pueda ayudarme? |
| 03:46.39 | Gamercjm | i dont think people in here speak spanish |
| 03:46.58 | Foxtro | :( |
| 03:47.04 | Gamercjm | speak english? |
| 03:47.16 | Foxtro | so so |
| 03:47.27 | Foxtro | :P |
| 03:47.32 | Foxtro | im try.. |
| 03:47.34 | Gamercjm | well just say it in spanish, i understand, just cant really type |
| 03:47.44 | Foxtro | i have a problem |
| 03:47.49 | Foxtro | with mysql bacjkend |
| 03:47.53 | Foxtro | for load data from mysql |
| 03:48.00 | Foxtro | for sip users |
| 03:48.17 | Foxtro | some dont work.. |
| 03:48.41 | Foxtro | my client (x-lite) say login failed |
| 03:49.40 | Gamercjm | oh, i dont use like mysql backend so i dont know |
| 03:49.59 | Foxtro | :( |
| 03:51.28 | *** join/#asterisk bmg505 (n=leon@dsl-146-2-203.telkomadsl.co.za) |
| 03:52.27 | Foxtro | Gamercjm |
| 03:52.41 | Foxtro | how configure ip addr where listen asterisk ? |
| 03:54.44 | *** join/#asterisk tessier_ (n=treed@adsl-70-137-65-15.dsl.sndg02.sbcglobal.net) |
| 03:56.43 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
| 04:00.08 | *** join/#asterisk b00mer_ (n=b00mer@204.9.61.37) |
| 04:02.58 | Red_Dragon_X | i need some advice from some one that is using a TDM card ... what all do you modprobe to get that card to work ? just wctdm ? or wcfxo ?? i just cant get this thing configured properly |
| 04:10.09 | Az_au | for me: |
| 04:10.11 | Az_au | modprobe zaptel |
| 04:10.11 | Az_au | modprobe wcfxs |
| 04:10.11 | Az_au | modprobe wcfxo |
| 04:10.34 | Az_au | although wcfxs is an alias to wctdm |
| 04:11.23 | Red_Dragon_X | i cant do that |
| 04:11.28 | Red_Dragon_X | i get an error |
| 04:11.30 | Red_Dragon_X | every time |
| 04:11.31 | Red_Dragon_X | modprobe wcfxs |
| 04:11.31 | Red_Dragon_X | ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 04:11.31 | Red_Dragon_X | FATAL: Error running install command for wctdm |
| 04:11.43 | Az_au | ok modprobe.conf should have some entries in it like this: |
| 04:11.45 | Red_Dragon_X | shit ... sorry i didnt mean to do that ... i was just pulling up the pastebin |
| 04:11.49 | Az_au | install wcfxo /sbin/modprobe --ignore-install wcfxo && /sbin/ztcfg |
| 04:11.49 | Az_au | install wctdm /sbin/modprobe --ignore-install wctdm opermode=AUSTRALIA fxshonormode=1 bootstringer=1 && /sbin/ztcfg |
| 04:12.05 | Az_au | alias wcfxs wctdm |
| 04:12.21 | Red_Dragon_X | Az_au: send them to me in a PM ... pasting to the channel is frowned upon here |
| 04:12.21 | Az_au | obviously yours will be a bit different as i am in australia |
| 04:14.18 | *** join/#asterisk Psykick (n=anon@203.167.226.250) |
| 04:14.20 | Psykick | hi guys |
| 04:14.44 | shido6 | excellent. |
| 04:14.50 | Psykick | my asterisk server appears to be leaving off zero on numbers when dialing ... either that or it's the client |
| 04:15.02 | Psykick | eg ... 0064 appears as 64 |
| 04:16.26 | Red_Dragon_X | shido |
| 04:21.55 | asterboy | Psykick, check your pattern matching before the Dial( |
| 04:22.37 | asterboy | I have just completed an essay on why /proc is not a filesystem |
| 04:22.50 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-187-145.dsl.chcgil.sbcglobal.net) |
| 04:22.52 | Red_Dragon_X | lol |
| 04:22.56 | Az_au | hehe where is it? |
| 04:23.03 | asterboy | lol just joking |
| 04:23.07 | Az_au | :P |
| 04:23.21 | asterboy | REd did you get your modprobe going? |
| 04:23.24 | Red_Dragon_X | n |
| 04:23.25 | Red_Dragon_X | o |
| 04:23.32 | Red_Dragon_X | everything is all messed up |
| 04:23.33 | asterboy | why does it say "span" for a TDM? |
| 04:23.38 | Red_Dragon_X | EXACTLY |
| 04:23.40 | asterboy | isn't that for T1 |
| 04:23.45 | Red_Dragon_X | yes |
| 04:23.52 | Red_Dragon_X | i have a TE210 in this server too |
| 04:23.55 | asterboy | something is missconfigured |
| 04:23.58 | asterboy | ah |
| 04:24.36 | Red_Dragon_X | when i have the fxo_ks stuff in the zapata and zaptel and i modprobe wct4xxp i get an error about the channels that i have setup on the TDM |
| 04:24.41 | Red_Dragon_X | its SOOO retarded |
| 04:24.56 | Red_Dragon_X | i checked IRQs and nothing is conflicting i dont know what the deal is |
| 04:25.18 | Red_Dragon_X | the only thing i havent done is move it to a new pci slot ... but i cant do that until the weekend cause its a live server and i cant bring the phones down |
| 04:25.52 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-187-145.dsl.chcgil.sbcglobal.net) |
| 04:26.33 | asterboy | do a paste bin of interrupts, modules and pci. |
| 04:26.42 | Flauto | hi people |
| 04:27.06 | Flauto | insecure=very has been changed in the new version? |
| 04:27.14 | *** join/#asterisk fparent (n=sujihz@modemcable252.177-131-66.mc.videotron.ca) |
| 04:27.42 | *** join/#asterisk tessier_ (n=treed@adsl-70-137-65-15.dsl.sndg02.sbcglobal.net) |
| 04:28.37 | asterboy | also, include modprobe stuff. |
| 04:29.38 | asterboy | /etc/modprobe.conf |
| 04:30.05 | asterboy | and the error message of course |
| 04:30.26 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
| 04:30.57 | *** join/#asterisk vopi (n=kkk@202.139.198.122) |
| 04:31.37 | fparent | Hello vopi, welcome to #asterisk |
| 04:32.45 | vopi | hi |
| 04:32.54 | vopi | thx :) |
| 04:33.17 | fparent | :) |
| 04:34.19 | Az_au | anyone know in the asterisk manager api can you check to see if a channel is being monitored via action: status or similar? |
| 04:35.07 | *** part/#asterisk fparent (n=sujihz@modemcable252.177-131-66.mc.videotron.ca) |
| 04:35.36 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 04:35.38 | *** join/#asterisk Derkommissar (n=Alberto@adsl-153-183-120.mia.bellsouth.net) |
| 04:37.26 | *** join/#asterisk Zipper_32 (n=test@s207-6-25-182.bc.hsia.telus.net) |
| 04:38.11 | asterboy | Anyone know what those 2 rows of 3 Chinese logograms are on the right hand side of the GXP-20000 Crystal Display? |
| 04:38.59 | Zipper_32 | After installing mpg123, putting mp3's in the appropriate directory, and placing users on hold, what is a common solution to hearing no sound (using the default musiconhold.conf) by the person being held? |
| 04:39.28 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 04:39.32 | asterboy | <PROTECTED> |
| 04:39.45 | asterboy | that was a funny response to that question. |
| 04:40.03 | asterboy | They are so owned. |
| 04:40.51 | dlynes | Zipper_32: not having a proper timing source, not having proper permissions on the mpg123 executable/mp3 directory/mp3 files/... |
| 04:42.16 | asterboy | what package has the mpg123? |
| 04:42.50 | Zipper_32 | 59r... |
| 04:43.20 | Zipper_32 | mpg123-0.59r |
| 04:45.40 | *** part/#asterisk Utah_Dave (n=boucha@12.118.109.86) |
| 04:49.40 | asterboy | ~mpg123 |
| 04:49.41 | jbot | i heard mpg123 is Real time MPEG Audio Player for Layer 1,2 and Layer3. URL: http://www.mpg123.de/. ONLY MPG123-R will work with asterisk. PERIOD. use 'make mpg123' in the asterisk source dir |
| 04:50.48 | Zipper_32 | Ahh... 'make mpg123' in the asterisk source dir!... *sigh* |
| 04:50.54 | Zipper_32 | Thanks. =) |
| 04:50.59 | asterboy | no prob |
| 04:52.14 | OliverX | is their a documentation for sip clients(more then one telephone) behind a nat? |
| 04:53.02 | Zipper_32 | Sweet, it's all working asterboy |
| 04:53.05 | Zipper_32 | Thank you kindly. |
| 04:53.23 | asterboy | glad to help |
| 04:54.25 | Zipper_32 | OliverX: All I know of is: http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:54.43 | OliverX | Zipper_32: Thank you very much! |
| 04:55.02 | Zipper_32 | Payin it forward, =) |
| 04:56.04 | OliverX | Hm yes i need a gatekeepter! |
| 04:56.09 | OliverX | like a proxy |
| 04:57.18 | OliverX | its a great pity that asterisk have no gatekeepter functions :( |
| 04:57.48 | OliverX | Protest! ;) |
| 04:59.09 | Derkommissar | gatekeeper? |
| 04:59.16 | Derkommissar | as an h323 gk? |
| 04:59.50 | Derkommissar | i think oh313 holds some of the features of an h323 gatekeeper, like enpoint registration and sucj |
| 04:59.59 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 05:01.12 | OliverX | i talk about sip (; |
| 05:01.24 | OliverX | or you give me a layer7 router :P |
| 05:02.19 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 05:04.21 | *** join/#asterisk Cglob (n=Cglob@202.8.86.162) |
| 05:04.54 | Cglob | what's the best FAX software that works with *? |
| 05:05.07 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 05:05.16 | Az_au | as in a pure software solution? |
| 05:05.55 | Cglob | Az_au: hmm, not really |
| 05:06.08 | Az_au | so you want to use a fax machine still? |
| 05:06.09 | asterboy | Cglob, Hylafax.org imho |
| 05:06.18 | Cglob | Az_au: with a fax machine via ATA |
| 05:06.42 | Cglob | asterboy: okie, will give it a try |
| 05:06.51 | florz | Cglob: Then you should best use the firmware that is already in your fax machine ... |
| 05:07.27 | Cglob | florz: with no additional software on my * box? |
| 05:08.15 | florz | Cglob: Well, what do you think should that additional software do? |
| 05:08.17 | Cglob | Az_au>: what would you recommend for the pure software solution? |
| 05:09.35 | Gamercjm | Im trying to use AGI, do any variables before the AGI(); get sent to the file? |
| 05:09.42 | OliverX | have anyone here script examples to connect with a mysql db? |
| 05:09.43 | Cglob | florz: to detect the fax signal probably, then redirect to a different extension otherwise |
| 05:10.48 | florz | Cglob: Well, but that's quite something different that what you'd usually call "fax software", isn't it? =:-) |
| 05:10.51 | VeNoMouS_ | <Cglob> what's the best FAX software that works with *? |
| 05:10.52 | VeNoMouS_ | lol |
| 05:10.59 | Az_au | Cglob: asterisk/iaxmodem/hylafax |
| 05:11.01 | VeNoMouS_ | Cglob spandsp is teh weakest link |
| 05:11.21 | VeNoMouS_ | i setup t.37 on all our ciscos |
| 05:11.27 | VeNoMouS_ | and jsut did it tthat way |
| 05:12.03 | Zipper_32 | Does anybody have any advice/recommendations on porting all old voicemails / configurations from one asterisk box (v1.0.4) to a new asterisk box (1.2.4)? |
| 05:12.29 | VeNoMouS_ | Zipper_32 : cp |
| 05:12.37 | Zipper_32 | =) |
| 05:12.48 | Zipper_32 | From where to where?... If you don't mind me asking. |
| 05:13.15 | Zipper_32 | And will custom recorded prompts stay if the extensions are all the same? |
| 05:14.19 | OliverX | Hmmmm |
| 05:15.10 | Zipper_32 | Hmm indeed. I'll have to come back to that voicemail question tomorrow... because I have to get out of work and go home... |
| 05:15.21 | Zipper_32 | Thanks for your help asterboy, |
| 05:16.42 | asterboy | dam, I just found another 6 rows of 2 Chineses logograhms on the left side of the GXP-2000! |
| 05:16.56 | asterboy | What the heck do they stand for? |
| 05:17.05 | Az_au | haha dunno |
| 05:17.28 | asterboy | sometimes there is someone on who is Chinese |
| 05:17.35 | asterboy | gotta ask one of those dudes |
| 05:17.48 | asterboy | Hard one to use with BableFish |
| 05:18.08 | Az_au | hehe gotta whip out the paint skills to submit it :D |
| 05:18.22 | asterboy | wonder if there is such a site...gotta be. |
| 05:19.22 | Az_au | some kind of ocr translator |
| 05:20.15 | asterboy | ya, that has to be one difficult task |
| 05:24.06 | *** join/#asterisk cybergypsy (n=cybergyp@APoitiers-156-1-4-59.w86-207.abo.wanadoo.fr) |
| 05:24.21 | Cglob | I try to send FAX through a Sipura Grand Stream to ordinary FAX machine and had this error => Unknown RTP codec 100 |
| 05:24.36 | Cglob | with g711u codec |
| 05:29.04 | *** join/#asterisk appdevx (n=appdevx3@203.172.17.212) |
| 05:29.22 | dlynes | asterboy: google does simplified chinese...don't know if it does traditional chinese |
| 05:30.06 | appdevx | guyz where can i buy asterisk card to connect to my PSTN.. |
| 05:30.11 | appdevx | im from Philippines |
| 05:30.12 | appdevx | :D |
| 05:32.46 | dlynes | appdevx: try ozvoip.com |
| 05:32.56 | dlynes | appdevx: it's a voip hardware reseller in australia |
| 05:36.20 | asterboy | thanks dlynes. |
| 05:37.35 | dlynes | asterboy: didn't realize i even helped :) |
| 05:37.54 | dlynes | asterboy: btw...you can try www.tigernt.com |
| 05:38.04 | dlynes | asterboy: it's a Chinese -> English dictionary |
| 05:38.23 | dlynes | asterboy: it does traditional and simplified chinese as well as hanyu pinyin |
| 05:39.30 | asterboy | what a fascinating language |
| 05:39.31 | dlynes | asterboy: and if you're using linux, you can download 'Hanzim' and install it |
| 05:39.40 | dlynes | asterboy: Hanzi means dictionary |
| 05:40.01 | asterboy | I need something that will go from logograhms to English |
| 05:40.15 | asterboy | How do you communicate that without a scratch pad? |
| 05:40.21 | dlynes | asterboy: try hanzim |
| 05:40.35 | dlynes | asterboy: you can look chinese up with it, by the number of strokes in the Chinese character |
| 05:41.18 | dlynes | asterboy: it'll only do one character at a time, but at least you might be able to get a general idea what the text says |
| 05:41.38 | dlynes | asterboy: some chinese characters have an entirely different meaning when combined with other chinese characters |
| 05:42.22 | dlynes | asterboy: or, if you want, email a photo of it, and i might be able to tell you waht it says |
| 05:42.41 | dlynes | I do know some Chinese, but my Chinese is far from being strong |
| 05:45.44 | asterboy | ya it seems they call them morphemes, meaning a dirivative of a meaning...so if you combine different characters, you get entirely different meanings. |
| 05:45.59 | dlynes | cool...there's a new dictionary for gtk2 on sourceforge |
| 05:46.02 | dlynes | and it does Chinese |
| 05:46.25 | asterboy | but again how do you convert Chinese to Engrish |
| 05:46.37 | asterboy | number of strokes and identification may help |
| 05:46.42 | dlynes | Exactly |
| 05:46.51 | dlynes | if you know how many strokes the character is |
| 05:46.58 | asterboy | don't have x11 |
| 05:46.59 | dlynes | you can look it up by the number of strokes, and go from there |
| 05:47.08 | asterboy | is there anything online? |
| 05:47.23 | dlynes | They're usually indexed by number of strokes, then phoneme, and then by tone |
| 05:47.41 | dlynes | www.tigernt.com |
| 05:48.24 | dlynes | Here's the dictionary that does it by stroke count for Linux: |
| 05:48.29 | asterboy | a logograhm is a graphame representing a word |
| 05:48.30 | dlynes | http://freshmeat.net/projects/hanzim |
| 05:48.41 | dlynes | It's Hanzi Master |
| 05:48.50 | asterboy | or a morpheme |
| 05:48.57 | dlynes | I think you mean morphine |
| 05:49.05 | asterboy | a meaningful unit of language |
| 05:49.14 | asterboy | morphine will make this easier to learn |
| 05:49.21 | Az_au | haha |
| 05:49.25 | Az_au | check the link |
| 05:49.27 | dlynes | If you want a for instance |
| 05:49.27 | Az_au | has the translations |
| 05:49.48 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 05:49.50 | dlynes | Try checking looking up 'zhongguo', using the pinyin translator on www.tigernt.com |
| 05:51.07 | asterboy | Az_au has a great link for the translation of the GXP-2000 |
| 05:51.54 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
| 05:52.12 | asterboy | left hand side: *English, *CAPS, *No CAPS, *Numeric/Digits/Numbers, *Symbol |
| 05:52.33 | asterboy | right hand side: *Receive new call, *New message/information. |
| 05:52.34 | dlynes | English would be 'Yingwen' |
| 05:52.47 | asterboy | the word Enlish? |
| 05:52.50 | dlynes | No idea on the rest of it |
| 05:52.55 | dlynes | correct |
| 05:53.01 | dlynes | Yingwen is mandarin for English |
| 05:53.38 | *** join/#asterisk mfedyk (n=mfedyk@adsl-63-194-240-129.dsl.lsan03.pacbell.net) |
| 05:54.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:55.09 | *** join/#asterisk somegeek (i=levin@unaffiliated/somegeek) |
| 05:56.00 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 05:56.16 | *** join/#asterisk buzzdee (n=buzzdee@host01.rapideye.medienstadt.net) |
| 06:00.16 | lokkju | is there a reason hangup would not be getting called after exiting a conference by just hanging up from the softphone? |
| 06:00.32 | asterboy | and the written and spoken chinese relationship is even more complex. |
| 06:00.35 | tainted- | hi i was wondering how to start a 'vonage'-like service with asterisk. i read on a techblog that its easy |
| 06:00.43 | asterboy | so learning one does not mean you can easily learn the other. |
| 06:01.00 | tainted- | it would be better than vonage though |
| 06:01.09 | tainted- | like a LOT better |
| 06:01.57 | asterboy | tainted-, wouldn't that be just setting up an * server and offering ata devices to connect to it. |
| 06:04.22 | *** join/#asterisk Creperum (n=ilya@mail.tex.kiev.ua) |
| 06:04.38 | tainted- | oh yea huh |
| 06:04.56 | tainted- | can u do it for me? i'll pay you at least $45 |
| 06:05.02 | Az_au | lol |
| 06:05.03 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
| 06:05.23 | Qwell | tainted-: western union? cashiers check? |
| 06:05.51 | mogorman | lol |
| 06:05.55 | tainted- | better than that |
| 06:05.57 | tainted- | i give u my word |
| 06:06.04 | vonage | mwaha |
| 06:06.07 | tainted- | that's like $12 right there |
| 06:08.17 | tainted- | i also like the, 'can someone make a billing system for me? just a simple one that does calling cards, LCR, and accounting - and user mgmt - and db-driven, but none of the fancy stuff' |
| 06:08.34 | vonage | rightttt |
| 06:09.04 | wasim | interconnect, with reconciliation and crm |
| 06:09.11 | wasim | franchise too |
| 06:09.12 | tainted- | something like vonage - but BETTER |
| 06:09.17 | tainted- | that's a golden statement |
| 06:09.29 | Qwell | tainted-: with stupider commercials? |
| 06:09.30 | vonage | rightttttttttttttttttt |
| 06:09.57 | tainted- | with vonage my whites are brighter than EVA!!!!!!111! |
| 06:10.09 | Qwell | Who's Eva? |
| 06:10.14 | asterboy | lol |
| 06:10.24 | tainted- | EVAR!!!!111!~1 |
| 06:11.08 | tainted- | wasim what's interconnect's URL |
| 06:11.25 | tainted- | i found interconnect - innovators in probe technology |
| 06:11.28 | wasim | tainted-: i mean, we need interconnect billing as well free |
| 06:11.49 | tainted- | ohh |
| 06:12.37 | tainted- | maybe i should franchise my stuff out |
| 06:13.03 | tainted- | http://www.voipwanker.com |
| 06:14.41 | tainted- | 'hi i had a question - i just signed up for voip-super-sonic.com service and need to get it to work with my asterisk box - the website says WE DO NOT ALLOW ASTERISK - but can someone help me?' |
| 06:15.38 | asterboy | lol |
| 06:16.08 | Gamercjm | I need help with AGI, I need a variable that was recieved with Read() to be used in the php |
| 06:16.11 | Qwell | I'm gonna start a provider... |
| 06:16.16 | Qwell | we're only gonna allow skinny |
| 06:16.26 | Gamercjm | i tried doing GET VARIABLE varname, but that doesnt seem to be working |
| 06:16.37 | tainted- | Qwell lol |
| 06:16.48 | *** join/#asterisk tessier_ (n=treed@ppp-71-134-211-86.dsl.sndg02.pacbell.net) |
| 06:16.50 | tainted- | chan_skinny ain't that great |
| 06:16.54 | asterboy | dlynes, when learning chinese I'm trying to pictograph the meanings. Too bad that is not how the script is setup. |
| 06:16.55 | Qwell | YET! |
| 06:16.56 | tainted- | j/k |
| 06:17.25 | tainted- | Gamercjm it's just STDOUT |
| 06:17.29 | wasim | damn ... qwell beat me to it, i was thinking of starting something for mgcp only |
| 06:17.38 | Qwell | wasim: It's been done |
| 06:18.43 | asterboy | Do all the GXP-2000 phones have logograhms on them? |
| 06:19.11 | asterboy | or did they make some for English...I don't want them lighting up. |
| 06:21.13 | Az_au | all the ones i've seen have them |
| 06:21.56 | lokkju | whew... now that I finished helping that guy - any of you have any ideas yet on what could be causing Festival to work fine, but standard Playback to never work? |
| 06:23.13 | Qwell | bed time |
| 06:26.26 | asterboy | Az_au, ok thanks, good to know. |
| 06:26.39 | asterboy | Do they every light up? |
| 06:27.41 | asterboy | I bet it depends on the firmware loaded. |
| 06:31.59 | Az_au | yea i've only seen it on boot |
| 06:32.06 | Az_au | maybe a language selection thing? |
| 06:37.37 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 06:39.17 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
| 06:39.48 | asterboy | don't see a language setting anywhere, no biggy |
| 06:40.36 | Az_au | could be a future thing... the gpx2k's have a lot of things that don't seem to be used yet |
| 06:40.48 | asterboy | well I'm impressed with the phone so far. |
| 06:40.55 | asterboy | The sound is better than my Polycom |
| 06:40.58 | asterboy | no hissing |
| 06:41.13 | Az_au | i've noticed the occasional electical interference on them but nothing too bad |
| 06:41.16 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 06:41.22 | Gamercjm | im getting 510 Invalid or unknown command from my AGI thing, But isnt 'GET VARIABLE varname' valid? |
| 06:41.25 | asterboy | Speaker phone at loudest setting, (and it's loud), feeds back into mic |
| 06:41.49 | Az_au | aparently they addressed some echo cancellation in the beta firmware but i haven't really checked it out |
| 06:42.00 | stoffell | asterboy: what phone is it you're talking about? |
| 06:42.05 | Az_au | gpx2000 |
| 06:42.16 | Az_au | grandstream |
| 06:42.17 | asterboy | yep |
| 06:42.44 | asterboy | The polycom has the hiss |
| 06:42.58 | *** join/#asterisk Falle (n=falle@falle.se) |
| 06:43.00 | stoffell | asterboy: then you have a bad polycom. you should also try the thomson ST2030, same price as gxp but better quality :) |
| 06:43.23 | asterboy | err...I just tried the polycom and it did *not* have the hiss. |
| 06:43.25 | asterboy | hmmm. |
| 06:43.39 | Az_au | be interesting to see what the gxv3000 quality is like |
| 06:43.41 | asterboy | could be something else, like my cheapo FXO clone cards |
| 06:43.42 | Az_au | http://blog.tmcnet.com/blog/tom-keating/voip/grandstream-gxv3000-video-phone.asp |
| 06:43.56 | stoffell | hehe |
| 06:44.01 | asterboy | ya I've seeen them and do like the style |
| 06:44.22 | asterboy | still, for the price the grandstream is impressive. |
| 06:44.33 | Az_au | we're just using eyebeam currently.. be nice to have something to sit on the desk |
| 06:44.39 | asterboy | but I'll get the real skinny when it goes into a production environment. |
| 06:44.41 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 06:44.58 | stoffell | asterboy: not after you've tried a ST2030, big quality difference imho |
| 06:45.00 | asterboy | this week, I won a small sale. |
| 06:45.00 | *** join/#asterisk NirS (n=NirS@84.95.90.229.cable.012.net.il) |
| 06:45.11 | asterboy | 5 phones, 4 lines setup |
| 06:45.22 | *** join/#asterisk ChulJin (n=kvirc@2001:5c0:8fff:fffe:0:0:0:4b05) |
| 06:46.07 | asterboy | fuck is that a nice looking phone., |
| 06:46.52 | Az_au | indeed.. just checking it out myself |
| 06:47.01 | asterboy | does it have a backlight? |
| 06:47.17 | Az_au | i'd have to import... |
| 06:47.21 | *** part/#asterisk ChulJin (n=kvirc@2001:5c0:8fff:fffe:0:0:0:4b05) |
| 06:47.22 | stoffell | asterboy: that's about the only drawback, no backlight.. |
| 06:47.43 | stoffell | i advise people to use USB light (pick your color) attached to the pc :) |
| 06:48.05 | *** join/#asterisk L0g0ff (n=thomas@pix89.global-e.nl) |
| 06:48.40 | asterboy | are they selling in NorthAmerica? |
| 06:48.52 | asterboy | Distribution seems to be on the other side of the world |
| 06:49.02 | Az_au | yea... and the wrong hemisphere for me :P |
| 06:49.16 | stoffell | currently europe yes :) but I belive they will be shipping april/may in US |
| 06:49.21 | L0g0ff | Hi, is there a way to change the volume on al sip phone in asterisk? All my sip related phones have a very low sound |
| 06:49.38 | asterboy | do you know the price? |
| 06:49.45 | lokkju | http://rafb.net/paste/results/YCtn6M40.html - full log shows answer, then wait, then playing beep, then nothing untill I hangup - hangin on the Playback, obviously, but *why* |
| 06:50.02 | stoffell | asterboy: approx. 125 EUR excl. VAT |
| 06:50.23 | asterboy | we don't have a VAT |
| 06:50.33 | asterboy | what is it 2 to 1 |
| 06:50.37 | asterboy | for the conversion |
| 06:50.47 | asterboy | so about 250$ USD? |
| 06:50.50 | stoffell | in USD ? |
| 06:51.05 | NirS | what phone are you guys talking about ? |
| 06:51.12 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 06:51.12 | lokkju | asterboy, no, but we commonly have import taxes, beleive it or not |
| 06:51.32 | stoffell | asterboy: USD price is 152 USD |
| 06:51.33 | asterboy | wait thats way off. |
| 06:51.36 | asterboy | ya |
| 06:51.45 | asterboy | $175 CDN |
| 06:51.47 | stoffell | NirS: ST2030 |
| 06:51.55 | NirS | which company ? |
| 06:52.05 | asterboy | So double what I paid for the Grandstream. |
| 06:52.14 | stoffell | NirS: thomson, currently only europe |
| 06:52.21 | Az_au | asterboy: did you get yours second hand? |
| 06:52.23 | NirS | well, I'm in europe |
| 06:52.26 | stoffell | asterboy: yeah, so wait till they are distributed in us :) |
| 06:52.27 | asterboy | I think I just found my MID Line phone |
| 06:52.28 | NirS | what's the website ? |
| 06:52.35 | austinnichols101 | right around the price of the linksys |
| 06:52.37 | asterboy | no, that's new |
| 06:52.41 | Az_au | that's not bad |
| 06:52.45 | stoffell | NirS: search ST2030 on voip-info |
| 06:52.51 | asterboy | ya it's a good price |
| 06:53.46 | asterboy | Grandsream = Low end, Thompson = Mid Market, Polycom/Cisco = High End |
| 06:54.07 | stoffell | asterboy: yeah, i'm thinking the same :) |
| 06:54.11 | *** join/#asterisk rollot (n=rollotom@c-68-32-33-163.hsd1.pa.comcast.net) |
| 06:54.18 | austinnichols101 | aastra = mid market too |
| 06:54.38 | asterboy | but doesn't look near as nice as that ST2030 |
| 06:54.46 | Gamercjm | write("GET VARIABLE id"); is still an invalid or unknown command ;/ |
| 06:54.50 | Gamercjm | :/* |
| 06:54.50 | asterboy | Especially with the expansion module. |
| 06:55.02 | austinnichols101 | how's the weight on the st2030 - can you club someone with the handset? |
| 06:55.09 | asterboy | lol |
| 06:55.31 | austinnichols101 | that's one of the key indicators of a good phone |
| 06:55.37 | stoffell | austinnichols101: not as heavy as a cisco (your need to workoot for that) but pretty heavy though.. |
| 06:55.43 | stoffell | workoot=workout |
| 06:56.02 | austinnichols101 | yeah - better not even get close to my desk or I'll knock you out |
| 06:56.11 | rollot | Anyone know why I'd be getting iax2 'unable to support trunking on peer without zaptel timing?" lsmod shows zaptel loaded, new build on pulled working config? |
| 06:56.13 | stoffell | lol |
| 06:57.03 | Shaun2222 | rollot: not sure if this matters but if you dont have a digium card i hear you need ztdummy loaded |
| 06:57.17 | Shaun2222 | do you see ztdummy module loaded when you do lsmod? |
| 06:57.33 | rollot | thanks Shaun. old machine had x100p installed. tried using ztdummy, went back to x100p just to use it as timing source |
| 06:57.50 | lokkju | hmf |
| 06:57.50 | rollot | same error message w/ ztdummy loaded too |
| 06:57.57 | lokkju | wtf could be causing sounds to hang |
| 06:58.12 | Shaun2222 | rollot: you have to manually enable ztdummy or somthing, in the make file, i also had to touch ztdummy.c or osmthing like that |
| 06:58.24 | stoffell | anyone know what a Polycom 501 with error "config file error 0x10020" would mean? |
| 06:58.59 | lokkju | rollot, what distro? |
| 06:59.06 | rollot | Shaun: had to modify Makefile to enable ztdummy in build then load, but same issue.. anyway, kind of doesn't apply now with x100p installed? |
| 06:59.30 | rollot | lokkju, deb 2.6.8-2 |
| 06:59.39 | asterboy | wow, 10 multilines on that st2030 |
| 07:00.17 | asterboy | polymorphic rings and 100 address book contacts |
| 07:01.02 | asterboy | bicolor led indicators. |
| 07:01.09 | asterboy | I want one now. |
| 07:01.19 | L0g0ff | Iis there a way to change the volume on all sip phones in asterisk? All my phones have a very low sound |
| 07:01.57 | asterboy | L0g0ff, what type of phones? |
| 07:02.10 | L0g0ff | grandstream & alcatel |
| 07:02.22 | Shaun2222 | anybody know what the name of a normal dial tone is when setting tone on the dialplan.xml for a cisco 7960 phone? |
| 07:02.26 | asterboy | well grandstream gxp2000 is pretty dam loud |
| 07:02.33 | asterboy | dunno about alcatel. |
| 07:02.43 | asterboy | they should have buttons on the phones for adjustment |
| 07:03.00 | L0g0ff | but is there a way to change the volume on the server self ? |
| 07:03.18 | rollot | Shaun: like ringer1.pcm ? |
| 07:03.34 | stoffell | asterboy: gxp does have buttons, but it doesn't remember settings after reboot |
| 07:03.35 | Shaun2222 | ringer1.pcm is a dial tone sound? |
| 07:03.43 | rollot | woops |
| 07:03.47 | rollot | :) |
| 07:03.47 | *** part/#asterisk rollot (n=rollotom@c-68-32-33-163.hsd1.pa.comcast.net) |
| 07:03.52 | *** join/#asterisk rollot (n=rollotom@c-68-32-33-163.hsd1.pa.comcast.net) |
| 07:04.40 | asterboy | ah, that sucks |
| 07:05.15 | asterboy | polycom keeps forgetting also |
| 07:05.53 | asterboy | sip.cfg has a setting suppoisedly |
| 07:06.03 | *** join/#asterisk Darkhalf (n=darkhalf@cpe-72-130-156-112.san.res.rr.com) |
| 07:06.18 | stoffell | asterboy: hm, okay, thanks for tip (just configuring a bunch of polycom 501's) |
| 07:07.18 | lokkju | any of you played with 802.11b/g sip/iax phones? any links to suggestions, etc? |
| 07:07.46 | Shaun2222 | lokkju: i played with one... havnt got it working yet but it's quiet annoying to configure... |
| 07:07.53 | asterboy | <volume voice.volume.persist.handset="1" voice.volume.persist.headset="0" voice.volume.persist.handsfree="1"/> |
| 07:08.04 | Shaun2222 | zyxtel was the maker |
| 07:08.12 | lokkju | ah, k, heard of that one |
| 07:08.32 | asterboy | ya I want a wireless SIP phone too. |
| 07:08.37 | asterboy | not sure which is best. |
| 07:08.39 | Shaun2222 | i was just annoyed because teh web interface was lame and wouldnt let you change bearly anything and it didnt support nat... |
| 07:08.54 | lokkju | eh |
| 07:08.55 | lokkju | sucky |
| 07:09.05 | lokkju | I am sort of wanting on that is java on linux |
| 07:09.09 | L0g0ff | asterboy, i dont have the sip.cfg. I have a sip.conf in my /etc/asterisk/sip.cong |
| 07:09.11 | L0g0ff | asterboy, i dont have the sip.cfg. I have a sip.conf in my /etc/asterisk/sip.conf |
| 07:09.14 | Shaun2222 | lokkju: also it didnt really want to save profiles very well |
| 07:09.14 | asterboy | I just setup my Polycom connected via a crossover cable to my laptop and bridged the wireless. |
| 07:09.19 | asterboy | Works excellent. |
| 07:09.20 | lokkju | cause the nat passthrough is going to be the biggest thing for me |
| 07:09.20 | Shaun2222 | was kinda crapy feeling too |
| 07:09.48 | lokkju | I want a small voip phone that I can carry like a cellphone, essentially |
| 07:09.50 | Shaun2222 | lokkju: it supports nat but you have to use external soh... or extenral proxys |
| 07:10.09 | Shaun2222 | lokkju: which is annoying. |
| 07:10.11 | asterboy | L0g0ff, that was just my own musings because I have a Polycom...you won't have that file if you don't have a Polycom |
| 07:10.14 | lokkju | shaun222, which types of proxies does it support? and is it sip or iax? |
| 07:10.32 | lokkju | (I am assuming SIP) |
| 07:10.35 | *** join/#asterisk h3x0r4t0r (i=hex@ip68-96-175-172.lv.lv.cox.net) |
| 07:10.46 | Shaun2222 | lokkju: sip, not sure, go their site it was the P-2000W |
| 07:11.09 | L0g0ff | ok |
| 07:11.25 | asterboy | I don't think there is a setting for that in * |
| 07:11.37 | asterboy | for SIP anyway. |
| 07:11.38 | *** join/#asterisk CGlob (n=Cglob@202.8.86.162) |
| 07:11.55 | asterboy | Zaptel stuff gives you tx/rxgain |
| 07:12.34 | asterboy | i can't believe your grandstream is quite...mine is hella loud |
| 07:12.47 | L0g0ff | ok. than i think in know enough. Thank you anyway :) |
| 07:13.06 | dlynes | grandstream doesn't have a volume control? |
| 07:13.07 | asterboy | no prob...check back later...some gurus on here now more |
| 07:13.24 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 07:13.28 | asterboy | yes, but not via SIP configuration |
| 07:13.50 | dlynes | ah..but why would you want to control it through sip? |
| 07:14.33 | L0g0ff | yes. For all phones not only the grandstream but alcatel's too |
| 07:14.51 | asterboy | guess he has a volume problem with all phones being quite. |
| 07:14.57 | asterboy | not sure what would cause that. |
| 07:15.11 | dlynes | bad hearing? |
| 07:15.50 | L0g0ff | yes, the other side of the line is bad hearing. When I call internally every think works fine |
| 07:16.11 | dlynes | ah...yeah...i get that problem, too |
| 07:16.22 | dlynes | when someone is using a sipura unit for a door entry system |
| 07:16.25 | Gamercjm | Is there an alternate way in AGI to retrieve a variable other then GET VARIABLE varname? |
| 07:16.25 | L0g0ff | and i there a way to fix it ? |
| 07:16.49 | dlynes | L0g0ff: haven't figured out a way, but then again, haven't spent much time on it yet, either |
| 07:17.05 | dlynes | L0g0ff: I suspect it's probably a voltage issue on the sipura unit |
| 07:17.40 | L0g0ff | Ok, i think i replace the phones with better one's |
| 07:18.18 | *** join/#asterisk Ansonmus (n=ahaeser@a213-84-26-148.adsl.xs4all.nl) |
| 07:18.37 | lokkju | shaun222, not too bad a price for that phone, either... hmm (http://www.gmprice.com/index.php?qstring=P%2D2000W&cat=61838) |
| 07:18.38 | Ansonmus | Hello can anyone recommend a SIP softphone for Windows which can do conf. and transfer? |
| 07:19.10 | lokkju | Ansonmus, sipXezphone, sipXphone, x-lite all do, don't they? |
| 07:19.25 | dlynes | Ansonmus: as does snom360 softphone (www.snom.de |
| 07:19.32 | lokkju | I personally like x-lite |
| 07:19.39 | lokkju | never used snom though |
| 07:19.40 | Ansonmus | in x-lite it seems disabled |
| 07:19.44 | lokkju | hmm |
| 07:19.45 | lokkju | odd |
| 07:19.50 | lokkju | but understandable |
| 07:20.16 | dlynes | Ansonmus: snom360 features are all enabled, but no g729 or g723 on the softphone |
| 07:20.26 | dlynes | Ansonmus: it's exactly the same as the hardphone |
| 07:20.56 | Ansonmus | is it freeware? |
| 07:21.08 | dlynes | the softphone is freeware for non-commercial use |
| 07:21.45 | dlynes | it's a very nominal fee for commercial use |
| 07:22.00 | dlynes | and I think you can even get the g729 codec for the commercially licensed snom softphone |
| 07:22.23 | Ansonmus | ok, sounds good |
| 07:22.48 | dlynes | yeah...you can download the snom360 manual from there, too |
| 07:22.55 | dlynes | it tells you how to use the softphone and the hardphone |
| 07:24.21 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:29.28 | lokkju | damn it |
| 07:29.35 | lokkju | I still need a logo for gmprice |
| 07:29.40 | lokkju | looks shitty right now |
| 07:31.16 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
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| 07:39.01 | Ansonmus | dlynes are there also other skins available? |
| 07:40.24 | Ansonmus | dlynes: yes fir commercial customers i think :) |
| 07:42.46 | dlynes | Ansonmus: no idea...sorry |
| 07:45.31 | Shaun2222 | whats allowguest do in the sip.conf, from the sound of it it allows any sip connection to be able to connect into the asterisk server and i assume use it as if they logged in? |
| 07:47.41 | *** join/#asterisk Tili (i=Tili@61.144.21.108) |
| 07:59.49 | Shaun2222 | i just configured exten 1001 and 1002 on 1 of my 7960's then configured 1002 on another 7960, when i use 1002 on that second 7960 why doesnt the first 7960 show that extension in use? |
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| 08:14.11 | kmilitzer | /j# asterisk-dev |
| 08:14.27 | kmilitzer | Oops, sorry ;) |
| 08:14.30 | iDunno | heh |
| 08:17.28 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:21.30 | Shaun2222 | when asterisk calls out my callerid on my cell shows as unknown.. is this somthing i need to set in asterisk so it's sent? |
| 08:22.00 | Zhadnost | it depends on what's between asterisk and the cell |
| 08:22.19 | Shaun2222 | voicepulse connect right now hooked through iax |
| 08:22.44 | Zhadnost | though to set a callerid, in the peer decleration just put callerid = Textual ID <numeric ID> (or similar). |
| 08:23.35 | Zhadnost | usually the provider will force a callerid on the call (but not always). |
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| 08:49.37 | Shaun2222 | is their documentation some where that tells what each module does? |
| 08:53.18 | rkr245 | hi every body |
| 08:54.21 | rkr245 | i need to add a peer in sip.conf can any body give a clear idea how to add a peer and required codecs |
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| 09:05.16 | sleepy_one | hello everyone |
| 09:08.36 | UnderMine | lo |
| 09:10.11 | *** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il) |
| 09:10.54 | sleepy_one | hi Romik |
| 09:11.22 | sleepy_one | what's up? |
| 09:14.37 | Romik | sleep_one: good |
| 09:14.46 | Romik | sleepy_one: what with you? |
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| 09:16.12 | sleepy_one | Romik, fine thanks |
| 09:16.50 | sleepy_one | it's quiet tonight |
| 09:18.21 | *** join/#asterisk yuta-vcnet (i=yuta-vcn@212.118.246.50) |
| 09:20.32 | Romik | i have problem with channel when agent try to transfer call to other agent or phone # - chan_zap.c: We're Zap/50-1, not Zap/50-2<ZOMBIE>" or "chan_zap.c: We're Zap/50-1, not SIP/???" |
| 09:20.47 | Romik | anybody can advice? |
| 09:25.21 | shiznatix | i have a problem with sending and recieving a fax. when i try sending a fax it rings the fax machine but just does not send anything. also when i try to recieve a fax it says its recieving and doing the rxfax thing but it just hangs up and does not save the file |
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| 09:28.00 | sleepy_one | Romik, what kind of hardware do you have? |
| 09:28.23 | Romik | sleepy_one: quad port digium card + zhone c/b fxs only |
| 09:28.59 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 09:29.31 | sleepy_one | TDM400 analog ? TE4xxp Quad span T1/E1? |
| 09:29.33 | Romik | sleepy_one: no problem to make transfer from phone to phone, but when make transfer when you in queue- time to time...not always it's come into such loop of error message, and only way to stop it - softhangup the channel or restart |
| 09:29.51 | Romik | sleepy_one: TE4xxp Quad span T1/E1 |
| 09:31.21 | sleepy_one | I see |
| 09:33.03 | sleepy_one | Romik, how many T1s or E1s do you have connected to the card ? |
| 09:33.49 | sleepy_one | shiznatix, what kind of hardware are you using? |
| 09:34.21 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 09:35.45 | sleepy_one | shiznatix, have you looked at voip-info.org ? http://www.voip-info.org/wiki-Asterisk+fax |
| 09:37.18 | shiznatix | sleepy_one, yes i have checked the site many times |
| 09:37.47 | shiznatix | sleepy_one, I am using a zapata card, let me get the details |
| 09:37.54 | Romik | sleepy_one: port 1 is E1 as PRI, port 2 is T1 as CB, timer run port 3 |
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| 09:39.37 | shiznatix | sleepy_one, It is a Digium TIGER 320 card |
| 09:40.09 | sleepy_one | Romik could you please post your zaptel.conf and zapata.conf on pastebin.com ? |
| 09:41.08 | sleepy_one | shiznatix, TDM400p series? with up to 4 FXO or FXS modules? http://www.digium.com/en/products/hardware/tdm400p.php ? |
| 09:41.23 | *** join/#asterisk marcel1 (n=chatzill@195.94.71.181) |
| 09:41.26 | marcel1 | hello |
| 09:41.37 | sleepy_one | hi marcel1 |
| 09:41.57 | marcel1 | give a software to use skype and asterisk ? |
| 09:42.40 | shiznatix | sleepy_one, yes that one |
| 09:43.35 | sleepy_one | marcel1, http://www.voip-info.org/wiki-bounty+skype I do not think there is software to interface skype with asterisk |
| 09:43.45 | sleepy_one | marcel1, I may be wrong |
| 09:43.48 | Romik | sleepy_one: when timer was on port 2 that hear clicks it's disappears when i moved it to port 3 |
| 09:44.29 | Romik | sleepy_one: http://pastebin.ca/49139 |
| 09:44.37 | sleepy_one | Romik, thanks |
| 09:45.01 | marcel1 | ok thx |
| 09:45.39 | marcel1 | i have that found "http://www.rsdevs.com/psgw.shtml" |
| 09:46.02 | Romik | sleepy_one: span 1 connected to PRI, and span 2 to CB - no other connection from the card |
| 09:47.02 | sleepy_one | Romik, ok so channel 50 is one of the channels on the channel bank |
| 09:47.14 | Romik | yes |
| 09:47.44 | Romik | sleepy_one: i even open bug in tracker - nobody even look into... http://bugs.digium.com/view.php?id=6876 |
| 09:48.31 | Romik | sleepy_one: it's happend even when transfer to SIP or other ZAP |
| 09:48.56 | sleepy_one | Romik, and your channel bank has 23 FXS modules? |
| 09:49.11 | Romik | sleepy_one: 24 regular CB |
| 09:49.47 | Romik | sleepy_one: may be this can help...from debug output http://bugs.digium.com/file_download.php?file_id=9795&type=bug |
| 09:50.59 | sleepy_one | Romik, thanks I'm looking at it now |
| 09:52.22 | sleepy_one | shiznatix, have you tried to barge in on the fax channel ( s ) ? are you using FXO and FXS or just FXS to the fax ? How are you receiving the fax? Over analog PSTN? |
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| 09:55.39 | rkr245 | sleepy_one : does the asterisk can be used as a voip service provider or just for home or office usage only |
| 09:56.01 | shiznatix | sleepy_one, i dont know what you mean by barging in. i am just using fx to send and recieve the fax. a guy sends me a fax through a regular fax machine and i am trying to save it to the asterisk computer. im using a standard regular phone line to get the incoming fax and to send the faxes |
| 09:56.03 | sleepy_one | rkr245, I can be used for all of the above |
| 09:56.27 | sleepy_one | s/I/it/ |
| 09:56.51 | rkr245 | sleepy_one:o.k |
| 09:57.42 | rkr245 | sleepy_one: iam a new employee in one telecom company and they asked me to implement this voip services using asterisk |
| 09:57.59 | sleepy_one | rkr245, asterisk can support at least 250 simultaneous calls with the right hardware, probably more. It supports T1 and E1 cards with 1, 2 and 4 T1s or E1s |
| 09:58.50 | shiznatix | sleepy_one, http://pastebin.com/655301 that is all of my settings and configurations. Asterisk is running as root just FYI |
| 09:58.51 | rkr245 | sleepy_one : i installed fedora core 4 even thogh iam new to linux and i installed asterisk -1.2.6 and it show asterisk running successfully |
| 09:59.09 | sleepy_one | Romik, so you had a 3 way call on Zap50 and after the hangup it became a zombie ? |
| 09:59.26 | Romik | rkr245: we have 1 server with 3 quad span cards....1 PRI + 11 x T1 to CB's (264 lines) |
| 09:59.32 | sleepy_one | shiznatix, thanks, let me take a look |
| 10:00.09 | sleepy_one | rkr245, what kind of hardware are you using or planning to use? which country are you from? |
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| 10:00.20 | rkr245 | sleepy_one: but here my problem is i cant move further iam not getting any ideas ,here iam testing on an old pc with no zapatel cards or something |
| 10:00.35 | rkr245 | iam from india |
| 10:01.16 | sleepy_one | rkr245, did you install zaptel? and ztdummy? |
| 10:01.29 | rkr245 | sleepy_one: i install ztdummy |
| 10:01.54 | Romik | sleepy_one: what you mean 3 way ? 1) agent receive call 2) client tell to him that he want to speak with agent #4732, 3) agent press flash and dial the 4732 4) speaking about client with new agent 5) press flash again... and 3 people speaking together..... and after it - agent hangup...to leave them speaking togehther |
| 10:02.36 | rkr245 | sleepy_one: can you tell me how i can register some one on this asterisk server |
| 10:02.41 | sleepy_one | Romik, ok that's what I meant it was a 3way call you had 3 people talking and then one left |
| 10:03.01 | Romik | sleepy_one: so how i can fix that it's becoume this problem? |
| 10:03.07 | sleepy_one | rkr245, You can use SIP or IAX2 softphones |
| 10:03.19 | rkr245 | sleepy_one: o.k |
| 10:03.25 | Romik | sleepy: or 3way does not work in asterisk with agents channels? |
| 10:03.31 | rkr245 | sleepy_one : iam using sip phones |
| 10:03.56 | *** part/#asterisk shmooz (n=shmooz@H142.C72.B0.tor.eicat.ca) |
| 10:04.18 | sleepy_one | Romik, I had 3way calling working just fine on my old T1 PRI, I don't remember but I think it worked when I had a TA750 channel Bank |
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| 10:04.55 | rkr245 | i have now one grandstream adapter and on another friends pc who is staying out of my city he installed x-lite soft phone now please tell me how can i register this person using x-lite phone having public ip address |
| 10:05.06 | sleepy_one | rkr245, edit /etc/asterisk/sip.conf and add your SIP phones to it then configure the phones to register with asterisk |
| 10:05.31 | rkr245 | sleepy_one: with the ip address of my freind |
| 10:05.38 | sleepy_one | rkr245, http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 10:05.43 | Romik | sleepy_one: I have 3way on other server but not from the agent channel. same configuration same CB's... same motherboards |
| 10:06.58 | sleepy_one | Romik, what kind of channel banks? some of them work strangely. When I had an Adtran TA750 we had echo problems, disconnect problems, ghost calls etc |
| 10:07.34 | rkr245 | sleepy_one:yeah here is some conf how to add users friends etc.. i will go through it now thanks a lot |
| 10:07.42 | sleepy_one | Romik, people would hang up but the channel bank or asterisk wouldn't realize they had disconnected so the channels would stay open IIRC |
| 10:07.59 | sleepy_one | rkr245, you're welcome :-) |
| 10:08.13 | sleepy_one | rkr245, voip-info.org is a great source of information :-) |
| 10:08.27 | Romik | sleepy_one: i have zhones 24fxs, i have in one location like 11 of them...other location 13...... this location is only 1! but they work as agents. |
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| 10:15.04 | sleepy_one | shiznatix, please add a Monitor before rxfax and see if you can record the fax being received |
| 10:16.53 | sleepy_one | shiznatix, http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor |
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| 10:25.24 | backblue | does anyone know how to enable alarms in zap spans or zap channels? |
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| 10:29.13 | sleepy_one | backblue, zttool should show alarms and I believe the CLI shows different info based on the verbosity level |
| 10:32.48 | backblue | sleepy_one: i want reports in ALARMS, like email, or IM notification. |
| 10:34.01 | sleepy_one | backblue, I believe you could write a script to monitor the CLI or the asterisk log file and send email or an IM if it detects an alarm |
| 10:44.44 | backblue | sleepy_one: i'm asking if there is something out there already done! |
| 10:45.28 | sleepy_one | backblue, sorry I don |
| 10:45.34 | backblue | ok tks |
| 10:45.43 | sleepy_one | s/sorry I don/sorry I don't know/ |
| 10:47.07 | sleepy_one | backblue, may these will help http://www.voip-info.org/wiki/view/Asterisk+monitoring http://www.marko.net/asterisk/archives/0211/0044.html |
| 10:47.27 | sleepy_one | s/may/maybe/ |
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| 11:12.21 | OliverX | where can i find a complete list of all standard variable`s? dont write the README or so :D |
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| 11:22.10 | shiznatix | when trying to send a with spandsp it starts to work but it only rings their fax machine once then it hangs up the call |
| 11:22.18 | OliverX | is ${REMOTE} a standard variable and if yes for what is this variable? |
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| 11:34.08 | OliverX | is ${REMOTE} a standard variable and if yes for what is this variable? |
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| 11:41.10 | Ahrimanes | anyone using snom phones with asterisk? i would like a little help with hints and subscribe |
| 11:46.47 | cced | YES. i snom phone. |
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| 11:49.33 | cced | Ahrimanes : what problem? |
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| 11:50.11 | Ahrimanes | cced: well i just need to understand it right i guess, i set the button on the snom to destination and the sip extension i want to monitor.. |
| 11:50.20 | Ahrimanes | cced: but what should go in the dialplan and where? |
| 11:51.12 | cced | snom phone need special set? |
| 11:51.34 | Ahrimanes | well it has a button ta |
| 11:51.47 | Ahrimanes | that's subscribed to the extension i want to monitor |
| 11:52.25 | Ahrimanes | but does exten => 2200,hint,SIP/2200 mean that SIP/2200 will be notified when extension 2200 is busy? |
| 11:52.48 | cced | sorry . I do not use this. |
| 11:53.51 | Ahrimanes | ok |
| 11:57.14 | OliverX | is ${REMOTE} a standard variable and if yes for what is this variable? |
| 11:58.29 | shiznatix | when trying to send a with spandsp it starts to work but it only rings their fax machine once then it hangs up the call. how can i fix this? |
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| 12:04.11 | JamesDotCom | bkw! |
| 12:04.18 | JamesDotCom | svn server is broken ;( |
| 12:07.24 | shiznatix | can anyone help me with spandsp? |
| 12:08.15 | sleepy_one | OliverX, no, it is not in asterisk-1.2.x/doc/README.variables and it is not defined in * when I tried it |
| 12:12.34 | OliverX | ^thanks |
| 12:21.29 | grem_lin | Hi, I wonder if anybody could help me with a DISA query. As I understand it upon entering the correct code you are then given a dial tone and can enter an extension which has been setup in the specified context - however I am finding that when I dial a known extension I get a fast-busy tone, would anybody know why this might be? |
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| 12:27.11 | grem_lin | Or is it only possible when using DISA to specify a context which then dials an extension? |
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| 12:28.51 | ramtha | hey |
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| 12:30.25 | ramtha | can some telle me, where is the difference between src / clid in the cdr? it seems that src is only the clleridnumber and clid is the calleridname. ho can i figure out only the calleridnumber in my dialplan |
| 12:31.02 | ramtha | i am not sure, tha CALLERIDNUM is really the (src filed in CDR) nummer or is it the callerid 8nummber and name) |
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| 12:50.14 | mut | is there a sangoma a102 with echo cancelling? |
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| 12:50.29 | mut | all i see is the A104D which is quad pri with echo cancel |
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| 12:57.04 | caio1982 | isnt realtime supposed to show me my users with "sip show peers"? i just can check them using "realtime load family username foo" |
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| 12:57.38 | xbit` | re |
| 12:57.57 | docelm0 | re what? |
| 12:58.10 | xbit` | something like hi |
| 12:58.37 | docelm0 | so why not just say hi.. its kinda universal... |
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| 12:59.16 | xbit` | you are right. |
| 13:00.35 | [TK]D-Fender | mut : only the A104d at this point. They are working on lower density versions. |
| 13:01.01 | thieumS | Hello, i'm looking for a way to use field "to" instead of field "invite" from sip INVITE message, for routing purposes in extensions.conf |
| 13:01.14 | mut | man |
| 13:01.27 | mut | argh |
| 13:01.54 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-48.claranet.co.uk) |
| 13:02.09 | xbit` | i get the incoming calls from isdn to the s extension, but i have ddi-s. how can i figure out what number was dialed? XXs extensions not working. |
| 13:03.04 | [TK]D-Fender | mut : Believe me the EC on it is worth it even for the overkill of being 4 port.... |
| 13:05.28 | mut | http://www.dagimp.org/owned/bridge.jpg |
| 13:06.52 | Nivex | heheh that's great |
| 13:07.21 | mut | yea |
| 13:07.30 | mut | i gotta get 2 of em tho tk |
| 13:07.33 | mitcheloc | that is one battered sign |
| 13:07.44 | mut | incase a freak lightning storm fries my card |
| 13:08.18 | docelm0 | THATS GREAT |
| 13:11.34 | Katty | mew. |
| 13:14.21 | mut | o_O |
| 13:14.30 | iDunno | blip |
| 13:14.57 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:16.52 | docelm0 | MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW |
| 13:17.03 | macTijn | wtf? |
| 13:17.12 | docelm0 | macTijn, you gots to be new |
| 13:17.14 | mut | musta got into the cat good again |
| 13:17.17 | mut | food |
| 13:17.31 | macTijn | mut: nice typo ;) |
| 13:17.41 | macTijn | docelm0: no, just been away for a while |
| 13:17.49 | *** join/#asterisk hypnox (n=dan@cornelyn.force9.co.uk) |
| 13:17.53 | docelm0 | How long is awhile? |
| 13:17.56 | hypnox | hi guys, i am thinking of writing my own voicemail interface. Is there a better way to go about it than to directly manipulate the files in /var/spool ? |
| 13:18.13 | docelm0 | Cause I have been in this room for over 2 years and dont remember seeing much of your nick |
| 13:18.28 | *** join/#asterisk motu (n=yu@192.165.166.163) |
| 13:18.29 | macTijn | docelm0: month or 3 or so |
| 13:18.44 | docelm0 | You should know about Katty and her mew. then |
| 13:18.59 | macTijn | haven't been paying much attention the last year or so |
| 13:19.07 | mut | ya |
| 13:19.08 | macTijn | busy busy busy :( |
| 13:19.16 | mut | i don't recall you talkin in here much either |
| 13:19.24 | macTijn | could be |
| 13:19.33 | macTijn | I'm here since asterisk 0.7.2 |
| 13:19.40 | macTijn | or so |
| 13:19.48 | mutilator | man i wish kiwi were bigger |
| 13:19.58 | mutilator | i just gobbled down 3 of em in like 20 seconds |
| 13:20.02 | macTijn | heh |
| 13:20.04 | mutilator | sooo gooood |
| 13:21.18 | *** join/#asterisk kfuq (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 13:21.44 | docelm0 | I dont know my version.. Just know its been spring of 04 when I started playing.. |
| 13:22.07 | *** join/#asterisk tdonahue-laptop (n=tdonahue@208.51.101.201) |
| 13:22.21 | mutilator | woulda been.. bout same time for me |
| 13:22.31 | mutilator | may/june '04 |
| 13:22.41 | *** join/#asterisk motu (n=motu@192.165.166.163) |
| 13:23.15 | macTijn | that's when * got a bit of news from slashdot etc |
| 13:23.32 | mutilator | well when i first started here |
| 13:23.35 | *** join/#asterisk inv_arp[work] (i=junya@adsl-11-225-195.mia.bellsouth.net) |
| 13:23.38 | mutilator | they wanted me to use.. um |
| 13:23.43 | mutilator | some crap |
| 13:23.57 | mutilator | i forgot what it was |
| 13:24.02 | macTijn | that gnu thing ? |
| 13:24.08 | mutilator | so i googled voip and found asterisk |
| 13:24.22 | mutilator | lemme see |
| 13:24.42 | mutilator | i forgot the name of it |
| 13:24.42 | macTijn | bayonne |
| 13:24.46 | mutilator | no |
| 13:24.48 | macTijn | oh |
| 13:24.50 | macTijn | hmm |
| 13:24.51 | mutilator | had some java interfaces and stuff |
| 13:24.55 | macTijn | uh |
| 13:25.02 | macTijn | sounds like that stuff intel bought |
| 13:25.06 | mutilator | i keep thining S something |
| 13:25.17 | mutilator | it did sip.. |
| 13:26.07 | stoffell | hm, is there an alternative to freepbx's user/device configuration? (for roaming users) |
| 13:26.15 | motu | is it possible to have asterisk receive prorietary SIP signals from a client and have it translate them into pbx commands without writing additional code (eg a plugin or something)? |
| 13:26.42 | macTijn | mutilator: dunnow |
| 13:26.52 | macTijn | mutilator: opensource ? |
| 13:27.33 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:29.23 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
| 13:29.25 | razu | hi |
| 13:29.48 | razu | has anyone got T.38 passthrue working for good in asterisk ? |
| 13:31.42 | jaiger | does Background() have any effect on the digit queue? ie. will multiple audio files queued via a bunch of Background() cause digits to be lost? |
| 13:32.38 | jaiger | I have some channels that either lose digits or misdetect them. The channels are SIP/IAX2 |
| 13:32.57 | jaiger | and the problem is sporadic |
| 13:35.08 | *** part/#asterisk slav_jb (n=k@pirus.securax.be) |
| 13:37.42 | tzafrir_laptop | stoffell, just have two devices register as the same extension? What exactly is the problem with that? |
| 13:38.38 | tzafrir_laptop | stoffell, other than that, you could try to use regexten |
| 13:44.27 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 13:44.53 | grem_lin | Hi, could anyone help me with DISA please? When I authenticate, get a dial tone, and then try and dial an extension in the specified context I always get a fast-busy tone :S |
| 13:45.10 | *** join/#asterisk Tili (i=Tili@218.19.1.152) |
| 13:46.52 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 13:47.56 | [TK]D-Fender | stoffell : Just ring 2 tech's at one... regexten isn't worth it unless you want calls to the # that would be associated with it to be considered completely non-existant in the case of not being connected. |
| 13:49.05 | thieumS | Hello, i'm looking for a way to use field "to" instead of field "sip:" as uri |
| 13:49.44 | stoffell | [TK]D-Fender, the principle of freepbx is nice, with users logging in/out, but is it the best way, that's my "concern" |
| 13:50.39 | mutilator | macTijn: it was called vocal |
| 13:50.45 | macTijn | ah |
| 13:50.50 | macTijn | dunnow that |
| 13:52.27 | mutilator | it was out of production as of late 2003 |
| 13:52.45 | mutilator | and was what they were trying to use early '04 |
| 13:52.48 | [TK]D-Fender | FreePBX is only good for existing * people consulting their services to customers who desperately want a GUI for themselves. Besides that my general opinion is that * GUI's do not teach you * and typically turn you into a chump. |
| 13:53.17 | bkw_ | [TK]D-Fender, the same can be said for windows |
| 13:53.18 | bkw_ | :P |
| 13:53.19 | mitcheloc | just like using windows instead of linux does ;) |
| 13:53.28 | mitcheloc | lol |
| 13:53.36 | mutilator | i use windows |
| 13:53.39 | mutilator | i'm not a chump |
| 13:53.42 | mutilator | -_- |
| 13:53.43 | bkw_ | mitcheloc, linux teaches people stupid ways to do stupid things |
| 13:53.59 | bkw_ | you sure? |
| 13:54.03 | mutilator | no |
| 13:54.05 | mitcheloc | did you mean *windows? |
| 13:54.11 | [TK]D-Fender | mutilator : I said "* GUI's" - ASTERISK |
| 13:54.12 | mutilator | but i've convinced myself of it |
| 13:54.14 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:54.14 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:54.27 | mutilator | speaking of... |
| 13:54.30 | mutilator | :O |
| 13:54.30 | mitcheloc | without the chumps how do we make our money? =P |
| 13:54.52 | mutilator | playing a violin on the corner with your tophat on the ground |
| 13:55.18 | *** join/#asterisk jhava (n=icechat5@200.58.26.21) |
| 13:55.19 | trelane_ | I'm reasonably talented on the nylon guitar |
| 13:55.31 | mutilator | trelane_ outta nowhere! |
| 13:55.52 | trelane_ | mutilator, have to keep you people guessing. |
| 13:55.56 | mitcheloc | i got vocals ;) |
| 13:55.57 | mutilator | :P |
| 13:56.02 | trelane_ | mitcheloc, bish I got vocals |
| 13:56.16 | mitcheloc | trelane_: you can have backup vocals |
| 13:56.20 | trelane_ | fair enough |
| 13:56.34 | stoffell | wasn't really an answer to my question, and certainly don't want to start a Gui/No Gui discussion :p |
| 13:56.36 | trelane_ | find us a winds player and a badass percussionist and we could cover dave matthews |
| 13:56.44 | *** join/#asterisk Teeli (i=Tili@219.136.106.87) |
| 13:56.46 | mitcheloc | "asterband" |
| 13:56.48 | trelane_ | stoffell, no GUI thanks |
| 14:00.21 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 14:02.33 | jhava | Hello all, quick question: tos = 0x8b in sip.conf gives error message in log (Unable to set TOS to 184). For any value under 170 it works. Any clues ? |
| 14:03.30 | tzafrir_laptop | [TK]D-Fender, have you tried any other GUIs? |
| 14:04.54 | tzafrir_laptop | stoffell, no, I don't think it is the best way. For instance, it is kind of limited to extensions with numbers |
| 14:05.15 | tzafrir_laptop | Which is a strange assumption is you also use SIP |
| 14:07.45 | stoffell | tzafrir_laptop, hm, okay, that's true... |
| 14:08.12 | grem_lin | Hi, can anybody tell me how I can pass variables to an PHP AGI script and how these can be retreived... thanks for any help |
| 14:09.26 | tzafrir_laptop | stoffell, destar is pretty simple to set up. Though it is rather aggressive in its config rewrite |
| 14:13.05 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@c-71-228-12-47.hsd1.il.comcast.net) |
| 14:13.56 | [TK]D-Fender | tzafrir_laptop : I use ScopServ at work... |
| 14:14.58 | RoyK | zoa: ding |
| 14:15.01 | Cherebrum | I think you want tos=b8 not 8b |
| 14:15.07 | tzafrir_laptop | [TK]D-Fender, I try to avoid anything that is non-free :-( |
| 14:15.09 | Cherebrum | er tos=0xb8 |
| 14:15.33 | Cherebrum | and you have to run as root afaik for that to work |
| 14:15.57 | Cherebrum | or you could do something like this |
| 14:15.58 | Cherebrum | iptables -t mangle -A OUTPUT -p udp -m udp --sport 16384:32767 -j DSCP --set-dsc |
| 14:15.59 | Cherebrum | p-class ef |
| 14:16.10 | Cherebrum | oops.. you get the idea... |
| 14:16.13 | tzafrir_laptop | [TK]D-Fender, anyway, what does it provide that you need? |
| 14:16.25 | jhava | Cherebrum: I tried both: 0xb8 & 0x8b, checking with ethereal, 0xb8 is the right value, but cannot be set |
| 14:16.36 | Cherebrum | jhava: is asterisk running as root? |
| 14:16.39 | jhava | a wiki document about it is ambiguous |
| 14:17.02 | Cherebrum | I gave up using asterisk to set the tos bit a long time ago.. I'm using iptables to do it now |
| 14:17.10 | [TK]D-Fender | tzafrir_home : NOTHING except the fact that the bosses here wouldn't accept * without it. They "sold" their product and it was the way to get commodity equipment in and not some POS Avaya toaster... |
| 14:17.42 | [TK]D-Fender | tzafrir_laptop : It does offer Queue stats and CDR stuff but we aren't using that yet so we aren't profiting from it. |
| 14:18.18 | [TK]D-Fender | tzafrir_laptop : However I must tell you ScopServ is a mountain above FreePBX in terms of configurability and interface quality.... |
| 14:18.35 | Cherebrum | FreePBX barely works at all |
| 14:18.38 | jhava | Ok, I will implement IPtables then, I just thought that as a workaround was fine but not proper solution. |
| 14:18.58 | Cherebrum | jhava: I think it's actually a better solution |
| 14:19.02 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 14:19.04 | jhava | thanks for your help Cherebrum |
| 14:19.11 | Cherebrum | np |
| 14:19.38 | Cherebrum | you will need to make sure you specify the rtp port range you have configured in rtp.conf |
| 14:19.48 | Cherebrum | you might want to make a rule for iax2 and sip as well |
| 14:20.32 | badboyz | we currently have a voice t1 in our building, and we want to connect it to our asterisk box for backing incoming and outgoing calls... which card is reccomended to do this? |
| 14:20.44 | Cherebrum | I always edit rtp.conf first thing and change the port range from 10000:20000 to 16384:32767 |
| 14:21.07 | tzafrir_laptop | Cherebrum, why such a large range? |
| 14:21.21 | Cherebrum | it doesn't really need to be that big.. it doesn't really matter |
| 14:21.38 | tzafrir_laptop | How many concurrent RTP streams will you have? more than a 1000? |
| 14:24.18 | brad_mssw | how much does setting the tos really help? |
| 14:24.27 | brad_mssw | I'm running asterisk in user-mode now (non-root) |
| 14:24.58 | a1fa | lol |
| 14:25.04 | Cherebrum | I guess 16384:17384 would work |
| 14:25.05 | a1fa | u are supposed to run asterisk in user mode |
| 14:25.07 | badboyz | anyone have a suggestion for a good t1 card i should buy? |
| 14:25.18 | a1fa | i dont know why you freakin' about it |
| 14:25.39 | Cherebrum | brad_mssw: setting tos doesn't do anything unless your upstream routers and switches know what to do with it |
| 14:25.40 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 14:25.55 | jhava | thank you again I will follow these tips |
| 14:25.56 | Cherebrum | I wouldn't want to run asterisk as root |
| 14:26.04 | zoa | brad_mssw: that doesnt work in like 99% of the cases :) |
| 14:26.33 | a1fa | TOS & QOS are gay |
| 14:26.42 | a1fa | they dont really work over WLAN |
| 14:26.54 | Cherebrum | yes they do |
| 14:27.00 | jhava | I need TOS because I offer service within my own router and wireless network |
| 14:27.09 | a1fa | Cherebrum : sure they do.. thats why we can packet people so easy these days |
| 14:27.23 | [TK]D-Fender | badboyz : one with echo cancellation on-board is preferable |
| 14:27.25 | Cherebrum | you can set priority on rtp packets leaving your router |
| 14:27.25 | brad_mssw | Cherebrum: well, I would hope my ISP supports it, they use all cisco gear ... we use HP Procurve switches ... the only question is if my firewall (linux) will honor those |
| 14:27.54 | a1fa | [TK]D-Fender : hello |
| 14:27.55 | Cherebrum | brad_mssw: they most likely don't. they would have to setup priority queues for it |
| 14:27.58 | [TK]D-Fender | a1fa : y0 |
| 14:28.00 | badboyz | [TK]D-Fender: what card is the most cost effective one to use for doing what im looking for? |
| 14:28.10 | brad_mssw | Cherebrum: any way to tell? |
| 14:28.14 | badboyz | im really in the dark as to what i need |
| 14:28.19 | a1fa | [TK]D-Fender : still 30s expiry time |
| 14:28.24 | [TK]D-Fender | badboyz : Maybe you could elaborate more. How many channels? What kind of signalling on it, budget, etc... |
| 14:28.35 | [TK]D-Fender | a1fa : *grumble* |
| 14:28.55 | Cherebrum | brad_mssw: stress test it maybe |
| 14:28.58 | Cherebrum | or ask them |
| 14:29.09 | Cherebrum | most ISPs don't do that unless you pay extra for it |
| 14:29.13 | badboyz | tk: its 24 channel voice t1, budget would be about the 500 range for a card to interface the t1 >> asterisk |
| 14:29.18 | a1fa | [TK]D-Fender : no word from the support |
| 14:29.35 | a1fa | [TK]D-Fender : they dont care aboutz me |
| 14:30.13 | brad_mssw | Cherebrum: dunno, we have direct fiber to our ISP (we're in the same building complex) ... |
| 14:30.56 | [TK]D-Fender | badboyz : Basically only 2 real choices... and neither will have onboard EC. That be Digium's TE110p or Sangoma's A102 |
| 14:31.32 | badboyz | [TK]D-Fender: the ones w/ EC onboard are more pricy? |
| 14:31.33 | [TK]D-Fender | badboyz : However be prepared to get stuck with Zaptel software EC which works well for some, and unlivivably for others. |
| 14:32.19 | [TK]D-Fender | badboyz : Only come in 4 port versions around $2200-2400 USD which if you run into echo problems makes it worth it... |
| 14:32.39 | shiznatix | can anyone help me with spandsp? |
| 14:32.40 | badboyz | yea, thats outta range =/ |
| 14:33.39 | *** join/#asterisk faljse (n=martin@213.235.245.210) |
| 14:33.50 | [TK]D-Fender | badboyz : Well I told you the ones that fit it, but be prepared for the possiblity of an uphill battle with echo. |
| 14:33.59 | badboyz | gotcha |
| 14:34.03 | badboyz | good to know |
| 14:34.14 | faljse | hi.. my asterisk doesnt seem to reconnect to my odbc database unless i type "show odbc".. is there a reconnect parameter? |
| 14:34.25 | [TK]D-Fender | badboyz : Sometimes some tweaking can make it either perfect or "decent" but in some cases it gets ugly. |
| 14:35.03 | jaiger | badboyz, you want hardware EC trust me |
| 14:35.12 | badboyz | well |
| 14:35.22 | jaiger | badboyz, you should budget accordingly |
| 14:35.22 | badboyz | the hopes are to use the t1 as a failover |
| 14:35.25 | *** join/#asterisk op3r (n=op3r@202.71.189.66) |
| 14:35.35 | op3r | who's using vicidial here? |
| 14:35.36 | badboyz | in the event that that our voip provider(s) fail us |
| 14:35.41 | [TK]D-Fender | badboyz : as a FAILOVER?! to what? |
| 14:36.01 | [TK]D-Fender | badboyz : thats a very backwards sounding idea... |
| 14:36.23 | [TK]D-Fender | badboyz : you don't pay for 1 PRI as a FALLBACK to VoIP, more the REVERSE |
| 14:36.35 | badboyz | well the t1 is going to be scaled back |
| 14:36.43 | badboyz | cost cutting =/ |
| 14:36.50 | [TK]D-Fender | badboyz : I hope to heck you're getting it dirt cheap where you are. |
| 14:37.15 | *** join/#asterisk Ariel_ (n=Ariel@209.168.221.130) |
| 14:37.22 | Ariel_ | hello everyone |
| 14:37.56 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 14:38.20 | Ariel_ | just a small question having an issue with finding the correcty way to fix the CentOS kernel bug for compiling the zaptel. it's the one with the rwlock or something like it. does anyone have the link to fix this? |
| 14:38.58 | [TK]D-Fender | ~centosbug |
| 14:39.01 | jbot | i heard centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
| 14:39.01 | [TK]D-Fender | JBOT has abandoned us! |
| 14:39.06 | Ariel_ | thanks |
| 14:39.12 | [TK]D-Fender | now what am I supposed to use to bludgeon newbs with?! |
| 14:39.33 | [TK]D-Fender | jbot: About time molassas-script! |
| 14:39.48 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 14:40.17 | shiznatix | when i try to send a fax with spandsp the fax machine i am sending it to will ring once then it will just hang up the call. can anyone help me with this? |
| 14:40.27 | Katty | morning Ariel_ (= |
| 14:42.56 | Katty | hi file |
| 14:42.59 | Katty | you tickler you |
| 14:43.00 | file | hola |
| 14:43.04 | Katty | como estas? |
| 14:43.13 | file | not bad, not bad at all |
| 14:43.32 | file | how is the ever fabulous Katty? |
| 14:43.33 | Katty | yay! |
| 14:43.35 | Katty | estoy bein! |
| 14:43.44 | Katty | sort of, anyway. |
| 14:44.05 | trelane_ | chattr +i file |
| 14:44.08 | trelane_ | :) |
| 14:44.19 | grem_lin | Hi, could anybody tell me if something is wrong with this. I'm setting the CID in the dialplan by using Set(callerid=${CALLERIDNUM}), then calling an AGI script and executing GET VARIABLE callerid |
| 14:44.27 | file | Katty: come into my arms, that's where you belong! |
| 14:44.49 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 14:45.15 | Katty | but that's practically forever away |
| 14:45.19 | *** join/#asterisk asterboy (n=kevin@S01060204ee2b6007.ed.shawcable.net) |
| 14:46.05 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 14:46.05 | *** mode/#asterisk [+o denon] by ChanServ |
| 14:46.37 | file | Katty: :( |
| 14:46.51 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com) |
| 14:47.36 | asterboy | Grandstream phone is working fantastic...now I want the ST2030...anyone know how to get one into NA? |
| 14:48.49 | shiznatix | when i try to send a fax with spandsp the fax machine i am sending it to will ring once then it will just hang up the call. can anyone help me with this? |
| 14:49.49 | *** join/#asterisk file[desk] (n=jcolp@mctnnbsa24w-142167060049.pppoe-dynamic.nb.aliant.net) |
| 14:51.08 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 14:51.42 | TheCops | hi, there's a way to send a text message to a pager via Asterisk ? |
| 14:51.45 | [TK]D-Fender | asterboy : EW.... VoIP bottom feeding! |
| 14:52.01 | [TK]D-Fender | TheCops : lookup SMS on the wiki... |
| 14:52.08 | [TK]D-Fender | TheCops : LInks and samples all over |
| 14:52.09 | TheCops | good thanks |
| 14:52.29 | grem_lin | After I have set the value of a variable, how can I then retreive the value of it? |
| 14:52.33 | [TK]D-Fender | TheCops : so did you ever get that conversion authorized and done? |
| 14:52.35 | asterboy | lol...I dunno, seems to work excellent |
| 14:52.57 | TheCops | [TK]D-Fender, what you mean ? |
| 14:53.01 | [TK]D-Fender | grem_lin : ${myvar} |
| 14:53.16 | [TK]D-Fender | TheCops : You were looking to convert your 200 POTS setup to PRI last I checked... |
| 14:53.23 | TheCops | Ho |
| 14:53.25 | TheCops | sorry |
| 14:53.39 | grem_lin | Hrm, that won't work from an AGI script will it? I'd need to perform a command which returns the value |
| 14:53.41 | TheCops | [TK]D-Fender, the coders is doing the application right now |
| 14:53.56 | TheCops | [TK]D-Fender, I found an ISP who will sell me VoIP services |
| 14:53.58 | TheCops | b2b2c |
| 14:54.21 | [TK]D-Fender | TheCops : hmm... I use them as my fall-back ISP... don't know about VoIP for a business like yours... |
| 14:54.24 | TheCops | 4$ DID everywhere in quebec, 30$ unlimited in and out call |
| 14:54.47 | TheCops | [TK]D-Fender, Je connais bien les gens labas |
| 14:54.50 | [TK]D-Fender | TheCops : good price.... |
| 14:55.30 | [TK]D-Fender | TheCops : Si ca marche au point de qualite et qu'il-est fiable, pourquoi pas... |
| 14:55.30 | TheCops | [TK]D-Fender, they are not offering this kind of services before but they made the package for me |
| 14:55.50 | TheCops | [TK]D-Fender, 4 1000MBPS internet backbone |
| 14:55.55 | [TK]D-Fender | TheCops : if they don't do it as a normal business, that scares me.... |
| 14:55.55 | TheCops | should be stable enought |
| 14:56.03 | asterboy | how do you flash the line? *70? |
| 14:56.15 | TheCops | [TK]D-Fender, Like I said, I know a lot of ppl there |
| 14:56.32 | TheCops | I'll get a contract between us and me |
| 14:58.02 | TheCops | [TK]D-Fender, did you every played with SMS ? |
| 14:58.07 | TheCops | I dont see canada in the list |
| 14:58.13 | [TK]D-Fender | TheCops : nope.... |
| 14:58.40 | [TK]D-Fender | TheCops : I use my work as my ITSP for home now that I'm on Dry-DSL :) |
| 14:58.50 | TheCops | :)\ |
| 15:00.17 | *** join/#asterisk Dandan (i=dandan@jestem.lama.ale.mam.super.konto.na.pacanka.com) |
| 15:01.12 | Dandan | re all :) |
| 15:02.03 | asterboy | can't have two sip phones with the exact same registry can you? |
| 15:02.12 | file | the SMS that is in Asterisk is for landline, it is NOT for cellphones and stuff... |
| 15:02.14 | [TK]D-Fender | asterboy : 2nd reg will kill the first... |
| 15:02.16 | file | different monster |
| 15:02.25 | asterboy | ya, that's what I thought. |
| 15:02.31 | mutilator | haha |
| 15:02.36 | mocker | Woo, digium says my card is bad. |
| 15:02.41 | asterboy | note to self...PBX not KSU |
| 15:02.41 | mocker | That means I'm not crazy! :) |
| 15:02.44 | file | the world is mine! |
| 15:02.46 | mutilator | cleaning lady just ran into my office like something was chasing her |
| 15:02.53 | mutilator | there was a snake next to the door |
| 15:02.59 | mutilator | she looked like it was going to kill her |
| 15:03.02 | [TK]D-Fender | asterboy : You'd need either SIP-B support(estimated for summer '06) or reg as a different ext, and just ring simultaneous to the other phone. |
| 15:03.26 | file | SIP-B is very interesting, but I have ideas on how we can do it |
| 15:03.34 | asterboy | ya, I've been reg as different....just hate the sip.conf getting so big. |
| 15:03.45 | [TK]D-Fender | asterboy : Whats your idea of big? |
| 15:04.01 | asterboy | lol...well, big for 3 phones. |
| 15:04.34 | file | I know there's something in the wake of your smile |
| 15:04.38 | asterboy | with multi extensions....gets big for such a small setup. |
| 15:04.50 | *** mode/#asterisk [+o file] by file[laptop] |
| 15:05.05 | *** join/#asterisk brif8 (n=chatzill@rrcs-71-41-50-162.se.biz.rr.com) |
| 15:05.21 | asterboy | oh oh, what now brif8? |
| 15:05.25 | asterboy | :P |
| 15:05.53 | brif8 | I'm trying to connect two * boxes http://pastebin.com/655715 It "rings" the server 2 but the phones does not ring nor does the console on server 2 see anything ? |
| 15:06.14 | [TK]D-Fender | file : yes.. BILGE... |
| 15:06.18 | brif8 | can anyone please help me find the dumb mistake I'm making |
| 15:06.30 | brif8 | hi asterboy: |
| 15:06.34 | asterboy | reads like one is overtaking the other...I just had a similar situation. |
| 15:06.36 | file | brif8: it can't send the packet to the second machine |
| 15:06.41 | asterboy | with phones though. |
| 15:07.02 | brif8 | file: I have port forwarding open on the firewall direct to the * boxes |
| 15:07.10 | asterboy | ping? |
| 15:07.18 | file | okay, I'm just telling you what it says |
| 15:07.23 | asterboy | use "sip debug" |
| 15:07.31 | brif8 | iax surely |
| 15:07.44 | [TK]D-Fender | brif8 : Stop using "user=" and use the entry name in [] |
| 15:08.13 | iCEBrkr | Grrrr. |
| 15:08.21 | x86 | what is SIP-B ? |
| 15:08.23 | shiznatix | If I have a fax machine connected to my asterisk box through a zapata card how do I send faxes to it? I know its like: Zap/2/ but then what? |
| 15:08.29 | asterboy | oh ya...* to *, dust to dust, iax to iax |
| 15:08.35 | iCEBrkr | The Wiki on sending Faxs claims you have can Answer() the channel before you Dial() |
| 15:08.38 | asterboy | ~sipb |
| 15:08.43 | [TK]D-Fender | brif8 : And you are using "host" at the same time as registering which is BACKWARDS |
| 15:08.45 | iCEBrkr | BZZZZZZZZZZT |
| 15:08.50 | asterboy | ~sip-b |
| 15:08.55 | asterboy | hmmm |
| 15:09.15 | asterboy | jbot, sipb is is SIP for Business soon to be supported by * |
| 15:09.17 | jbot | okay, asterboy |
| 15:09.19 | [TK]D-Fender | shiznatix no 2nd slash. Dial(Zap/2,30) |
| 15:09.22 | brif8 | [TK]D-Fender: so you are suggesting to drop user and secret and use register ? |
| 15:09.30 | shiznatix | [TK]D-Fender, thanks |
| 15:09.40 | a1fa | ~sipb |
| 15:09.41 | jbot | somebody said sipb was is SIP for Business soon to be supported by * |
| 15:09.53 | file | soon is relative. |
| 15:09.59 | a1fa | jbot, no sipb is SIP for Business soon to be supported by * |
| 15:10.01 | jbot | okay, a1fa |
| 15:10.05 | a1fa | ~sipb |
| 15:10.06 | jbot | methinks sipb is SIP for Business soon to be supported by * |
| 15:10.20 | a1fa | jbot, thanks |
| 15:10.20 | jbot | pas de quoi, a1fa |
| 15:10.28 | Dandan | hehehe |
| 15:10.30 | Dandan | sipb? |
| 15:10.51 | [TK]D-Fender | brif8 : Keep secret, the [meadow] *IS* the user, and your register should terminate with /[context] for the place to have incoming calls land in on. Go re-read the WIKI on how to set this all up.. |
| 15:11.05 | *** join/#asterisk bweschke (n=bweschke@66.152.225.74) |
| 15:11.06 | [TK]D-Fender | lol, french jbot! |
| 15:11.12 | *** join/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net) |
| 15:11.21 | asterboy | why did you redo that a1fa? |
| 15:11.24 | [TK]D-Fender | jbot : va-t'en ostie! |
| 15:11.49 | [TK]D-Fender | asterboy : For propaganda of course! |
| 15:11.58 | brif8 | [TK]D-Fender: I did that is where I got the iax.conf information Example 2 of dual servers on wiki |
| 15:12.07 | asterboy | qwell does that too |
| 15:12.24 | asterboy | Is there some sort of contest to populate jbot? |
| 15:12.39 | Dandan | [TK]D-Fender: I got a word today that 101a (test card) is on its way... |
| 15:12.50 | *** join/#asterisk salviadud (n=ralfalfa@201.137.164.110) |
| 15:12.52 | brif8 | example 2 from wiki has register with user, secret etc.. which is a better example to follow then ? |
| 15:12.59 | Dandan | if it works I will buy 104 |
| 15:13.37 | *** join/#asterisk BugKham (n=HamYai@125.24.5.218) |
| 15:14.06 | BugKham | where to download the Asterisk Span DSP? |
| 15:14.06 | [TK]D-Fender | Dandan : 101a? |
| 15:15.38 | BugKham | or it's built in to the core asterisk release? |
| 15:16.06 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 15:16.54 | zoa | www.softswitch.org |
| 15:17.01 | zoa | iirc |
| 15:18.10 | [TK]D-Fender | Dandan : I presume you mean A101 |
| 15:18.10 | *** join/#asterisk cybergypsy (n=cybergyp@APoitiers-156-1-41-189.w86-213.abo.wanadoo.fr) |
| 15:18.13 | dlynes | no web site is configured at that address :) |
| 15:19.17 | Flauto | Apr 12 10:19:09 WARNING[9941]: file.c:498 ast_openstream_full: File minutes does not exist in any format |
| 15:19.33 | dlynes | www.soft-switch.org |
| 15:19.57 | Flauto | what is that |
| 15:20.47 | *** join/#asterisk CMike (i=daemon@c-544171d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 15:24.38 | shiznatix | can you make 1 zap channel call another zap channel when they are on the same card? |
| 15:25.33 | shiznatix | like i have a incoming line that connects to the outside world come into my zapata card then i am trying to have a fax that comes in on that to goto another port on that same zapata card which has a fax machine connected to it. is this possible? |
| 15:25.47 | zoa | yes |
| 15:25.49 | zoa | that is possible |
| 15:26.35 | shiznatix | ok then why does it hang on '- Attempting native bridge of Zap/1-1 and Zap/4-1' |
| 15:26.39 | shiznatix | it just stops there |
| 15:26.50 | file | because that's perfectly normal |
| 15:27.28 | shiznatix | why? |
| 15:27.53 | file | what else is it supposed to do? it's bridged the two channels together... so audio and DTMF/etc goes between them |
| 15:29.01 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 15:29.07 | shiznatix | well the fax machine on Zap/4 does not get any call of any kind |
| 15:29.09 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 15:29.17 | shiznatix | it just sits there and looks stupid. why? |
| 15:29.20 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:29.35 | shiznatix | the fax comes in and it starts to redirect it to the fax machine line but no luck |
| 15:30.05 | Flauto | insecure=very now is changed? |
| 15:30.21 | file | well what is Zap/4, what kind of card? did you hook a regular phone up to see if it rings? |
| 15:30.35 | mutilator | anyone know a device that'll push a t1 ~1+ miles of copper |
| 15:30.37 | mutilator | 24guage |
| 15:30.57 | file | have to debug the situation, take out as many variables as you can to narrow down the problem |
| 15:31.02 | mutilator | something i can put modular cards in possibly |
| 15:32.12 | LostFrog | I have calls coming in to my * box through SIP, and then routed to another office via IAX. When the other office transfers calls back to the first office, how can I avoid the extra router? I.e. SIP -> Office1 -> Office2 -> Office1? |
| 15:32.38 | LostFrog | -r |
| 15:32.49 | file | LostFrog: you IAX2 native transfers so that Office1 talks to itself, the call will migrate off of Office2 |
| 15:32.52 | file | er you = use |
| 15:33.02 | file | give me just a chance, let's go out and dance |
| 15:33.22 | sevard | So.. I can't figure this out. My VoIP provider requires that I dial my area code before any local prefix, is there a way with dialplans to match a dialed prefix on a sip line and append an area code before it dials out on the iax trunk? |
| 15:33.36 | shiznatix | file, the Zap 1-4 is the same card, my zapata card. the regular phone line comes in on the 1st port (yes this was tested and works). the fax machine is connected to the 4th port. the fax comes in on the 1st port (regular line) and then seams to try to connect to the 4th port with the fax machine but it fails somewhere doing that. here is my output from asterisk and my extensions.conf: http://pastebin.com/655780 |
| 15:33.47 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.89) |
| 15:33.47 | file | zapata card is very generic, we make lots of cards |
| 15:33.53 | file | and so do others |
| 15:33.58 | file | I'm assuming a TDM400 card |
| 15:34.10 | mutilator | that sangoma ds3 thing doesn't work with zap does it? |
| 15:34.13 | shiznatix | file, if your talking to me then you are right |
| 15:34.39 | file | shiznatix: and module 4 is an FXS module? |
| 15:35.21 | file | well, the channel you're calling... |
| 15:35.36 | file | goes to a port on the TDM400 that has an FXS module? (making sure here) |
| 15:35.43 | Katty | FILE |
| 15:35.48 | shiznatix | file, http://pastebin.com/655785 there i just added my zapata.conf |
| 15:35.50 | file | KATTY |
| 15:35.52 | Katty | let's hug |
| 15:35.53 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:35.53 | sevard | PANTS |
| 15:35.56 | sevard | LOUD NOISES. |
| 15:35.58 | Dandan | [TK]D-Fender: yes A101 |
| 15:36.03 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 15:36.04 | Katty | i'm needing a hug. |
| 15:36.06 | [TK]D-Fender | UGH |
| 15:36.26 | file | shiznatix: I believe you have the wrong module for that port, if you're connecting a phone or fax machine to Asterisk it needs an FXS module - not FXO |
| 15:36.51 | Katty | :< |
| 15:36.51 | salviadud | hiya! |
| 15:37.01 | salviadud | it's on the cheek |
| 15:37.04 | shiznatix | file, but it is fxs? "signalling=fxs_ks" |
| 15:37.09 | salviadud | i'm a nice guy, come on |
| 15:37.15 | file | shiznatix: you signal fxo ports with fxs, and fxo ports with fxs |
| 15:37.21 | sevard | ass cheek? |
| 15:37.30 | file | wow |
| 15:37.31 | *** join/#asterisk Eggplant (i=No@dsl-469.cascadeaccess.com) |
| 15:37.32 | file | I just repeated myself |
| 15:37.32 | Katty | nice guys hug. |
| 15:37.35 | file | and fxs ports with fxo |
| 15:37.37 | sevard | hahaha |
| 15:37.45 | shiznatix | file, alright. so i have to make zap/4 a fxo since zap/1 is a fxs? |
| 15:38.08 | file | shiznatix: what color are the modules on your TDM400 card? |
| 15:38.12 | sevard | I wouldn't want to hug salviadud unless I wanted to get poked. |
| 15:38.14 | LostFrog | Ok.. I must be stupid.. I don't have native transfers turned off.. do I need to do something special when I dial the extension to transfer the call? |
| 15:38.19 | salviadud | you've never been to mexico |
| 15:38.30 | [TK]D-Fender | shiznatix : Signalling is "fxo_ks" for an FXS module (green) |
| 15:38.40 | file | LostFrog: no |
| 15:38.52 | file | [TK]D-Fender: I want to make sure he has the right module first... he might have two FXO |
| 15:38.54 | shiznatix | file, they are red |
| 15:39.01 | LostFrog | Does it matter that my phones are SIP? |
| 15:39.06 | brettnem | ~seen connor |
| 15:45.31 | jbot | connor <n=billy@198-144-165-65.knx.tn.nxs.net> was last seen on IRC in channel #asterisk, 21d 17h 33m 23s ago, saying: 'Hey guys.. question.. I want to setup a pre-queue.. I want to queue up calls and then send them down a pri to another phone system.. I want to limit the number of calls the other phone system gets to about 2 or 4 calls.. How can I do ... |
| 15:45.45 | [TK]D-Fender | shiznatix : then those are for plugging in LINES |
| 15:45.45 | file | shiznatix: you can't plug phones into those |
| 15:45.46 | [TK]D-Fender | shiznatix : like file said.. |
| 15:45.46 | shiznatix | damn |
| 15:45.46 | *** join/#asterisk timscott (n=a@d198-166-221-177.abhsia.telus.net) |
| 15:45.46 | file | http://www.digium.com/en/products/hardware/s110m.php you need one of those |
| 15:46.49 | shiznatix | file, if i put my fax machine into my IDSN card it would probably not work right? or is this at least worth a try? |
| 15:46.50 | file | I don't know ISDN. |
| 15:46.50 | file | 'nor do I use it, configured it, whatever |
| 15:46.50 | LostFrog | If anyone would help me with this, I would paypal them a few $$. |
| 15:46.51 | file | LostFrog: okay you - server 1 calls server 2 using IAX2, which calls someone's SIP phone, they transfer to an extension that calls server 1 using IAX2 |
| 15:47.07 | sevard | LostFrog: gender identity? |
| 15:47.17 | LostFrog | file: correct. |
| 15:47.27 | file | server 2 should try to do a native transfer between both sides, so server 1 will talk to itself - pastebin your iax.conf minus passwords, plus console output with iax2 debug |
| 15:47.28 | LostFrog | ok.. give me a few. |
| 15:47.28 | file | I'm here all day. |
| 15:47.32 | sevard | file: how about mine? :) |
| 15:47.32 | file | sevard: yes, there is - it's standard dialplan logic |
| 15:47.32 | sevard | It's confusing crap. |
| 15:47.40 | file | exten => _878XXXX,1,Dial(IAX2/myuser@myprovider/1506${EXTEN}) |
| 15:59.33 | file | matches something like 8781234 and appends 1506 in front before sending to myprovider |
| 15:59.34 | [TK]D-Fender | file : EEK, exchange-lever rounting? |
| 15:59.41 | sevard | What if I have a whole list of prefixes to match and on top of that still want to dial to other area codes? |
| 15:59.45 | [TK]D-Fender | level* |
| 15:59.48 | file | [TK]D-Fender: yeah I'm evil |
| 15:59.56 | file | sevard: then learn dialplan logic and you can figure something out... it's really not that bad |
| 16:00.00 | [TK]D-Fender | file : oh God.. you're not going on about that from the alst time we mentioned it are you? With your "psycho" LCR SQL deal? |
| 16:00.37 | file | [TK]D-Fender: that worked. |
| 16:00.44 | [TK]D-Fender | file : You SQL one? |
| 16:00.57 | file | oh yes, it worked |
| 16:01.18 | file | but that has nothing to do with this |
| 16:01.19 | *** part/#asterisk thieumS (n=darkmind@bea75-1-82-234-122-35.fbx.proxad.net) |
| 16:01.21 | *** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju) |
| 16:01.22 | DoktorGreg | ok, im having a tiny problem, where it sounds like soft sip clients are over compressing |
| 16:01.23 | *** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org) |
| 16:01.24 | asterboy | jbot, sipb is also defined here: http://www.bandwidth.com/wiki/article/SIP-B |
| 16:01.49 | jbot | asterboy: okay |
| 16:01.50 | grem_lin | Can anyone help me with PHP AGI, I'm about to give up :P All I want to do is get a global variable I've set, I've tried using the example on www.voip-info.org but to no avail :( |
| 16:01.51 | asterboy | jbot, sipb is also http://www.bandwidth.com/wiki/article/SIP-B |
| 16:02.09 | jbot | asterboy: okay |
| 16:02.09 | asterboy | ~jbot |
| 16:02.30 | jbot | somebody said jbot was only marginally useful at best, He got a C- on his Turing Test |
| 16:02.30 | asterboy | what happened to the poor fellow? |
| 16:02.31 | asterboy | ~docs |
| 16:02.52 | jbot | extra, extra, read all about it, 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 |
| 16:02.53 | asterboy | ~jbotgetyourassbackhere |
| 16:03.56 | Katty | jbot: hello? |
| 16:04.37 | jbot | Howdy Bub |
| 16:04.37 | Katty | jbot needs hugging :< |
| 16:04.50 | asterboy | they killed jbot...you bastards. |
| 16:04.54 | lokkju | http://rafb.net/paste/results/YCtn6M40.html - full log shows answer, then wait, then playing beep, then nothing untill I hangup - hangin on the Playback, obviously, but *why* |
| 16:04.55 | backblue | how can i see if there is a spawn down or up? |
| 16:04.56 | DoktorGreg | closer examination reveals... only problem im having with xlite is... it sounds over compressed |
| 16:13.21 | DoktorGreg | ok maybe im examining it too closely |
| 16:13.56 | *** join/#asterisk Pazzo (n=thomas@host130-250.pool8172.interbusiness.it) |
| 16:14.02 | *** part/#asterisk Pazzo (n=thomas@host130-250.pool8172.interbusiness.it) |
| 16:14.03 | *** join/#asterisk nguyep (n=chatzill@64.34.203.231) |
| 16:14.16 | nguyep | i know this is a faq but an1 know how to compile chan_alsa.so? |
| 16:14.16 | asterboy | jbot, where have you been? Having a toke weren't you! |
| 16:14.33 | salviadud | share the stash bro |
| 16:14.34 | asterboy | lol |
| 16:14.34 | wunderkin | DoktorGreg, what do you mean by over-compressed |
| 16:14.35 | shiznatix | does anyone have any idea why when I try to use spandsp everything seams like it is working just fine but when I am recieving a fax it just times out and when i try to send a fax it rings the fax machine once then hangs up? |
| 16:14.35 | mutilator | anyone ever hear of a cordless phone not working with an ata? |
| 16:14.42 | mutilator | if a corded phoen is hooked, it gets dialtone and can recieve calls |
| 16:14.44 | mutilator | but cordless gets no dialtone |
| 16:14.47 | mutilator | but it can recieve calls |
| 16:14.52 | salviadud | lies, all lies |
| 16:14.55 | salviadud | i use a cordless phone on my sipura 3000 |
| 16:14.57 | salviadud | works ok |
| 16:14.57 | salviadud | i get my dialtone and everything |
| 16:14.58 | salviadud | i prank with that phone in the bathroom |
| 16:15.00 | salviadud | calling embassy after embassy, offering tacos |
| 16:15.00 | salviadud | they never learn... |
| 16:15.02 | trelane_ | I want tacos :( |
| 16:15.20 | salviadud | where do you live? |
| 16:15.21 | trelane_ | USA |
| 16:15.21 | trelane_ | got any in pink? |
| 16:15.24 | trelane_ | pink tacos are the best |
| 16:15.25 | DoktorGreg | wunderkin, the client i am using is making sounds like... |
| 16:15.27 | DoktorGreg | compression artifacts |
| 16:15.27 | DoktorGreg | xlite |
| 16:15.29 | salviadud | you better off eating at crapdonna's |
| 16:15.31 | salviadud | taco bell are not real tacos btw |
| 16:15.31 | *** join/#asterisk heka (n=heka@82.114.68.124) |
| 16:15.34 | DoktorGreg | but its only happening on this one client |
| 16:15.35 | DoktorGreg | iaxComm works, |
| 16:15.36 | salviadud | real tacos are manly |
| 16:15.38 | *** join/#asterisk Pazzo (n=thomas@host130-250.pool8172.interbusiness.it) |
| 16:15.40 | salviadud | some have 2 types of meat, aguacate and cheese |
| 16:15.42 | Nugget | http://nucleartacos.com/ |
| 16:15.42 | DoktorGreg | without the compression artifact sounds... |
| 16:15.44 | salviadud | if you're really man enough, you add some salsa |
| 16:15.45 | trelane_ | salsa++ |
| 16:15.45 | trelane_ | and not the bland crap sold in the us, or the stuff that tastes like vinagar |
| 16:21.27 | salviadud | that's why most of us mexicans do our salsa at home |
| 16:21.28 | Nugget | right. because there's no good salsa in the entire united states. |
| 16:21.32 | DoktorGreg | no, but the US has the best hot sauces bar none |
| 16:21.35 | DoktorGreg | and I disagree |
| 16:21.38 | DoktorGreg | Daves insanity Salsa is quite good |
| 16:21.41 | Nugget | there's thousands of good salsa available in the united states |
| 16:21.41 | salviadud | if you call tabasco a hot sauce |
| 16:21.47 | salviadud | you're a bunch of pussies |
| 16:21.49 | Nugget | you're the only one here who has mentioned tabasco. |
| 16:21.51 | salviadud | tabasco is like freaking tomato sauce down here |
| 16:21.54 | salviadud | well. it SELLS |
| 16:21.56 | DoktorGreg | I use pure cap brand capsation |
| 16:21.58 | salviadud | and they market it as hot |
| 16:21.58 | DoktorGreg | http://www.hotsauceworld.com/purecap.html |
| 16:21.59 | grem_lin | When running an AGI script and executing a command to asterisk is "GET VARIABLE callerid" valid, it seems to come up with an error saying it cannot find the application - any help would be *greatly* appreciated |
| 16:22.01 | Nugget | we make our nuclear tacos with red savina. It's pretty righteous. |
| 16:22.06 | salviadud | Screaming Sphincter Hot Sauce, 5oz. |
| 16:22.09 | salviadud | lol |
| 16:22.14 | salviadud | well, i am truly impressed, you really hit the salsa underworld now |
| 16:22.39 | Hmmhesays | i feel kind of bad for calling in sick |
| 16:22.41 | DoktorGreg | Ring of Fire Xtra Hot Hot Sauce |
| 16:22.51 | sevard | don't feel bad |
| 16:22.55 | sevard | get off irc |
| 16:22.55 | Hmmhesays | aww you twisted my arm |
| 16:22.58 | DoktorGreg | ahh here we go, they have one that is 1.5 million scovill units |
| 16:23.07 | DoktorGreg | I didn't realize they made hotter than pure cap |
| 16:27.21 | *** join/#asterisk Pazzo (n=thomas@host130-250.pool8172.interbusiness.it) |
| 16:27.30 | DoktorGreg | Im somehow in the mood for a bloody marry now |
| 16:29.10 | asterboy | ya tabasco is tomato juice |
| 16:29.14 | timscott | You guys are SO HARDCORE. |
| 16:29.15 | asterboy | the hottest ones have oils that keep the spice buring and are difficult to wash down |
| 16:29.17 | timscott | Liking hot sauce is like BDSM |
| 16:29.20 | timscott | It's like getting pleasure out of ripping yourself up. |
| 16:29.21 | asterboy | You need one of those handicap bars next to the toilet if you've consumed the hottest. |
| 16:29.22 | *** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 16:29.23 | DoktorGreg | I was with a bud eating chicken wings with various hot sauces at Hooters one night |
| 16:29.23 | sevard | god damnit you fools |
| 16:29.23 | sevard | tobasco sauce is not hot sauce |
| 16:29.23 | sevard | it is pepper sauce |
| 16:29.23 | sevard | PEPPER SAUCE. |
| 16:29.23 | timscott | Yeah. |
| 16:29.23 | timscott | It tastes good though. |
| 16:29.23 | sevard | it says so on the damned bottle, it is NOT hot sauce. |
| 16:29.23 | g__ | Sorry, I think I have the wrong channel. |
| 16:29.24 | asterboy | having a coach to help you through the hot sauce ring of fire is nice also. |
| 16:29.25 | DoktorGreg | He was still going, I started crying |
| 16:29.26 | asterboy | "push"...."push" |
| 16:29.26 | sevard | you can't go around saying "tobasco sauce is pussy ass hot sauce" it's not freaking hot sauce. |
| 16:29.31 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:29.39 | sevard | i had this argument with an idiot at the ihop "boy why you be a damned pussy and like that there tobasco sauce" me "uhh, it's pepper sauce" |
| 16:29.44 | shiznatix | does anyone have any idea why when I try to use spandsp everything seams like it is working just fine but when I am recieving a fax it just times out and when i try to send a fax it rings the fax machine once then hangs up? |
| 16:29.44 | DoktorGreg | I like tobasco sauce |
| 16:29.45 | sevard | I love the stuff |
| 16:29.47 | DoktorGreg | Its just not that hot |
| 16:29.47 | salviadud | unfortunately the good hot sauces are the one's you can only use in moderation, too much and you blow up like a steamroller |
| 16:29.48 | lokkju | http://rafb.net/paste/results/YCtn6M40.html - full log shows answer, then wait, then playing beep, then nothing untill I hangup - hangin on the Playback, obviously, but *why* |
| 16:29.50 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
| 16:29.53 | DoktorGreg | Then my computer beeped, and half my paper was gone, and i was like huh? |
| 16:36.01 | g__ | Background: I have a customer complaining about audio quality problems, but I can't figure out why. Question: is there an iax debugging mode I could use to log problems? (Such as "packets never showed up" or "packets showed up very late") |
| 16:36.01 | timscott | iax2 debug |
| 16:36.01 | sevard | this is the first time i've ever had to use AC in minnesota |
| 16:36.01 | timscott | at the cli |
| 16:36.16 | g__ | Will it tell me about that stuff? |
| 16:36.17 | timscott | well |
| 16:36.20 | timscott | it might |
| 16:36.21 | timscott | but it's rather cryptic |
| 16:36.24 | timscott | What I would suggest is downloading a program called "iperf" |
| 16:36.26 | timscott | I'll link you, g. |
| 16:36.26 | terrapen | any service providers here? |
| 16:36.26 | g__ | Would 'iax2 jb debug' be better? thanks timescott. |
| 16:36.31 | timscott | http://dast.nlanr.net/Projects/Iperf/ |
| 16:36.33 | g__ | terrapen: plenty |
| 16:36.36 | terrapen | any that deploy Sipura stuff to their customers? |
| 16:36.38 | timscott | g_, if you download that program, you can analyze UDP traffic with it |
| 16:36.44 | terrapen | I'm trying to get some info on how I can configure these SPA-942 phones from a text file |
| 16:36.46 | timscott | you install it on both the home and remote machine |
| 16:36.50 | timscott | and you can send custom streams and such, and it'll tell you what is happening. |
| 16:36.53 | terrapen | reconfiguring 350 phones via a web interface is not something that I look forward to |
| 16:36.57 | timscott | if you want to go through the trouble of reading through all the "iax2 debug" crap, g__, then that's cool |
| 16:36.57 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 16:36.57 | timscott | But I just use iperf, wherever possible |
| 16:36.58 | timscott | that's my 0.02 |
| 16:36.58 | DoktorGreg | hey i have to do a similar number of nortel phone and set did through F**CONFIG |
| 16:36.58 | g__ | timscott: neat stuff. Does it take output from both servers and give you a usable report? |
| 16:36.58 | timscott | yes |
| 16:36.58 | timscott | It does. |
| 16:36.58 | brif8 | I can ping from server 1 to Server 2 but still it won't phone connect |
| 16:36.59 | shiznatix | when I use spandsp to send a fax the other fax machine rings once then hangs up. Can anyone help me please? |
| 16:37.09 | timscott | Use POTS for your faxes. |
| 16:37.10 | timscott | :p |
| 16:37.10 | g__ | timscott: thanks.. up until now I've been using MRTG and "smokeping". |
| 16:37.11 | timscott | MRTG doesn't tell you anything useful |
| 16:37.14 | timscott | It doesn't tell you jitter or latencies, does it? |
| 16:37.17 | timscott | I'm not familier with smokeping, though. |
| 16:37.21 | g__ | But smokeping does.. we've been sending data from it to our ISP for months now :) |
| 16:37.22 | timscott | oh |
| 16:37.24 | Katty | mister fender, you around? |
| 16:37.27 | timscott | Well, maybe smokeping will work for you, I'm not really familier with it. :/ |
| 16:38.35 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:38.42 | timscott | I'll hit it on google. |
| 16:51.23 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 16:51.23 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.2.6 and Zaptel 1.2.5 Released! (March 27, 2006) -=- http://www.asterisk.org/ -=- AMP/FreePBX/Asterisk@Home users should join #freepbx for support |
| 16:52.27 | brif8 | if on ServerB I have in sip.conf a register, auth and [serverA], where do I set up the user and password on Server A ? |
| 16:53.09 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-174-51.lsanca.fios.verizon.net) |
| 16:53.24 | LostFrog | file: http://pastebin.com/655949 |
| 16:53.31 | jaiger | nettie, I guess some sort of custom map command/agi. I name my sip users by extension |
| 16:53.47 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 16:53.55 | brif8 | because on Server B's console I keep getting Registeration for 'user@serverA' timed out |
| 16:53.58 | nettie | so, 101, 102, 103 and so on |
| 16:53.59 | nettie | ? |
| 16:54.33 | file | LostFrog: ah it's an attended transfer? |
| 16:54.36 | [TK]D-Fender | asterboy : I've worked with CWCID before, what about it? |
| 16:54.38 | LostFrog | yes |
| 16:54.43 | nettie | well I think I'll just end up specifying the callerid= in sip.conf |
| 16:55.11 | file | that's probably why... the way we do attended transfers is fun, so the bridge might not know that they're the same technology and do a true native bridge... |
| 16:55.42 | LostFrog | Grr.. I don't have a way around it. |
| 16:55.49 | LostFrog | I don't trust parking. |
| 16:55.50 | *** join/#asterisk skkip (n=Skipper@216.160.91.91) |
| 16:56.07 | *** join/#asterisk telamon (n=telamon@pac.isn.net) |
| 16:56.08 | LostFrog | We can't use blind transfer. |
| 16:56.41 | file | I'm pondering if there is a way... |
| 16:57.00 | asterboy | [TK]D-Fender, Just trying to get call-waiting and waiting indication on my Polycom working. |
| 16:57.20 | asterboy | first: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Flash |
| 16:57.26 | file | can you turn on debug in logger.conf set verbosity to high, turn off iax2 debug using iax2 no debug, and pastebin what it says? |
| 16:57.27 | asterboy | trying to get flash working |
| 16:57.33 | telamon | Anyone know why I can't login to my voicemail when I call VoiceMailMain(mailbox@context) but it works when I just use VoiceMailMain(@context)? It prompts me for the mailbox number with both, but it doesn't work when I specify the mailbox in the call. |
| 16:57.42 | LostFrog | sure.. I already have debug on. |
| 16:57.47 | asterboy | then i'd like to be able to see callerid of next call |
| 16:57.52 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 16:57.56 | LostFrog | I just have to pull out the approriate parts. |
| 16:57.58 | gbodemantv | hey all |
| 16:58.05 | gbodemantv | so who is an xlite god in here |
| 16:58.06 | brif8 | [TK]D-Fender: could you possibly explain what I'm missing (half my brain I know, but what else) |
| 16:58.07 | gbodemantv | ?? |
| 16:58.15 | asterboy | When having more than 1 DYNAMIC_FEATURES, how do you assign the variable? |
| 16:58.32 | asterboy | just add another line in globals? |
| 16:59.34 | gbodemantv | trying to get call waiting sound to turn off in xlite |
| 16:59.40 | gbodemantv | any ideas? |
| 17:01.06 | [TK]D-Fender | asterboy : You'd need to set up a custom featuremap entry in features.conf for something like that.... |
| 17:01.59 | [TK]D-Fender | asterboy :Forget about CWCID on a ZAP channel.... the card doesn't listen for stuff like that... it can't afford to interfere with the EXISTING call. |
| 17:02.15 | [TK]D-Fender | asterboy : Analog line CW + * = dumb idea |
| 17:02.37 | *** join/#asterisk Foxtro (i=foxtro@30-78-246-201.adsl.terra.cl) |
| 17:02.38 | Foxtro | :) |
| 17:02.39 | Foxtro | hi |
| 17:03.01 | asterboy | ok, good to know. |
| 17:03.08 | Foxtro | where i can download a sip phone for web page ? |
| 17:03.33 | gbodemantv | if it is not possible in xlite to turn off, can you tell asterisk to turn off? |
| 17:03.43 | asterboy | I setup this line in features.conf |
| 17:03.45 | asterboy | zapflash => *3,caller,flash,() ; Flash Button |
| 17:03.49 | telamon | Foxtro: I think there is one called mozphone. |
| 17:04.01 | Foxtro | :D |
| 17:04.05 | Foxtro | where can download? |
| 17:04.19 | Dandan | ~google |
| 17:04.21 | jbot | google is, like, a search engine found at http://www.google.com/ |
| 17:04.25 | Foxtro | :P |
| 17:04.26 | [TK]D-Fender | I know *I'm* completely turned "off" by X-Lite ;) |
| 17:04.26 | asterboy | but the zap channel is not listening for the *3 or any key combo |
| 17:04.42 | Dandan | [TK]D-Fender: firefly all the way! |
| 17:04.51 | [TK]D-Fender | asterboy : You may need a parm in your dial command... |
| 17:04.59 | telamon | Foxtro: http://www.voip-info.org/tiki-index.php?page=MozPhone has more info, http://moziax.mozdev.org/ is the main page with download links |
| 17:05.12 | [TK]D-Fender | Dandan : Firefly was nifty, but had limited functionality and crashed a lot. |
| 17:05.17 | asterboy | jbot, google is also not the only search engine in existance, try clusty.com and mama.com for a better choice |
| 17:05.49 | Dandan | [TK]D-Fender: i tried cubix but it sux, and firefly is surpsisingly good enough for me |
| 17:06.08 | Dandan | any better softphones (going to europe in a few days) |
| 17:06.13 | Dandan | for PC/PDA? |
| 17:06.13 | *** join/#asterisk Strom_M (n=strom@gateway.digium.com) |
| 17:06.37 | asterboy | [TK]D-Fender, also set this exten => s,5,Set(DYNAMIC_FEATURES=zapflash) |
| 17:06.46 | [TK]D-Fender | danalien : Better off with X-Lite in most cases... |
| 17:07.00 | Dandan | blah i hate that... |
| 17:07.04 | Dandan | but ok :) |
| 17:07.16 | [TK]D-Fender | asterboy : Ok, well beyod my general advise I have no practical experience with that... you'll need to figure it out, but keep going... |
| 17:07.36 | asterboy | [TK]D-Fender what parameter are you suggesting? |
| 17:07.44 | [TK]D-Fender | Dandan : in the "free" category, then again, eyeBeam kills the rest... |
| 17:07.57 | Dandan | oh sipps! |
| 17:08.05 | Dandan | i think my company bought the license for me... |
| 17:08.16 | *** join/#asterisk Hmmhesays (n=Neg@72.24.105.126) |
| 17:08.19 | Dandan | thx for reminding me |
| 17:09.24 | *** join/#asterisk jsaunders (n=root@216.86.121.58) |
| 17:11.22 | gbodemantv | Dandan: any idea about how I can disable call waiting |
| 17:11.28 | gbodemantv | in either asterisk or xlite? |
| 17:12.33 | Dandan | gbodemantv: i do not know xlite |
| 17:12.43 | Dandan | in gbod: err... yeah there is... but do not remember :D |
| 17:12.44 | Dandan | hold on |
| 17:14.34 | asterboy | so zap channels + Call Waiting = A No |
| 17:15.04 | Foxtro | his! |
| 17:15.37 | Foxtro | the web phone i need is for clients.. |
| 17:15.38 | Foxtro | (: |
| 17:15.40 | Foxtro | :( |
| 17:15.43 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 17:15.50 | salviadud | idefisk |
| 17:15.58 | gbodemantv | call waiting is beeping in users ears |
| 17:15.58 | salviadud | or somethin |
| 17:16.04 | gbodemantv | very dustracting |
| 17:16.11 | LostFrog | file: http://pastebin.com/655993 |
| 17:16.39 | *** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net) |
| 17:18.58 | file | LostFrog: yeah guess it won't do it for when an attended transfer happens |
| 17:19.25 | asterboy | gbodemantv, I'm working on call waiting now...to disable in zap, you can add the callwaiting=no in zapata.conf. |
| 17:19.50 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.222) |
| 17:20.04 | Foxtro | [Airwolf] <-- chile ? |
| 17:20.05 | asterboy | But I want mine working so I don't miss calls. |
| 17:20.27 | asterboy | Doesn't seem like call waiting and * get along in ZAP anyway. |
| 17:20.59 | asterboy | seems like * is ignoring any key press after a call is established. |
| 17:20.59 | file | LostFrog: just for kicks put notransfer=no in general and in the peers/users... never know |
| 17:21.24 | gbodemantv | astorboy: if I can just turn off the sound for it that would do |
| 17:21.34 | gbodemantv | can't seem to find info on xlite to do it |
| 17:22.51 | asterboy | gbodemantv, some phones like the Polycom seem to have a variable you can set it sip.cfg like <CALL_WAITING lcl.cpt.chord.cp.1.6.freq.1="450" lcl.cpt.chord.cp.1.6.level.1="-19" lcl.cpt.chord.cp.1.6.onDur="400" lcl.cpt.chord.cp.1.6.offDur="0" lcl.cpt.chord.cp.1.6.repeat="1"/> |
| 17:23.22 | asterboy | looks like it controls sound and duration |
| 17:24.05 | *** join/#asterisk Mike (n=mike@201.138.165.154) |
| 17:24.07 | asterboy | How do you get zap channels to monitor a call for key presses? |
| 17:24.13 | asterboy | callprogress? |
| 17:24.47 | Mike | guys for high load what codec could help me more ilbc or gsm? |
| 17:25.14 | [av]bani | asterboy: thats the wrong one |
| 17:25.36 | [av]bani | i disabled callwaiting tone on polycoms, it's non intuitive and non documented |
| 17:25.43 | [av]bani | (thanks polycom!) |
| 17:25.56 | asterboy | how did you disable it? |
| 17:26.27 | asterboy | godemantv wants to know...I'm trying to ENABLE mine |
| 17:26.33 | [av]bani | ?! |
| 17:26.37 | [av]bani | it's enabled by default |
| 17:27.26 | *** join/#asterisk MacDome (n=eseidel@A17-255-104-111.apple.com) |
| 17:28.03 | brif8 | please can someone assist on this dual server issue :( |
| 17:28.28 | *** join/#asterisk MacDome (n=eseidel@A17-255-104-111.apple.com) |
| 17:29.06 | [TK]D-Fender | [av]bani : no, he's talking about sending a hook-flash to an analog LINE using a polycom. not dealing with it as a SIP concept. |
| 17:29.30 | [Airwolf] | Foxtro, what is a chile ? |
| 17:29.47 | [av]bani | [TK]D-Fender: you can't do that from any sip phone, that's something that a gateway would have to do |
| 17:30.06 | [av]bani | program your gateway to interpret dtmf combo as a hook flash |
| 17:30.26 | [av]bani | or asterisk, if you have zap channels to pstn on it |
| 17:30.42 | brif8 | serverA has Registration for 'voip@serverB' timed out, trying again while serverB has Registration from '<sip:voip@serverB' failed for 'serverA' |
| 17:31.04 | [TK]D-Fender | [av]bani : You can trigger it from a script in features.conf, but the fact is that CW is too impractical for his tastes. |
| 17:31.23 | [TK]D-Fender | [av]bani : And we're talking a Zap analog card |
| 17:31.27 | brif8 | in each sip.conf I have [servera] and [serverb] in both servers sip.conf |
| 17:31.42 | Foxtro | can connect skype with asterisk ? |
| 17:31.48 | jaiger | is there any way to show in the * log (cmd prompt) exactly what digits are detected/received? |
| 17:32.30 | asterboy | no, I'm trying to setup Call Waiting so I can flash my line from a SIP phone...but * ignores any key presses after the call is established. |
| 17:32.33 | [av]bani | [TK]D-Fender: then his gateway is the asterisk pbx... problem is hook-flash is unreliable |
| 17:32.45 | [av]bani | everything on analogue is unreliable :/ |
| 17:33.09 | [av]bani | asterboy: you need to add parameters to Dial() so you can do functions post-bridge |
| 17:33.58 | asterboy | have that |
| 17:34.17 | gbodemantv | and I am dialing from xlite |
| 17:34.21 | gbodemantv | no interface for it there |
| 17:34.39 | asterboy | exten => s,6,Dial(SIP/Distance&SIP/Distance2&SIP/Distance3,15,tT) |
| 17:34.40 | brif8 | is there another wiki beside voip-info that will help, since I have followed their dual asterisk server and it does not work |
| 17:35.22 | asterboy | [av]bani, are you taking about tT ? |
| 17:35.33 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 17:36.05 | LostFrog | file: didn't work with blind transfer.. :( |
| 17:36.19 | LostFrog | Do I have to use the Transfer() app? |
| 17:36.23 | file | no |
| 17:36.30 | file | it's not a transfer as you know it... |
| 17:36.35 | file | did you do what I said and put notransfer=no? |
| 17:36.41 | LostFrog | yep.. both ends. |
| 17:36.46 | LostFrog | and restarted both ends. |
| 17:37.16 | [av]bani | asterboy: is hook flash really a function in features.conf ? |
| 17:38.02 | *** join/#asterisk brettnem (n=brettnem@nemeroff.com) |
| 17:38.03 | LostFrog | I wish I had a lab of two asterisk boxes. |
| 17:38.37 | LostFrog | Maybe I need to build one. |
| 17:39.18 | [av]bani | aha |
| 17:39.27 | [av]bani | asterboy: you want to create a special extension which executes a flash() |
| 17:39.35 | iCEBrkr | I'll become pure energy. Once I've entered in the neural net...my birth cry will be the sound...of every phone on this planet ringing in unison. |
| 17:39.36 | [av]bani | then transfer to it with # |
| 17:39.36 | [av]bani | :/ |
| 17:40.00 | iCEBrkr | I love doing asterisk testing and the whole office rings. |
| 17:40.08 | [av]bani | yay? |
| 17:40.36 | trelane_ | iCEBrkr, yawn |
| 17:41.32 | asterboy | [av]bani, zapflash => #3,caller,flash,() ; Flash Button |
| 17:41.48 | LostFrog | This is an awesome time to play with NFS root. |
| 17:42.17 | LostFrog | Becasue I am short on HDDs. |
| 17:42.30 | jaiger | iCEBrkr, I've firewalled you so you can forget ringing my phones |
| 17:42.52 | trelane_ | I like it when someone gets a phone call and "White House Switchboard" shows up on their caller ID |
| 17:44.12 | Foxtro | can connect skype with asterisk ? |
| 17:44.29 | timscott | 202.456.1212 |
| 17:44.32 | Foxtro | users sky call asterisk users? |
| 17:45.18 | timscott | err 1414 |
| 17:46.04 | [TK]D-Fender | iCEBrkr : I don;t think there will be enough REN to support you, sorry!! |
| 17:46.18 | LostFrog | iCEBrkr: That's from an old SF story by Asimov or Clarke, isn't it? |
| 17:46.55 | asterboy | no thats from LawnMowerMan iirc |
| 17:48.30 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:49.56 | *** join/#asterisk Arcu (n=436c6f90@maynard.cac.psu.edu) |
| 17:49.56 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-48.claranet.co.uk) |
| 17:50.30 | *** join/#asterisk argos73 (n=mike@w010.z208036240.chi-il.dsl.cnc.net) |
| 17:51.43 | *** join/#asterisk oej_ (n=olle@apollo.webway.se) |
| 17:53.16 | *** join/#asterisk radhios (n=radhios@bue215-194.is.net.ar) |
| 17:53.31 | *** part/#asterisk oej_ (n=olle@apollo.webway.se) |
| 17:54.02 | asterboy | I'll keep playing with it, here is what I'm trying: |
| 17:54.04 | asterboy | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Flash |
| 17:54.27 | asterboy | the zapflash does not work for me cause my zap channel is ignoring any key presses. |
| 17:56.40 | LostFrog | file: is there a possibility that the RTP packets are staying on the first * box? |
| 17:57.57 | file | LostFrog: IAX2 doesn't use RTP, but the way it's supposed to work is that if IAX2 is bridged to itself, then a native bridge function will be called - and it's in the protocol that both parties will try to talk directly, and if possible - then the middle box drops out |
| 17:58.27 | LostFrog | It said "Native bridge" on the console |
| 17:58.39 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:58.45 | file | that's deceiving |
| 17:59.05 | generalhan | whats up everyone ! |
| 17:59.07 | file | if it did occur, you would see more IAX2 activity and the call would eventually disappear |
| 17:59.50 | generalhan | im having a serious issue with my 2 digium cards together, and i need some advice .... |
| 17:59.51 | *** join/#asterisk Rowter (n=Silver@201.138.157.112) |
| 18:00.04 | file | generalhan: if you ask questions, people may answer |
| 18:00.07 | file | it's funny how that works :D |
| 18:00.16 | LostFrog | lol |
| 18:00.23 | LostFrog | Don't ask to ask.. just ask. |
| 18:00.25 | generalhan | file ... i have been asking this same question for over a week now and i cant get it fixed |
| 18:00.31 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
| 18:00.44 | file | okay, people in here are here out of their free time... they don't have to help you |
| 18:00.49 | file | and there's no guarantee you will get help |
| 18:00.51 | generalhan | and im not asking about the question, im asking about advice on how to approach support @ digium |
| 18:01.01 | file | well you didn't say that :) |
| 18:01.15 | file | but, you call them up... and explain your problem |
| 18:01.27 | generalhan | file: everyone in here has been MORE than helpful to me .. you included. i know how to ask a question in here |
| 18:01.30 | *** part/#asterisk Arcu (n=436c6f90@maynard.cac.psu.edu) |
| 18:01.45 | Juggie | file is it warm in NB today? |
| 18:01.51 | file | Juggie: not too bad |
| 18:01.57 | brettnem | ~gwypf |
| 18:02.02 | jbot | somebody said gwypf was Get What You Pay For - this channel is full of volunteers who are here to help you. However, we can't hold your hand. If you need a specific problem solved immediately, there is a list of for-hire consultants located at: http://www.voip-info.org/tiki-index.php?page=Asterisk+Consultants |
| 18:02.03 | Juggie | its 21c here in ottawa! :) |
| 18:02.04 | Juggie | in April! |
| 18:02.23 | Juggie | which is 70f for anyone who doesnt do celcious |
| 18:02.32 | generalhan | well i have ... and they want to ssh in ... the only problem is that this is a live server and it cant be brought down (which is what they need to do) they arent open when we are closed so i want to senf them an email with all the important info and see if they can help that way ... i just need to know what it is that they will need to see in the email to give them enough information to beable to help me ! |
| 18:03.02 | file | why don't you ask them? |
| 18:03.07 | generalhan | lol |
| 18:03.12 | generalhan | i didnt want to have to call them again |
| 18:03.22 | generalhan | but i guess i can |
| 18:03.41 | trelane_ | generalhan, you should already have your ticket number (you *DID* write it down, right? |
| 18:03.49 | generalhan | file: what is the most amount of digium hardware you have configured in one server ? |
| 18:04.04 | wunderkin | 42! |
| 18:04.08 | generalhan | ... |
| 18:04.15 | file | me? personally? I don't use any hardware |
| 18:04.21 | brettnem | hardware sux |
| 18:04.21 | generalhan | hmm |
| 18:04.37 | brettnem | I bet you can contract someone to do it |
| 18:04.41 | trelane_ | I've got a couple tdm400's in a system here |
| 18:04.42 | generalhan | you were the only person i could think of that wasnt involved in this conversation that we were having yesterday |
| 18:04.58 | trelane_ | generalhan, what cards are you using, what are they doing, what do you want them to do? |
| 18:05.24 | *** join/#asterisk Iam8up|lpy (n=iam8up@cpe-24-210-253-66.woh.res.rr.com) |
| 18:06.13 | generalhan | well i have a TE210 (that has been working just great for 2 months now) i just got a TDM40B to do some fax lines and i cant get it working... if i make entries for zaptel ans zapata for the new card, the wct4xxp drivers (for the TE card) thinks that the FXS ports are part of that card and it wants them to be setup as spans |
| 18:06.42 | *** join/#asterisk brif8 (n=chatzill@rrcs-71-41-50-162.se.biz.rr.com) |
| 18:06.43 | generalhan | i have had many people look at my zaptel.conf and zapata.conf files and everyone says it should be setup correctly ... but still no go |
| 18:06.49 | Iam8up|lpy | can anyone tell me what the iax port is? |
| 18:06.59 | brif8 | can someone help debug this IAX with me http://pastebin.com/656103 |
| 18:07.28 | Iam8up|lpy | i'm going to netscan the network to find the ip address of the digium ata; i don't know how to work our router (mikrotik, some latvian thing) |
| 18:07.33 | file | generalhan: module load order matters, never forget where they are! |
| 18:07.35 | file | hahahaha... that's great |
| 18:08.03 | generalhan | what do you mean? |
| 18:08.20 | generalhan | cause ive tried loading the TDM first ... ive tried loading the TE first ... i dont know what else to try |
| 18:08.33 | file | but Digium support is your best bet |
| 18:08.36 | generalhan | im loading wctdm for the TDM40B and wct4xxp for the TE210 |
| 18:08.39 | file | they're there for this sort of stuff |
| 18:08.45 | trelane_ | generalhan, sounds like your zaptel.conf config is wrong though I havn't used the 210, have you tried using genzaptelconf.sh? |
| 18:09.08 | generalhan | trelane_: i have never even heard of that |
| 18:09.12 | generalhan | lol ... is that bad ? |
| 18:09.40 | trelane_ | generalhan, nah, there's a script floating around online that is very good at autoconfiguring zaptel hardware |
| 18:10.49 | trelane_ | see http://lists.digium.com/pipermail/asterisk-dev/2004-December/007911.html for more info+links to the scripts |
| 18:11.04 | tzafrir_laptop | trelane_, actually, it has recently been commited into zaptel trunk |
| 18:11.08 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.96.Dial1.SanJose1.Level3.net) |
| 18:11.24 | trelane_ | tzafrir_laptop, right but as this is a production system I'm not assuming his zaptel is up to date or running trunk |
| 18:11.37 | tzafrir_laptop | wget http://svn.digium.com/svn/zaptel/trunk/xpp/genzaptelconf |
| 18:12.16 | generalhan | im using zaptel 1.2.5 |
| 18:12.18 | tzafrir_laptop | ~genzaptelconf |
| 18:12.26 | jbot | methinks genzaptelconf is a script to general zaptel.conf and zapata.conf snippet for TDM cards, availble from http://tzafrir.org.il/genzaptelconf or from the Rapid zaptel packaged. Ask tzafrir about it. ignore warning about missing ast-cmd. |
| 18:12.34 | generalhan | trelane_: you can look at my zaptel if you would like to ... im pulling it up for the digium people anyway, it looks like its perfectly set up. http://generalhan.pastebin.ca/49194 |
| 18:12.45 | Netgeeks | anyone here have a good recommendation for a gigabit ethernet switch? Looking for something that is reliable and has redundant power if possible. QoS not required (will be doing VoIP traffic only) |
| 18:13.09 | [TK]D-Fender | Netgeeks : Why do you need gigabit for VoIP? |
| 18:13.48 | *** join/#asterisk ToTo (n=ToTo@host188-67.pool8260.interbusiness.it) |
| 18:13.52 | generalhan | tzafrir_: http://tzafrir.org.il/genzaptelconf is no good |
| 18:13.56 | Netgeeks | The switch will have as many as 20 servers with 200 concurrent calls per server using ulaw |
| 18:14.01 | tzafrir_laptop | jbot, no, genzaptelconf is a script to general zaptel.conf and zapata.conf snippet for most zaptel hardware. availble from http://svn.digium.com/svn/zaptel/trunk/xpp/genzaptelconf. Ask tzafrir about it |
| 18:14.06 | jbot | tzafrir_laptop: okay |
| 18:14.07 | *** join/#asterisk azzie (n=az@azzie.net) |
| 18:14.20 | generalhan | haha |
| 18:14.45 | [TK]D-Fender | Netgeeks how much traffic / port? |
| 18:14.53 | trelane_ | generalhan, <tzafrir_laptop> wget http://svn.digium.com/svn/zaptel/trunk/xpp/genzaptelconf |
| 18:14.55 | *** join/#asterisk nahirean (n=nahirean@67.132.43.2) |
| 18:15.02 | timscott | kancho |
| 18:15.02 | tzafrir_laptop | generalhan, it should work just as well. I'm not aware of any required changes in different versions |
| 18:15.08 | trelane_ | generalhan, no need to look at those configs, genzaptelconf will replace them |
| 18:15.32 | [TK]D-Fender | Netgeeks : though if you're talking inter-server traffic, I guess GBIT would be fairly cheap, esp w/o QoE or anything special. I Hear HP ProCurves are pretty highly respected. |
| 18:16.19 | nahirean | Hello, I am trying to use BackgroundDetect as it's intended (to detect voice/dtmf) with this syntax: exten => s,3,BackgroundDetect(30silence|500|1000|) Basically, it should only take half a second of no vocal/dtmf for it to go to "talk" right? All this does, however is play the entire 30second silence file .. is it an issue with my syntax, or perhaps with dtmf detection/voice detection? |
| 18:16.20 | tzafrir_laptop | I'm open for feedback regarding PRI cards. Also I haven't tried a TDM2400 card yet |
| 18:16.32 | Netgeeks | Fender: no more than 80Mbps per port I would think. depends upon how much we can load the servers, we are starting out with 200 concurrent per server which is 36M, but I think they will go to 400 |
| 18:16.41 | Iam8up|lpy | can anyone tell me what the default iax port is? i'm looking for this ata on my lan... |
| 18:16.47 | SplasPood | What are people recommending for a reasonable 1 or two line SIP phone these days... Wanna give it to our support people for working from home... |
| 18:17.09 | nahirean | Iam: 4568 I beleive |
| 18:17.42 | *** join/#asterisk DeeJay[2] (n=bleh@office.abi.ca) |
| 18:18.02 | Iam8up|lpy | 4569 <-- found on the ethereal wiki =) |
| 18:18.04 | DeeJay[2] | How can I limit a sip user to use only 1 line + call waiting (2) at any moment? |
| 18:18.08 | [TK]D-Fender | Netgeeks : What the hell.. go for a nice ProCurve switch... |
| 18:18.12 | SplasPood | something on the cheaper side... speaker phone is not needed |
| 18:18.27 | [TK]D-Fender | tzafrir_laptop : What kind of feedback concerning PRI cards? |
| 18:18.43 | nahirean | Splas, get a ebayed adapter and use analog phones ;) |
| 18:18.55 | DeeJay[2] | I need to let my user choose the agent he want so I cannot use a locked CPE. |
| 18:19.13 | gbodemantv | was just told that it cannot be turned off from xlite |
| 18:19.16 | Netgeeks | Fender: thanks, looking at them now on HP's site |
| 18:19.17 | DeeJay[2] | But I don't want this user to be able to use many of my PRI channels.. |
| 18:19.19 | tzafrir_laptop | regarding genzaptelconf . Specifically, what to write as the timing source. And also if all the signalling parameters are correct. I begin they tend to differ by country |
| 18:19.41 | *** join/#asterisk stoffell_home (n=stoffell@d51A4D7C9.access.telenet.be) |
| 18:19.51 | SplasPood | nah: well, other than that option :) |
| 18:20.26 | generalhan | i love calling places and hearing the same hold music i have !! lol |
| 18:20.31 | nahirean | splaspood: how much are the SPA941/2s nowadays? They're easy to configure.. may be a bit priecy though |
| 18:20.38 | *** part/#asterisk stoffell_home (n=stoffell@d51A4D7C9.access.telenet.be) |
| 18:20.45 | generalhan | i mean its obvious that youll hear that from digium ... but i catch it alot now-a-days |
| 18:20.48 | [TK]D-Fender | nahirean : Where are you located? |
| 18:20.59 | nahirean | Fender, Tri-state area |
| 18:21.08 | *** join/#asterisk jsaunders (n=root@216.86.121.58) |
| 18:21.12 | DoktorGreg | <PROTECTED> |
| 18:21.23 | gbodemantv | astorboy: any idea how to turn off call waiting in asterisk |
| 18:21.25 | gbodemantv | ?? |
| 18:21.30 | [TK]D-Fender | nahirean : FORGET the SPA's. Go Polycom. far better products |
| 18:21.37 | tzanger | polycom rock |
| 18:21.41 | DoktorGreg | I am trying to undertand how how pri channels relate to extensions.conf contexts |
| 18:21.44 | timscott | Aastra. |
| 18:21.54 | tzanger | except that you cant' set vlan, username or pass from DHCP.. heh |
| 18:21.55 | nahirean | D-Fender, I was just suggesting a 941 for a simple to use IP phone for Splas ;) Polycom isnt exactly "Friendly" without Asterisk behind it ;) |
| 18:21.57 | generalhan | i use Aastras here ... i love them |
| 18:22.09 | [TK]D-Fender | aastra is still flakey and the screens are fugly even though they are backlit. |
| 18:22.12 | tzanger | I might be buying a citel gateway for norstar though, just put this system out of its misery |
| 18:22.18 | SplasPood | [TK]D-Fender: We deploy polycom normally... we're lookin for something cheaper to stick in people's homes for off-hours support duties |
| 18:22.21 | timscott | your fugly. I saw aastra phones and was smitten. |
| 18:22.24 | tzafrir_laptop | DoktorGreg, in zapata.conf , context= tells you where in the dialplan incoming calls start |
| 18:22.24 | timscott | *you are |
| 18:22.26 | generalhan | well i just use their economy SIP Phone the p112i |
| 18:22.31 | tzanger | buying a couple of F1000Gs for testing too |
| 18:22.32 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 18:22.33 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 18:22.34 | SplasPood | nahirean: well we ahve asterisk behind em, duh :) |
| 18:22.35 | generalhan | SIP -112i rather |
| 18:22.52 | nahirean | Splas, I wasn't aware of how you were configuring em ;) |
| 18:23.12 | tzafrir_laptop | DoktorGreg, and then in the dialplan (extensions.conf) you can dial to Zap/ channels, such as the PRI ones |
| 18:23.17 | SplasPood | if the IP 301 was $75 or so, we'd buy those |
| 18:23.23 | *** join/#asterisk reth (i=reth@2001:16d8:20:2:211:11ff:fe58:35cb) |
| 18:23.42 | DoktorGreg | tzafrir, thanks |
| 18:23.48 | *** join/#asterisk sercz (n=sercz@dslb-084-056-233-021.pools.arcor-ip.net) |
| 18:24.07 | nahirean | Anyone here familiar with the BackgroundDetect function? exten => s,3,BackgroundDetect(30silence|500|1000|) <-- this is only playing the .gsm and not stopping when I speak.. is it a syntax error to anyone's knowledge? |
| 18:24.58 | jaiger | during the first few seconds of an inbound call (or perhaps Background()) my * seems to miss DTMF tones. |
| 18:25.01 | jaiger | has anyone seen this? |
| 18:25.39 | jaiger | my callers need to wait before entering an extension or else the first digit or two is missed |
| 18:25.58 | *** join/#asterisk stoffell_home (n=stoffell@d51A4D7C9.access.telenet.be) |
| 18:26.06 | SplasPood | jaiger: add a Wait(1) or so before the Background() ? |
| 18:26.15 | jaiger | SplasPood, already have that |
| 18:26.31 | jaiger | they seem to need to wait a few seconds into the prompt |
| 18:26.36 | DoktorGreg | jaiger, as an usability point that i have noticed |
| 18:26.45 | DoktorGreg | people are used to waiting |
| 18:26.57 | jaiger | not my wife |
| 18:27.00 | DoktorGreg | dont fix long wait times until THEY complain about them |
| 18:27.05 | DoktorGreg | ahh |
| 18:27.12 | jaiger | I'm already getting complaints |
| 18:27.20 | jaiger | from customers and family |
| 18:27.32 | LostFrog | I get complaints all the time.. doesn't mean I do anything about them. :) |
| 18:27.36 | *** join/#asterisk dlynes_ (n=dlynes@216.251.149.66) |
| 18:28.32 | dlynes_ | Has anyone run into a problem with asterisk whereby a caller is put on hold, then taken off hold...when they come back, the person receiving the call can hear the person talking, but the person on hold cannot hear the other person? |
| 18:28.38 | brif8 | I got it ! ! ! ! ! ! ! at last the only thing I not it doesn't forward caller ID information I guess there is no way to do that is there ? :) :D |
| 18:29.16 | dlynes_ | The SIP device in question is an Aastra 9133i, which is connected to an asterisk box that is connected to another asterisk box via iax; the remote asterisk box received the call via pri |
| 18:29.57 | brif8 | I have DIAL (IAX2/ServB/888,80) and then exten => 888,Macro(inbound). how can I pass the caller ID information ? |
| 18:30.44 | *** join/#asterisk Ciber311 (i=Ciber@216-211-204-48.firstgate.net) |
| 18:30.57 | dlynes_ | Set(CALLERIDNUM(34982508)) |
| 18:31.06 | dlynes_ | Set(CALLERIDNAME("BLAHBLAHBLAH")) |
| 18:31.58 | brif8 | dlynes: no I pass from one * server to another. The first server gets the caller ID information. and I pass the call using IAX2 to the other server. I would like the actual CallerID information that the first * server received not my own |
| 18:32.18 | dlynes_ | brif8, Is the call bridged? |
| 18:32.29 | brif8 | not sure |
| 18:32.45 | dlynes_ | If it's bridged, it should pass the caller id information automatically |
| 18:32.51 | dlynes_ | Check your logs |
| 18:33.03 | dlynes_ | Your logs will tell you if the call was bridged, or not |
| 18:33.09 | Ciber311 | anyone using a gxp-2000 here? |
| 18:33.29 | stoffell_home | Ciber311, a lot of people |
| 18:33.50 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 18:33.53 | dlynes_ | Last night, it seemed like half the channel was using gxp-2000's :) |
| 18:33.58 | Ciber311 | lol |
| 18:34.07 | stoffell_home | dlynes_, and the night before, and before, ... ;) |
| 18:34.27 | ljam | [TK]D-Fender: I don't want to know your name, I just want, !, !, ! |
| 18:36.00 | dlynes_ | mirc is commercial software now? |
| 18:36.06 | asterboy | that was me trying to figure the chinese logogrhams. |
| 18:36.10 | LostFrog | shareware. |
| 18:36.20 | asterboy | irssi rules! |
| 18:36.29 | brif8 | dlynes: it would appear not http://pastebin.com/656173 |
| 18:36.34 | SplasPood | mirc was always shareware |
| 18:36.49 | asterboy | there is no mirc...only irssi |
| 18:37.07 | stoffell_h | asterboy, I've been playing with irssi also, also looks nice, but you have to get used to it :d |
| 18:37.23 | dlynes_ | isn't irssi some buggy piece of software for linux? |
| 18:37.26 | brif8 | dlynes: is there a way to bridge the call by changing a setting ? |
| 18:37.28 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 18:37.47 | dlynes_ | brif8, no...it happens automatically, if it can make the two channels compatible |
| 18:37.59 | *** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il) |
| 18:38.06 | dlynes_ | brif8, If you're using 'Answer', it's not bridging though |
| 18:38.26 | brif8 | dlynes: what makes the channels compatible ulaw, alaw ? |
| 18:38.44 | brif8 | no I'm not using answer on either side |
| 18:38.46 | dlynes_ | brif8, if that's the case, you'll probably need to get the callerid from the one channel, and set it on the new channel |
| 18:39.09 | Romik | anybody can advice about problem of 3 way transfer from the queue ? I getting following loop of error message - chan_zap.c: We're Zap/50-1, not Zap/50-2<ZOMBIE>" or "chan_zap.c: We're Zap/50-1, not SIP/???" |
| 18:39.11 | dlynes_ | brif8, If you're answering on iax channel and dialing on an iax channel, .. |
| 18:40.13 | *** join/#asterisk Ciber311 (i=Ciber@216-211-204-48.firstgate.net) |
| 18:40.22 | *** join/#asterisk pigpen2 (n=mark@207.71.48.222) |
| 18:40.41 | brif8 | no I pickup the call on Zap (T1) and re-route using IAX2 to a SIP IP Phone |
| 18:40.43 | brif8 | [macro-inbound] |
| 18:40.48 | brif8 | http://pastebin.com/656182 |
| 18:41.15 | pigpen2 | Hi all, I was having issues with chan_agent. If I statically define my agents in the queues.conf file, will it bypass the use of chan_agent? |
| 18:41.39 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 18:41.43 | znoG | hrm, should a P3 500 MHz box with 128MB be enough to handle 20-25 extensions? |
| 18:41.47 | pigpen2 | I have an open ticket with Digium regarding this, but I don't have 30 mins to be on hold.... |
| 18:42.07 | dlynes_ | brif8, normally, you would have exten => xxxx,1,Dial(SIP/myfubarsipnode) |
| 18:42.07 | Ciber311 | is there any way to map the 4 line buttons on the gxp-2000 to my 4 zap lines? |
| 18:42.14 | dlynes_ | brif8, and it should just pass caller id |
| 18:42.22 | *** join/#asterisk b_52CEntos (n=b_52Cent@adsl-190-124-192-81.adsl.iam.net.ma) |
| 18:42.24 | dlynes_ | brif8, are you sure you've even got caller id on your line? |
| 18:42.33 | pigpen2 | znoG, If you want it to be decent, I would not. |
| 18:42.40 | brif8 | yes on the T1 |
| 18:43.01 | dlynes_ | brif8, Try putting in a noop statement to test your theory |
| 18:43.15 | dlynes_ | brif8, then you'll know if you're at least receiving the caller id signalling info |
| 18:43.17 | asterboy | Digium has 30 min wait times? |
| 18:43.22 | brif8 | and I do exten s,7,Dial(SIP123&SIP/334,40,tr) |
| 18:43.22 | asterboy | yikes! |
| 18:43.32 | *** join/#asterisk MikeJ__ (n=vircuser@c-71-228-12-47.hsd1.il.comcast.net) |
| 18:43.34 | asterboy | They better start hiring more support staff. |
| 18:43.46 | dlynes_ | asterboy, Dood...I've been on hold with digium sometimes for close to an hour |
| 18:43.57 | asterboy | That's fucking rediculous! |
| 18:44.00 | pigpen2 | asterboy, yeah..the last few times it was about 30min... |
| 18:44.10 | pigpen2 | maybe they were busy...or taking a beer brake... |
| 18:44.11 | asterboy | The manager there need a kick in the ass. |
| 18:44.23 | LostFrog | They are too popular for their own good. |
| 18:44.23 | pigpen2 | well, I am about to call...so i will time it. |
| 18:44.42 | Ciber311 | guys am i supposed to do anything special for asterisk to use one of my other incoming lines if the line being called is busy? |
| 18:45.03 | asterboy | Management needs to get their act together. Hopefully the calls are not routed to India. |
| 18:45.07 | dlynes_ | Ciber311, is it an analog line? |
| 18:45.11 | Ciber311 | yes |
| 18:45.19 | Ciber311 | 4 regular pots lines |
| 18:45.26 | dlynes_ | Ciber311, enable call forward when busy on it with the telco |
| 18:45.41 | ghento | Hi folks. I'm trying to execute a php script with agi, and I do: 'exten =>...,agi,checker.php|${CALLERID(num)} ..but get the error '../var/lib/asterisk/agi-bin/checker.php' No such file or directory' even though I know the file is in there, with chmod a+x |
| 18:45.42 | pigpen2 | ringing..... |
| 18:45.42 | Ciber311 | so i gotta call verizon? gah |
| 18:45.47 | asterboy | or get your telco to setup a rotary |
| 18:45.51 | dlynes_ | Ciber311, also make sure you give it about 4 or 5 seconds before answering the line with your setup |
| 18:46.02 | brif8 | ok I'll add that thanks guys |
| 18:46.07 | brif8 | esp. dlynes :) |
| 18:46.09 | dlynes_ | Ciber311, that way you don't pick up on the first couple of rings |
| 18:46.48 | dlynes_ | Ciber311, *72 is usually call forward unconditional; I can't recall what the industry standard is for call forward when busy |
| 18:47.07 | dlynes_ | Ciber311, but that'll only work if every line has its own did |
| 18:47.19 | dlynes_ | Ciber311, if you have a bunch of anonymous overlines, that won't work |
| 18:47.32 | dlynes_ | Ciber311, then the only solution is asterboy's solution |
| 18:47.59 | dlynes_ | Ciber311, but with call forward when busy, that's usually something you can set up without calling the telco |
| 18:49.03 | pigpen2 | Got an operator, who is verifying my ticket... |
| 18:49.20 | pigpen2 | because the automated system didn't understand my dtmf |
| 18:49.31 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:49.46 | dlynes_ | pigpen2, The only way I've been able to get digium's system to recognize dtmf is by sending inband |
| 18:50.47 | pigpen2 | Well, so far they don't know what to do with me, but they say they will figure it out... |
| 18:52.31 | pigpen2 | Dam ...I have a tech...level 1 |
| 18:53.08 | *** part/#asterisk b_52CEntos (n=b_52Cent@adsl-190-124-192-81.adsl.iam.net.ma) |
| 18:58.39 | *** join/#asterisk Assid (n=assid@203.115.64.8) |
| 18:58.47 | *** join/#asterisk lzhang (n=rjrae@adsl-69-153-32-71.dsl.snantx.swbell.net) |
| 18:59.25 | lzhang | how does call pickup work? when I dial *8, I get a busy even though I have reloaded asterisk and have it in features.conf |
| 19:00.00 | [TK]D-Fender | lzhang : got it defined in your phone definition? |
| 19:00.16 | lzhang | where would I do that |
| 19:00.42 | lzhang | do you mean sip.conf? |
| 19:01.36 | lzhang | I suppose I need to find where to define the pickup group also |
| 19:01.44 | asterboy | ok, I'm thinking that getting SIP to flash a ZAP channel is just not going to happen...as you said [TK] |
| 19:01.47 | asterboy | <PROTECTED> |
| 19:02.02 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 19:02.15 | asterboy | it would work if the code was passed the ZAP channel. |
| 19:02.31 | asterboy | and was able to reconnect the SIP |
| 19:02.35 | [TK]D-Fender | lzhang : that would be good idea... |
| 19:03.13 | [TK]D-Fender | asterboy : you could do some sort of script to pick it up... wouldn |
| 19:03.16 | [TK]D-Fender | t be too hard |
| 19:03.22 | sevard | Is there a NANPA Vertical Service Code for a wakeup / callback service? or would you sluff those off under local assignment (*94-*99) |
| 19:03.31 | asterboy | or set the variable |
| 19:05.01 | [TK]D-Fender | asterboy : not realistic. I don't believe you get channel inheritance from a features.conf created channel. |
| 19:05.03 | dlynes_ | sevard, local assignment; NANPA doesn't define one |
| 19:05.31 | sevard | dlynes: Is it bad practice to start adding *100-... for local assignment stuff? |
| 19:05.41 | dlynes_ | sevard, I wouldn't know...sorry |
| 19:05.53 | sevard | But it's common to have *97 and *98 be voicemail, right? |
| 19:06.19 | dlynes_ | sevard, I just know the feature you're looking for (typical of hotels, and what-not) doesn't have NANPA vertical service code |
| 19:06.38 | dlynes_ | sevard, Nortel usually uses *980, *981, *982, *983 |
| 19:06.43 | *** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
| 19:06.53 | sevard | Would I be correct in assuming there are standards for non-nanpa assigned service codes? |
| 19:06.56 | dlynes_ | sevard, Or Feature 980, ... |
| 19:07.12 | dlynes_ | sevard, Maybe manufacturer specific standards, yeah |
| 19:07.21 | sevard | I see, no real industry accepted standards. |
| 19:07.26 | dlynes_ | sevard, not really |
| 19:07.43 | sevard | I suppose southren-bell might not be bad to follow after |
| 19:09.09 | dlynes_ | Probably not, assuming they have a wakeup call feature |
| 19:09.17 | asterboy | Seems to trying to dial the SIP Domain portion: |
| 19:09.19 | asterboy | Executing Dial("SIP/Distance2-57d1", "Zap/g1/192.168.1.8:5060|90|twTW") in new stack |
| 19:09.55 | asterboy | Thats when pressing "#" |
| 19:09.56 | sevard | dlynes_: I would just like to follow some sort of guideline... confusing users isn't fun :) |
| 19:10.08 | asterboy | Otherwise the Flash( function give this: Executing Flash("SIP/Distance2-4b1f", "") in new stack |
| 19:10.14 | ghento | any ideas why i can't run agi? keeps saying the file isn't found |
| 19:10.25 | dlynes_ | sevard, lol |
| 19:10.28 | asterboy | trying to dial the SIP again |
| 19:10.32 | dlynes_ | sevard, anyways, afaik, there isn't a standard |
| 19:10.38 | sevard | ghento: That error message might tell you more than you think. |
| 19:10.43 | dlynes_ | sevard, but if you've got a bunch of users used to wake up calls already |
| 19:10.51 | terrapen | THE SHARIF DON'T LIKE IT....ROCK THE CASBAH, ROCK THE CASBAH! |
| 19:10.51 | asterboy | Apr 12 13:05:46 WARNING[7733]: app_flash.c:105 flash_exec: SIP/Distance2-4b1f is not a Zap channel |
| 19:10.52 | dlynes_ | sevard, why not find out from them what standard they're used to? |
| 19:10.57 | ghento | sevard, it's saying Failed to execute '/var/lib/asterisk/agi-bin/phone_callid.php': No such file or directory |
| 19:10.58 | dlynes_ | sevard, and then use that standard |
| 19:11.01 | terrapen | this is definitely going in my musiconhold |
| 19:11.13 | terrapen | maybe even the Richard Cheese version? |
| 19:11.19 | sevard | dlynes_: Sure sure. I see where you're going. Are you used to dialing a vertical serivce code for voicemail? |
| 19:11.21 | asterboy | so I need a way for flash to flash the ZAP channel NOT the SIP |
| 19:11.36 | *** join/#asterisk viperdudeuk (n=viperdud@84-45-168-61.no-dns-yet.enta.net) |
| 19:11.43 | dlynes_ | sevard, No, I'm used to dialing Feature 980 to get into voicemail |
| 19:11.49 | [TK]D-Fender | asterboy : you just need to grep out the channel. not a huge deal... |
| 19:12.00 | [TK]D-Fender | asterboy : given a few hours even I could figure it out ;) |
| 19:12.02 | dlynes_ | sevard, Or dialing my phone number from the phone that did is assigned to |
| 19:12.12 | sevard | dlynes_: is 'feature' an A B C D DTMF deal that I'm not familiar with? |
| 19:12.22 | Romik | anybody can advice about problem of 3 way transfer from the queue ? I getting following loop of error message - chan_zap.c: We're Zap/50-1, not Zap/50-2<ZOMBIE>" or "chan_zap.c: We're Zap/50-1, not SIP/???" |
| 19:12.46 | *** join/#asterisk UdontKnow (i=udontkno@freenode/staff/udontknow) |
| 19:12.57 | dlynes_ | sevard, it's a special button on nortel digital handsets |
| 19:13.20 | *** join/#asterisk pigpen2 (n=mark@207.71.48.222) |
| 19:13.21 | sevard | dlynes_: i see. |
| 19:13.23 | [TK]D-Fender | Norstar.... *shudder* |
| 19:14.04 | asterboy | I just upgraded a Vantage 12...gotta admit, they are fantastic work horses. |
| 19:14.12 | dlynes_ | asterboy, yep |
| 19:14.24 | asterboy | worked for 20+ years without a hickup. |
| 19:14.26 | dlynes_ | asterboy, norstars are pretty much unbreakable |
| 19:14.42 | asterboy | ya, I was impressed. |
| 19:14.58 | ghento | I launch it with exten => #,5,cgi,phone_callid.php|${CALLERID(num)} |
| 19:15.10 | ghento | s/cgi/agi |
| 19:15.17 | caio1982 | does anyone here knows how to convert an existing dialplan (quite complex) to realtime? maybe using some script? |
| 19:16.12 | asterboy | that flash() function seems to only work for Zap channels...not SIP answered ZAP calls. |
| 19:16.37 | [TK]D-Fender | asterboy : my first introduction to telephone systems was building a CDR / CID management console for a Bell tech who had one in his home... |
| 19:17.09 | asterboy | lol, typical telephone tech with toys in the house |
| 19:17.19 | [TK]D-Fender | yup |
| 19:17.50 | asterboy | my wife hates all my wires and equipment everywhere...resistance is futile... |
| 19:18.01 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 19:18.03 | asterboy | starting to look borgish here |
| 19:18.28 | MRH2 | hi anyone know what #define MONITOR_CONSTANT_DELAY does in channel.c |
| 19:18.40 | MooingLemur | I have a female roommate. I'm glad she's a geek and doesn't mind my wires. :P |
| 19:18.43 | [TK]D-Fender | I wrote a program that collected the CDR's trough serial ports and combined with 2 external CID modules. It also used a custom designed circuit that would pick up/hangup a line triggered through the LPT port which is how I implemented blacklists :) |
| 19:19.00 | asterboy | I'm always pulling my wire. :P |
| 19:19.19 | [TK]D-Fender | asterboy : Keep it up and that may become less figurative ;) |
| 19:19.23 | h3x0r | [tk: it would have been easier to use a modem to do that :P |
| 19:19.32 | asterboy | lol |
| 19:20.09 | [TK]D-Fender | h3x0r : Not really... he had multiple lines and limited ports... and it was COOL :) |
| 19:20.24 | [TK]D-Fender | we got to multiplex it up :) |
| 19:20.30 | h3x0r | besides modems can do cid too |
| 19:20.39 | h3x0r | use a multi serial board |
| 19:21.58 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 19:22.26 | Cybertoy | hi ... anyone running asterisk on a Soekris? |
| 19:22.35 | Cybertoy | and is the soekris fast enough to do codec transcoding? |
| 19:22.36 | asterboy | so the CIDs must have had some custom pcbs to interface the serial ports. |
| 19:22.48 | asterboy | What did you use for the database? |
| 19:23.15 | asterboy | DOS/Borland |
| 19:23.24 | asterboy | FOXpro |
| 19:23.33 | asterboy | one started with a P... |
| 19:24.13 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:24.33 | asterboy | Built a custom Gym Membership with it...started with P...can't remember |
| 19:24.46 | jaiger | paradox |
| 19:24.55 | asterboy | yep |
| 19:25.05 | viperdudeuk | hi |
| 19:26.42 | asterboy | bet he either did it in Paradox or Foxpro |
| 19:27.16 | viperdudeuk | i have a agi script that gets called when a user wants to dialout externally. it gets passed in the exten number and the number dialled and looks up in a db to see if they are allowed to dial the number. the problem is if someone forwards there phone to a external number the CALLERIDNUM is the CLID of the calling party not the extension forwarded thus the call is denied. Can anyone think of a way around this? |
| 19:29.29 | *** part/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:29.41 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:30.25 | [TK]D-Fender | asterboy : it was a wierd generic white-box.... |
| 19:30.39 | [TK]D-Fender | asterboy : for DB I did it all custom in TP7.0 |
| 19:30.56 | [av]bani | lol |
| 19:31.16 | [TK]D-Fender | asterboy : probably thinking of Paradox |
| 19:32.09 | asterboy | Pascal...well I had the P right |
| 19:32.11 | asterboy | :P |
| 19:32.38 | asterboy | Pascal is such a picky language to write in...one little ";" missing and you get stupid error messages that don't help you debug. |
| 19:33.00 | asterboy | it's the strictest I've ever programmed in. |
| 19:33.18 | asterboy | that did have some advantages for syntax regiment though. |
| 19:33.54 | triple-e | any1 had problems with a Poly501 not registering when NATing -- outbound call will work -- not registered so inbound won't work |
| 19:34.34 | stoffell_h | triple-e, dangerous combination of words: polycom and NAT... |
| 19:35.00 | asterboy | triple-e, I had that when I did not setup my register statement in sip.conf with an extension at the end. |
| 19:36.20 | [TK]D-Fender | asterboy :no, TP was great at debugging... |
| 19:36.36 | asterboy | maybe turbo was, not the version I had. |
| 19:36.45 | asterboy | vanilla pascal |
| 19:36.48 | [TK]D-Fender | asterboy : Yeah, Borlad WAS the king... |
| 19:37.04 | asterboy | Ya I miss those programming days. |
| 19:37.14 | asterboy | so much simpler |
| 19:37.44 | asterboy | now they want you to know C++, C#, ruby, perl, php, java, javascript....blah blah blah |
| 19:37.49 | asterboy | and of course cgi all that |
| 19:38.14 | asterboy | I just live in bash for all I can. |
| 19:38.52 | *** join/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 19:38.52 | jsharp | Wanted: Programmer - Must have 20 years Java experience. Pay $6.50/hr |
| 19:39.19 | asterboy | LOL |
| 19:39.31 | *** join/#asterisk jsaunders (n=root@216.86.121.58) |
| 19:39.36 | mroth_imm | Does anybody know if there is an appreciable performance benefit to disabling UDP checksums on RTP traffic? |
| 19:39.37 | asterboy | ya I have a client who thinks I get paid to much and is constantly trying to cut my rates. |
| 19:39.48 | asterboy | I told them to fuck off, I might as well flip burgers. |
| 19:39.48 | LostFrog | WTF?? $6.50?? |
| 19:40.05 | LostFrog | You make more flipping burgers. |
| 19:40.09 | asterboy | so true |
| 19:40.15 | asterboy | I kid you not. |
| 19:40.15 | LostFrog | You can start at $7.5-$8 here. |
| 19:40.44 | mroth_imm | May 23, 1995: Javatechnology launched |
| 19:40.52 | asterboy | triple-e, |
| 19:40.55 | Ciber311 | thanks for the help dlynes and asterboy :) |
| 19:41.00 | Ciber311 | have one more question |
| 19:41.04 | mroth_imm | you would need a time machine for 20 years of java experience...unless drinking coffee counts |
| 19:41.04 | asterboy | ;register => 2345:password@sip_proxy/1234 make sure your have 1234 in your extensions.conf |
| 19:41.26 | asterboy | i.e. exten => 1234,1,Dial( |
| 19:41.33 | Ciber311 | how the heck do you guys manage putting people on hold for other people to pick up at their extensions with gxp-2000's? |
| 19:41.43 | asterboy | park |
| 19:41.52 | asterboy | but I still have yet to get my park to work. |
| 19:42.06 | [TK]D-Fender | mroth_imm : Not entirely true if you consider double-counting time to account for SMP ;) |
| 19:42.08 | asterboy | It parks, but when I pickup the rtp is not transmitted. |
| 19:42.11 | De_Mon | erg.. I created a gsm of .25secons of silence but now I can't figure out a good filename |
| 19:42.22 | Ciber311 | so dialing like 700 or whatever and then telling the person over the intercom to go dial #701 or whatever to get the call? |
| 19:42.42 | De_Mon | Ciber311 you got it |
| 19:42.52 | Ciber311 | that's a big annoying hassle lol |
| 19:43.01 | De_Mon | Ciber311 well, how do you want to do it? |
| 19:43.07 | mroth_imm | [TK]D-Fender: I'll keep that in mind when I'm writing my resume |
| 19:43.09 | Ciber311 | any way to get people to juust connect to one of the zap channels |
| 19:43.11 | *** part/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 19:43.17 | Ciber311 | like grab line 2 or whatever |
| 19:43.21 | De_Mon | Ciber311 you can transfer the call? |
| 19:43.39 | Ciber311 | but who knows which of the phones the person will grab? |
| 19:43.41 | pauldy | anyone know if vontage owns broadvoice now or whats going on there |
| 19:43.52 | De_Mon | how is saying "grab line 2" different from "grab extension 701" |
| 19:44.20 | Ciber311 | would just be easier to hit one of the 4 line buttons on the phone |
| 19:44.42 | De_Mon | can't the buttons be programmed to do whatever you want? |
| 19:44.57 | *** join/#asterisk ToTo (n=ToTo@host188-67.pool8260.interbusiness.it) |
| 19:45.00 | Ciber311 | not from what i can see |
| 19:45.08 | Ciber311 | seems i can just setup accounts on them |
| 19:45.33 | triple-e | asterboy: i dont see what you were refering to |
| 19:45.41 | *** part/#asterisk Nix (n=Nix@81.213.125.220) |
| 19:46.01 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-122-123.telkomadsl.co.za) |
| 19:46.37 | Ciber311 | so basically the 4 line buttons on the gxp-2000 are useless? heh |
| 19:46.41 | *** part/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 19:46.48 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 19:46.51 | De_Mon | If the call comes in on Zap/2 wouldnt another person picking up Zap/2 join the callers together? |
| 19:47.04 | Ciber311 | makes sense |
| 19:47.20 | asterboy | triple-e, see the register statement in your sip.conf? |
| 19:47.25 | De_Mon | Ciber311 that's not what happens? |
| 19:47.27 | Ciber311 | except i don't see a way of making the line buttons actually pick up a certain zap channel |
| 19:47.34 | MRH2 | hi can anyone shed some light what this means: WARNING[6918]: channel.c:787 channel_find_locked: Avoid Initial Deadlock for'blahblah', 10 retries! |
| 19:47.49 | De_Mon | oh, you said accounts |
| 19:47.49 | dlynes_ | Ciber311, does gxp-2000 support blf? |
| 19:47.51 | jsharp | Why would you want the line buttons picking up zap channels? |
| 19:47.53 | MRH2 | is it worth a bug report? |
| 19:47.54 | Ciber311 | i figured you could map the line channels to zap channels |
| 19:47.54 | asterboy | make sure there is a suffix component that point to an extension in extensions.conf |
| 19:47.59 | Ciber311 | yes it does dlynes |
| 19:48.10 | dlynes_ | Ciber311, then you cna manage what you want to do, using blf |
| 19:48.14 | triple-e | no |
| 19:48.22 | Ciber311 | how? |
| 19:48.41 | dlynes_ | Ciber311, erm...nvm |
| 19:48.58 | Ciber311 | jsharp, so i can just hit line 4 button and pick up the call on zap channel 4 |
| 19:49.03 | dlynes_ | Ciber311, i mean does it have line appearances where you can map different line buttons to different sip accounts? |
| 19:49.08 | *** join/#asterisk MacDome (n=eseidel@A17-255-104-111.apple.com) |
| 19:49.12 | Ciber311 | yes it does |
| 19:49.25 | Ciber311 | that's what they do heh |
| 19:49.29 | MRH2 | i get it every time i do AgentMonitorOutgoing(c,n) |
| 19:49.31 | asterboy | triple-e, are you using ZAP or SIP for the lines? |
| 19:49.33 | *** join/#asterisk lyroy (n=toor@modemcable146.87-83-70.mc.videotron.ca) |
| 19:49.33 | dlynes_ | Ciber311, then map the particular sip account for line 1 so that by default it goes out on line 1 |
| 19:49.49 | triple-e | sip -- i have a register for the sip trunk |
| 19:50.00 | triple-e | but not for the individual sip extensions |
| 19:50.07 | asterboy | and you don't have a "register" statement in sip.conf? |
| 19:50.13 | De_Mon | If the call comes in on line 1 wouldnt another person picking up line 1 join the callers together? |
| 19:50.17 | Ciber311 | dlynes, and that will also pick up incoming calls on the mapped line? |
| 19:50.18 | jsharp | So your smart softswitch becomes a key system? |
| 19:50.19 | lyroy | Does someone can tell me why I receive a error message ( ...No application 'SetAccount' for extension... ) when I try to use the SetAccount command in my Dialplan? |
| 19:50.23 | triple-e | i do for the sip trunk |
| 19:50.41 | asterboy | ok, so paste that line here...less the password of course |
| 19:50.46 | stoffell_h | Ciber311, create a call queue for each user that gives you the option : 1; keep waiting untill person gets free or 2; leave a message |
| 19:50.47 | dlynes_ | Ciber311, most voip phones however only allow you to have 'line buttons', so when you pick a line button, it just gives you dial tone and it's not particular |
| 19:50.49 | *** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net) |
| 19:51.05 | dlynes_ | Ciber311, assuming it's set up correctly and it's set up the way you say it is |
| 19:51.15 | triple-e | register=econsulting:<secret>@telasip-out |
| 19:51.28 | niZon | can two seperate asterisk proccesses co-exist on the same machine and do meetme? (with different port numbers of course) |
| 19:51.59 | Ciber311 | stoffell??? |
| 19:52.10 | asterboy | triple-e, ok so now add an extension: register=econsulting:<secret>@telasip-out/<ext> |
| 19:52.24 | asterboy | <ext> can be like 1234 or whatever |
| 19:53.07 | asterboy | so in extensions.conf you have exten => 1234,1,Dial( |
| 19:53.17 | asterboy | see the relationship? |
| 19:53.28 | *** join/#asterisk Luhiwu (n=marsosa@200.63.89.187) |
| 19:53.34 | *** join/#asterisk pigpen2 (n=mark@207.71.48.222) |
| 19:53.42 | stoffell_h | Ciber311, i use a few gxp's, but I don't understand your problem I guess |
| 19:53.45 | triple-e | yeah i -- but the phone isn't showing in ship show peers |
| 19:53.58 | triple-e | so i think its on the phone |
| 19:54.22 | Luhiwu | hi all, i'm having problems using 'show channels', it doesn't give rigth information always |
| 19:54.27 | viperdudeuk | is the a way to disable or remove the CFwdALL button on the cisco 7940? |
| 19:54.28 | Ciber311 | stoffell, i want the 4 line buttons at the top to each be connected to my zap lines |
| 19:54.34 | asterboy | ya it's a good idea to have the peer working |
| 19:54.48 | Ciber311 | aka i want the line 3 button to blink etc when a call comes in on zap 3 |
| 19:55.03 | Ciber311 | and that if you hit the line 3 button you can actually pick that call up |
| 19:55.08 | stoffell_h | Ciber311, okay. simple; you can't. but if you explain what you want to do, there might be better/other ways |
| 19:55.23 | pauldy | uhm why would the 4 line buttons on the gxp-2000 be useless |
| 19:55.26 | Ciber311 | lol well that's what i want |
| 19:55.28 | pauldy | I use them all the time |
| 19:55.39 | lyroy | Does someone can tell me why I receive a error message ( ...No application 'SetAccount' for extension... ) when I try to use the SetAccount command in my Dialplan? |
| 19:55.41 | dlynes_ | stoffell_h, so the gxp-2000 doesn't actually support multiple lines then? |
| 19:55.44 | stoffell_h | Ciber311, ok. but 'wich phone' rings if zap/1 is ringing? |
| 19:56.05 | stoffell_h | dlynes_, yes it does, even diff SIP accounts, but you can't designate the button "Line1" to a ZAP channel |
| 19:56.26 | dlynes_ | stoffell_h, can you designate line 1 to username blahblah and password blehbleh? |
| 19:56.36 | Ciber311 | umm line 1 always rings on all the phones i have set for the ring group |
| 19:56.41 | dlynes_ | stoffell_h, and line 2 to usernmae blehbleh and password blahblah? |
| 19:56.44 | pauldy | yes you could you assign different sip accounts to each line |
| 19:56.47 | Ciber311 | yes dlynes |
| 19:57.00 | dlynes_ | Then you can assign line 1 to zap 1 |
| 19:57.03 | *** join/#asterisk iPBX (n=owned@68-169-204-147.agstme.adelphia.net) |
| 19:57.05 | iPBX | Hi all |
| 19:57.06 | stoffell_h | dlynes_, as pauldy says.. ack |
| 19:57.21 | Ciber311 | how dlynes? lol |
| 19:57.22 | dlynes_ | Assign that account to go out into context 'line1' |
| 19:57.30 | stoffell_h | dlynes_, hm, that's a workaround indeed |
| 19:57.33 | dlynes_ | Assign line1 incoming to ring on account 1on that phone |
| 19:57.36 | pauldy | with any amount of flexability comes a certain amount of complexity |
| 19:57.42 | dlynes_ | I do that all the time on aastra phones |
| 19:57.46 | Luhiwu | anyone uses 'show channels' usually? |
| 19:57.51 | dlynes_ | That's not a workaround |
| 19:57.57 | dlynes_ | It's a normal way of thinking :) |
| 19:58.05 | pauldy | you can also use the rln or whatever it is to assign when trunks are in use to light up th lights |
| 19:58.15 | stoffell_h | dlynes_, yeah, in some situations it is ;) but you're absolutely right! |
| 19:58.29 | dlynes_ | I come from an interconnect background, not a voip background, so I think of working with phones like i would on a digital keysystem |
| 19:58.50 | dlynes_ | Where you can have up to eight line appearances ringing in |
| 19:58.58 | stoffell_h | hehe dlynes_, seems to come in handy :) |
| 19:59.16 | dlynes_ | but my computer background is stronger |
| 19:59.36 | dlynes_ | to me it's just another computer program that i need to figure out how to pigeonhole |
| 19:59.41 | Ciber311 | so dlynes, i'm basically gonna have to add that to all 12 phones? |
| 20:00.00 | dlynes_ | Ciber311, correct, and set up hints if you want to, so you can monitor what extensions are in use |
| 20:00.12 | dlynes_ | Ciber311, assuming the gxp-2000 supports blf |
| 20:00.19 | Ciber311 | i just care about the 4 line buttons |
| 20:00.21 | Ciber311 | it does |
| 20:00.35 | Ciber311 | not enough buttons for all the extensions anyway heh |
| 20:00.41 | dlynes_ | that way you're not trying to transfer a call to an extension where someone's already talking on the phone |
| 20:00.42 | pauldy | thats what I was looking for blf |
| 20:00.55 | iPBX | i have a gxp2000 and i haven't seen anything about BLF |
| 20:00.59 | pauldy | and it does I experimented with it and some of the other phone sI have here |
| 20:01.14 | stoffell_h | iPBX, check voip-info.org, you need a beta firmware |
| 20:01.14 | pauldy | iPBX: get the newer firmware |
| 20:01.20 | Ciber311 | blf works on our gxp's |
| 20:01.24 | dlynes_ | heh |
| 20:01.33 | dlynes_ | Grandstream's following the lead of asterisk |
| 20:01.44 | dlynes_ | where the cvs version is more stable than the release version :) |
| 20:01.50 | iPBX | ooooh BLF AWESOME |
| 20:01.54 | stoffell_h | rofl |
| 20:01.55 | Ciber311 | dlynes, so i'm guessing i have to make these changes in extensions.conf? |
| 20:02.13 | dlynes_ | Ciber311, extensions.conf and set up your contexts in sip.conf |
| 20:02.37 | Ciber311 | wait |
| 20:02.42 | dlynes_ | so effectively each phone will probably have four accounts if you have four zap channels |
| 20:02.55 | Ciber311 | am i going to have to setup 4 accounts on each phone to get this to work? |
| 20:03.12 | Ciber311 | oh god |
| 20:03.18 | pauldy | dlynes everything seem more stable except the display |
| 20:03.27 | dlynes_ | what i do on my system though, is if someone tries to go out on line 1 and it's busy, they automatically try line 2, and then line 3, and then line 4 |
| 20:03.28 | pauldy | I get random pixels |
| 20:03.41 | dlynes_ | pauldy, i wouldn't know...i don't use gxp-2000 |
| 20:03.47 | De_Mon | Ciber311 buy a better phone? |
| 20:04.00 | Foxtro | how can call an extension from CLI ? |
| 20:04.03 | Foxtro | for testing |
| 20:04.05 | Ciber311 | you need to do the same crap on all of them De_Mon :P |
| 20:04.06 | stoffell_h | Ciber311, use auto provisioning.. (there's a perl script that does the trick) |
| 20:04.19 | De_Mon | Foxtro Dial(exten) |
| 20:04.20 | dlynes_ | stoffell_h, no kidding :) |
| 20:04.35 | Foxtro | *CLI> dial(100) |
| 20:04.35 | Foxtro | No such command 'dial(100)' (type 'help' for help) |
| 20:04.59 | De_Mon | Foxtro try 'dial exten' |
| 20:05.03 | Ciber311 | won't each phone need 4 accounts with each having 4 different extensions? or am i confused... |
| 20:05.31 | dlynes_ | Each phone will need four accounts |
| 20:05.36 | Foxtro | De_Mon: nothing =/ |
| 20:05.45 | dlynes_ | And you need to set up four extensions in your dialplan |
| 20:05.49 | jsharp | Each phone will need one account for each "line" button configured on it. |
| 20:05.59 | dlynes_ | Depending on whether they come in on line 1, line 2, line 3, or line 4 |
| 20:06.00 | jsharp | If you want to tie four extensions to four lines, then you need four accounts. |
| 20:06.21 | De_Mon | Foxtro no error? mine says 'exten is not available in context local' because it's not... |
| 20:06.28 | pauldy | jsharp: or you could just use the softkeys to predial an ext |
| 20:06.32 | Foxtro | De_Mon: *CLI> dia|TAB| no appers command completion |
| 20:06.37 | pauldy | that sends you out a specific zap channel |
| 20:06.40 | *** join/#asterisk zigman (i=zigman@irc.zigman.de) |
| 20:06.43 | dlynes_ | So line 1 would look like: [line1] exten => _X.,1,Dial(SIP/100_1&SIP/101_1&SIP/102_1&SIP/103_1) |
| 20:06.47 | Foxtro | *CLI> dial |
| 20:06.47 | Foxtro | No such command 'dial' (type 'help' for help) |
| 20:07.04 | jsharp | You only have dial on the command line if you're loading chan_oss or chan_alsa. |
| 20:07.17 | Foxtro | ok |
| 20:07.21 | Foxtro | how can setup ? |
| 20:07.26 | Foxtro | modules.conf ? |
| 20:07.32 | dlynes_ | correct |
| 20:07.34 | Ciber311 | dlynes so all the phones will have the same extensions? |
| 20:07.50 | De_Mon | is oss and alsa for local playback only? |
| 20:07.53 | dlynes_ | Ciber311, say on extension 100, you would have 100_1, 100_2, 100_3, 100_4 |
| 20:07.57 | Foxtro | hum... |
| 20:07.59 | Foxtro | dlynes |
| 20:08.03 | Foxtro | ; DON'T load the chan_modem.so, as they are obsolete in * 1.2 |
| 20:08.06 | Foxtro | noload => chan_modem.so |
| 20:08.12 | dlynes_ | Foxtro, ? |
| 20:08.24 | Foxtro | at modules.conf |
| 20:08.32 | dlynes_ | Foxtro, how does chan_modem.so look like chan_oss.so or chan_alsa.so? |
| 20:08.50 | De_Mon | put down the crackpipe you've had enough for today |
| 20:09.27 | Foxtro | load => chan_alsa.so |
| 20:09.28 | Foxtro | load => chan_oss.so |
| 20:09.30 | Foxtro | is this right ? |
| 20:09.36 | De_Mon | OR, not AND |
| 20:09.37 | jsharp | You can load one or the other. |
| 20:09.48 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
| 20:09.58 | jsharp | AND you have to have a sound card in the computer to use either. |
| 20:10.05 | jsharp | Now, why are you trying to Dial from the CLI? |
| 20:10.06 | *** join/#asterisk admin0 (n=root@203.91.130.212) |
| 20:10.11 | admin0 | hi guys .. |
| 20:10.15 | De_Mon | jsharp testing |
| 20:10.22 | Foxtro | bast with a reload ? |
| 20:10.26 | admin0 | exactly where do I change configs to remove ringback and get real ringing tone while calling from asterisk ? |
| 20:10.27 | Foxtro | or i stop ? |
| 20:10.33 | Foxtro | before start |
| 20:10.42 | De_Mon | can I change rings for a softphone? |
| 20:11.01 | De_Mon | in * |
| 20:11.10 | admin0 | in * |
| 20:11.36 | dlynes_ | the whole dialing from the command line seems kinda pointless to me :) |
| 20:11.46 | Foxtro | asterisk crash when load chan_alsa.so chan_oss.so |
| 20:12.01 | admin0 | there must be some file in the asterisk which deals with ringback .. and how to get the real ringing tone .. or some settings |
| 20:12.07 | dlynes_ | You load one or the other, not both, Foxtro....try to pay attention |
| 20:12.20 | Foxtro | ahh. ok |
| 20:12.24 | Foxtro | try it |
| 20:12.33 | *** join/#asterisk _AxS_ (i=admin@CPE00062595a2ad-CM00e06f1f9b10.cpe.net.cable.rogers.com) |
| 20:12.34 | De_Mon | he must be an IT manager |
| 20:12.46 | dlynes_ | lol |
| 20:12.47 | Foxtro | Asterisk Ready. |
| 20:13.12 | dlynes_ | why on earth would you want to dial from the command line, anyways? |
| 20:13.17 | Foxtro | *CLI> dial 100 |
| 20:13.17 | Foxtro | No such extension '100' in context 'local' |
| 20:13.28 | _AxS_ | hey all -- quick newbie question.. I want to make all incoming calls answerable from the controlling console, so what do i need in extensions to do this? bare minimum. |
| 20:13.32 | De_Mon | huzzah now do help dial |
| 20:13.33 | dlynes_ | Foxtro, how about Dial SIP/100? |
| 20:14.42 | Foxtro | *CLI> dial SIP/100@srv.node.cl |
| 20:14.42 | Foxtro | No such extension 'SIP/100' in context 'srv.node.cl' |
| 20:14.43 | admin0 | gurus, no answer :( |
| 20:14.48 | _AxS_ | exten => s,1,Answer() ? |
| 20:15.00 | dlynes_ | De_Mon, You change the softphone ringtones in the softphone, not asterisk |
| 20:15.05 | admin0 | so its not possible to get a real ringback in asterisk > |
| 20:15.38 | Foxtro | *CLI> dial SIP:100@srv.node.cl |
| 20:15.39 | Foxtro | No such extension 'SIP:100' in context 'srv.node.cl' |
| 20:15.40 | dlynes_ | Foxtro, do you have a context in your extensions.conf file that looks like [srv.node.cl]? |
| 20:15.47 | De_Mon | dlynes_ yeah.. know of any softphones that support ringtones based on the caller? |
| 20:15.48 | brad_mssw | wow, www.teliax.com is looking real good: |
| 20:15.52 | brad_mssw | not connecting to mysql |
| 20:16.18 | dlynes_ | De_Mon, No idea...but you can try snom360 softphone...it seems to be more advanced than some of the others |
| 20:16.25 | tzanger | I'm thinking of using one of those Citel SIP gateways for the norstar phones... anyone used 'em before? |
| 20:16.28 | Foxtro | dlynes: how ? |
| 20:16.29 | brad_mssw | and their voip service isn't doing much better |
| 20:16.29 | dlynes_ | De_Mon, You can get it at http://www.snom.de/ |
| 20:16.47 | dlynes_ | tzanger, We were going to, until we seen the price tag :) |
| 20:18.10 | asterboy | how much? |
| 20:19.14 | tzanger | yeah I saw the tag |
| 20:19.18 | *** join/#asterisk skyboy (n=skyboy@72.18.13.34) |
| 20:19.27 | tzanger | it's until I get the te405 talking to the MICS expansion modules natively :-) |
| 20:20.10 | skyboy | hello have a question about routing from openser to asterisk and the nomenclature needed..can some one help validate the routing commands? |
| 20:20.47 | dlynes_ | Foxtro, try going to http://www.voip-info.org/wiki/index.php?page=Asterisk, and do more reading on sip.conf and extensions.conf |
| 20:21.09 | dlynes_ | Foxtro, most of the questions you're asking are pretty basic; it seems like you haven't even read the most basic of help files |
| 20:21.32 | admin0 | dlynes_, any pointers for me also ? |
| 20:21.41 | dlynes_ | admin0, for? |
| 20:21.58 | dlynes_ | admin0, ringback should just happen; you shouldn't need to force it |
| 20:22.10 | admin0 | when i make a call, i get a rbt instead of the real ringing ... where do i specify |
| 20:22.19 | dlynes_ | admin0, assuming you mean feedback on the call progression |
| 20:22.28 | skyboy | if (uri=~"sip:[0-9]+@devtest.foo.org") { forward(localhost,5065"); |
| 20:22.31 | admin0 | ye s |
| 20:23.04 | dlynes_ | what's the difference between the ring back tone and the real ringing? |
| 20:23.09 | dlynes_ | aren't they the same thing? |
| 20:23.16 | skyboy | is that correct assuming the machine in question is devtest and forwards to itself for asterisk? |
| 20:23.53 | asterboy | damn, I really want visual call waiting working on my ZAP --> SIP Phone. |
| 20:24.02 | admin0 | if i ring say india, i should get tur tur and not what asterisk sends .. right now, every country/destination, we get the same tone .. . need to get the original tone .. |
| 20:24.22 | dlynes_ | admin0, That's all dependent on your terminator for the most part |
| 20:24.24 | Assid | hrmm..anyoone know any good pstn terminators.. |
| 20:24.38 | Assid | with < 2c/min |
| 20:24.39 | dlynes_ | admin0, We're calling india and we get the really crappy sounding rbt |
| 20:24.55 | dlynes_ | admin0, We're using a white route though, too |
| 20:24.57 | asterboy | Assid, try DIDx |
| 20:24.58 | *** join/#asterisk weasel00 (n=weaesl00@c-71-198-203-98.hsd1.ca.comcast.net) |
| 20:25.10 | dlynes_ | admin0, are you using a gray route, or a white route? |
| 20:25.12 | asterboy | white route is bigotry! |
| 20:25.18 | admin0 | the terminator, we are sending calls via mvts, gnugk and voipswitch .. we get the real tone .. via aterisk rbt |
| 20:25.20 | admin0 | white |
| 20:25.23 | dlynes_ | admin0, some of the gray routes use cell phones to do termination |
| 20:25.53 | dlynes_ | admin0, What's your Dial command look like? |
| 20:25.56 | admin0 | its a white one .. with gnugk, mvts and voipswitch sending real tones .. just with this asterisk, getting asterisk rbt |
| 20:26.09 | admin0 | i have * routed to that gw .. |
| 20:26.24 | admin0 | X. i meant |
| 20:26.24 | dlynes_ | gw? |
| 20:26.44 | admin0 | gw of the provider .. |
| 20:26.57 | admin0 | say 1.2.3.4 for example |
| 20:27.00 | dlynes_ | admin0, yeah, but does it look like Dial(SIP/.../...,30,r), Dial(SIP/.../...,30,R), or Dial(SIP/.../...,30)? |
| 20:27.11 | admin0 | ok .. let me get the logs |
| 20:27.19 | dlynes_ | it should be in your extensions file |
| 20:27.25 | dlynes_ | shouldn't need the logs |
| 20:28.31 | admin0 | i am not sure where I would find that :) |
| 20:28.39 | dlynes_ | /etc/asterisk/extensions.conf? |
| 20:29.13 | dlynes_ | Are you using AMP, or something? |
| 20:29.36 | Assid | didx.com ?? |
| 20:29.49 | docelm0 | AMP CUCCA! |
| 20:29.56 | dlynes_ | didx is a clearinghouse |
| 20:30.04 | Assid | oh.. |
| 20:30.11 | Assid | i need terminators.. not DID's |
| 20:30.18 | docelm0 | Assid what are you looking for? |
| 20:30.23 | admin0 | lots i extensions.conf ... used Ast@home to do the config ... |
| 20:30.26 | docelm0 | termination wise? |
| 20:30.28 | dlynes_ | It's just a third party charging everyone so much to sell their long distance and/or did service |
| 20:30.35 | dlynes_ | admin0, oh god |
| 20:30.48 | Assid | docelm0: outgoing pstn termination |
| 20:31.01 | docelm0 | Assid, destinations? |
| 20:31.17 | Assid | us/canada and MAYBE europe |
| 20:31.21 | dlynes_ | admin0, Please try #freepbx; so that they can guide you through the gui for getting everything set up properly |
| 20:31.32 | dlynes_ | admin0, unless you know something about the config files, I can't help you much |
| 20:31.38 | docelm0 | AMP IS CUCCA! Build a real asterisk system |
| 20:31.47 | LostFrog | Cucca? |
| 20:31.55 | dlynes_ | LostFrog, shiet |
| 20:31.56 | docelm0 | cucca == SHIT |
| 20:32.06 | admin0 | dlynes_ i am trying to look |
| 20:32.08 | admin0 | :) |
| 20:32.22 | LostFrog | What language is that? |
| 20:32.28 | dlynes_ | LostFrog, english |
| 20:32.45 | LostFrog | Never heard it, or seen it. |
| 20:32.48 | dlynes_ | LostFrog, only normally i think it's spelled cucka |
| 20:32.50 | LostFrog | caca, yes. |
| 20:32.54 | dlynes_ | or maybe kucka |
| 20:32.57 | dlynes_ | or something like that |
| 20:33.07 | dlynes_ | if there is a spelling for it |
| 20:33.11 | asterboy | Park Call procedure; [Transfer], Dial exten for Park(), [Transfer] |
| 20:33.15 | dlynes_ | i've never seen anyone write it or type it :) |
| 20:33.22 | asterboy | Is that what you guys are doing? |
| 20:34.06 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 20:34.08 | *** join/#asterisk jsaunders (n=root@216.86.121.58) |
| 20:34.13 | *** part/#asterisk radhios (n=radhios@bue215-194.is.net.ar) |
| 20:34.41 | admin0 | ,Dial(SIP/..../......|60|L(3120000:60000:30000)) |
| 20:35.16 | dlynes_ | admin0, Yeah...it should just work then |
| 20:35.29 | dlynes_ | Perhaps someone else is generating ring back tone along the way |
| 20:36.19 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-122-123.telkomadsl.co.za) |
| 20:36.58 | De_Mon | I want to use features.conf but whenever I press # it beeps on the phone.. But I still want to call out to systems that require pressing #.. |
| 20:37.43 | Assid | hrmm the did's at didx.org .. are they free incoming? |
| 20:37.50 | admin0 | dlyness, could it be possible that the ,Dial(SIP/..../.......|60|L(3120000:60000:30000)) be overwritten/over ruled by some other directive regarding ringing ? |
| 20:37.53 | stoffell_h | De_Mon, what phone is it? |
| 20:37.57 | asterboy | De_Mon, what shows on your CLI when you press #? |
| 20:38.13 | De_Mon | stoffell_h eyebeam (softphone) |
| 20:38.38 | De_Mon | asterboy I don't see anything |
| 20:38.48 | asterboy | turn up verbosity |
| 20:38.48 | admin0 | i meant dlynes_ :) |
| 20:38.54 | admin0 | if both are different, i am sorry |
| 20:38.55 | asterboy | iirc debug 10 |
| 20:39.03 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 20:39.31 | De_Mon | attempting native bridge |
| 20:39.42 | asterboy | debug level 10 |
| 20:39.57 | De_Mon | yep debug and verbose at 10 |
| 20:40.09 | dlynes_ | admin0, possibly...I don't know how buggy the configuration tool is for freepbx |
| 20:40.10 | De_Mon | same thing at debug 20 |
| 20:40.17 | asterboy | ok so it's not even sending. |
| 20:40.36 | asterboy | check phone config |
| 20:40.39 | De_Mon | maybe this is the wrong extension to test from |
| 20:41.11 | asterboy | there is some confusion caused by differnt digit maps with different phones |
| 20:41.25 | admin0 | dlynes_ , in normal case, if there is an issue with RBT , which directives do we look into ? just the Dial format ? |
| 20:41.28 | asterboy | my polycom won't send a * or a # |
| 20:41.46 | asterboy | If I hit [transfer] it does |
| 20:41.50 | stoffell_h | asterboy, you can change that in the local dialplan, can't ya ? |
| 20:42.22 | asterboy | yes in sip.cfg, but not sure what to put there, not tested it yet. |
| 20:42.29 | dlynes_ | admin0, make sure the dial commands are not using the ',r', or ',R' parameters |
| 20:42.31 | dlynes_ | admin0, or '|r', or '|R' |
| 20:42.37 | stoffell_h | asterboy, no in the poly phone config I mean |
| 20:43.07 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 20:43.12 | admin0 | mine is L , so was wondering if any other files/config that might be overwritting that |
| 20:43.20 | asterboy | yes sip.cfg is the poly phone config, unless you want to be phone specific in which case its <mac>-phone.cfg |
| 20:43.57 | De_Mon | when I press # it says 'starting music on hold' (because I'm transfering the call) then it starts playing the extension I sent them to |
| 20:44.58 | stoffell_h | ow, sorry, i wasn't reading correctly :) I use an empty local dialplan, works just as good. your * then has to "work it out" ... |
| 20:46.45 | asterboy | ah got ya...so you put it in extensions.conf |
| 20:47.05 | dlynes_ | admin0, I wouldn't have a clue...I don't have access to your system, and I haven't a clue how freepbx sets everything up |
| 20:47.22 | dlynes_ | admin0, I would imagine freepbx makes everything one big file |
| 20:47.39 | dlynes_ | admin0, i prefer to break everything down into smaller files so that they're easier to manage |
| 20:48.16 | admin0 | if it was not freepbx (asterisk-at-home), and if it were a normal your own asterisk system, where Dial already had L and r or R , where might you be looking at :) ? |
| 20:48.27 | stoffell_h | asterboy: "The digitmap is available on the web interface under SIP and Local Settings" -> deleting the digitmap is safe, "the phone will simply not detect any numbers intelligently" |
| 20:48.48 | dlynes_ | admin0, Well, personally |
| 20:48.55 | dlynes_ | admin0, I put the main tree into extensions.conf |
| 20:49.21 | dlynes_ | admin0, and then each customer has their own subdirectory in /etc/asterisk/extensions/, with their own configuration files |
| 20:49.27 | Assid | spurious 8259A interrupt: IRQ7. |
| 20:49.52 | admin0 | hmm.. since I am using radius, each client is automatically the default client .. |
| 20:49.59 | dlynes_ | Assid, Yeah...I've been getting that error lately after upgrading to 2.6.15.5 and zaptel trunk |
| 20:50.41 | dlynes_ | admin0, so I'd have like /etc/asterisk/extensions/247, /etc/asterisk/extensions/247/office, and so on |
| 20:51.11 | asterboy | stoffell_h, ya I just use vim |
| 20:51.13 | dlynes_ | admin0, and then all of my common configs for outbound calls are in /etc/asterisk/extensions/*.conf |
| 20:51.53 | admin0 | :) |
| 20:51.55 | dlynes_ | admin0, btw...asterisk@home is not freepbx...it uses freepbx |
| 20:52.07 | dlynes_ | admin0, freepbx is the new name of amp (asterisk management portal) |
| 20:52.25 | asterboy | Hey I have Grandstream GXP-2000 but the key presses don't work in voicemail. firmware 1.0.1.2 |
| 20:52.35 | asterboy | Anyone else have this problem or know what is going on? |
| 20:52.43 | docelm0 | dlynes_, I thought freepbx was the fork of asterisk |
| 20:52.53 | asterboy | My polycom works no problem. |
| 20:53.15 | docelm0 | ast_freak, check DTMF |
| 20:53.17 | *** join/#asterisk Ciber311 (n=Ciber@216-211-204-48.firstgate.net) |
| 20:53.23 | docelm0 | I have 80 of those phones and they all work fine organization wide.. |
| 20:53.44 | stoffell_h | asterboy ; send dtmf info -> via sip info |
| 20:53.59 | asterboy | where is that setting? |
| 20:54.12 | docelm0 | Under line1 to 4 |
| 20:54.15 | dlynes_ | docelm0, you're thinking of freepbx.org, not http://coalescentsystems.ca/index.php?page=freePBX_AMP |
| 20:54.30 | dlynes_ | docelm0, for whatever reason, both projects decided to use the same name |
| 20:54.34 | docelm0 | sigh another shitty product.. |
| 20:55.00 | generalhan | OMG .... freaking digium ! lol |
| 20:55.13 | docelm0 | generalhan, huh? |
| 20:55.30 | generalhan | i called in to find out the best thing to do with a broken production server ... i told them not to take it down ... they agreed ... then they brought my lines down TWICE |
| 20:55.35 | Nugget | http://colo.slacker.com/stuff/flightaware_hold_music.mp3 |
| 20:55.35 | generalhan | my boss is PISSED |
| 20:55.38 | stoffell_h | uh dlynes_, those are the same |
| 20:56.02 | dlynes_ | stoffell_h, oh...they've got different home pages, so I thought they were different |
| 20:56.17 | dlynes_ | stoffell_h, different domain names, too |
| 20:56.22 | De_Mon | when I enable core debugging verbose seems to go DOWN |
| 20:56.31 | asterboy | what is wrong with Digium? That is crazy. |
| 20:56.35 | [hC] | Nugget: porn music with air traffic control in the back ground? |
| 20:56.41 | Nugget | pretty much, yeah |
| 20:56.43 | stoffell_h | dlynes_, the coalescentsys.. one is the "commercial" side, by starting freepbx, they try to be more "community-minded' |
| 20:56.50 | stoffell_h | wich i think is a good thing.. |
| 20:56.53 | [hC] | I also love how air traffic control ALWAYS sounds like dudes from the south. |
| 20:57.04 | dlynes_ | ah |
| 20:57.05 | [hC] | :) |
| 20:57.32 | dlynes_ | well, anyways...freepbx.org doesn't seem to have any easy links for amp, but the coalescentsystems site does |
| 20:57.55 | stoffell_h | dlynes_, yeah :) true :) but it's improving (slowly) it seems :) |
| 20:58.29 | dlynes_ | I thought amp was just a management interface, not a complete fork of asterisk as well? |
| 20:58.52 | dlynes_ | besides...makes no sense to fork asterisk...if you don't like asterisk, why not do a complete rewrite? |
| 20:58.59 | stoffell_h | correct, it's a management interface (that replaces your dialplan) no fork |
| 20:59.18 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 20:59.27 | stoffell_h | but openpbx.org 'is' a fork of asterisk |
| 20:59.36 | dlynes_ | exactly my point |
| 20:59.38 | dlynes_ | why fork it? |
| 20:59.41 | De_Mon | openpbx.org is a complete distribution |
| 20:59.48 | asterboy | Great DTMP _>SIP fixed! yippie |
| 20:59.51 | asterboy | thanks |
| 21:00.00 | stoffell_h | dlynes_, we don't want to discuss that here :) (it's like: use debian or red hat, or whatever :p) |
| 21:00.05 | asterboy | I love it when the fix is a simple little change. |
| 21:00.21 | stoffell_h | asterboy, most fixes are, that's why they are hard to find ;) |
| 21:00.29 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 21:00.30 | asterboy | true |
| 21:01.08 | asterboy | Something tells me we are going to see a lot more Chinese logogrhams on electronics in the future. |
| 21:01.18 | asterboy | Especially on the cheap stuff. |
| 21:01.44 | stoffell_h | sure will... |
| 21:02.01 | De_Mon | nevermind i take that the distribution thing |
| 21:02.04 | Foxtro | No such extension '100' in context 'local' <-- how set this? |
| 21:02.14 | asterboy | How do I enable p0rn video on my GXP-2000 crystal display? |
| 21:02.25 | docelm0 | asterboy, I WISH! |
| 21:02.35 | asterboy | I has a great startup logo. |
| 21:02.43 | De_Mon | Foxtro how do you set the context using the dial command? |
| 21:02.45 | generalhan | Foxtro: exten => 100,1,whatever_you_want_it_to_do |
| 21:03.10 | Assid | make[2]: warning: Clock skew detected. Your build may be incomplete. |
| 21:03.13 | De_Mon | pbx*CLI> help dial |
| 21:03.14 | De_Mon | Usage: dial [extension[@context]] |
| 21:03.16 | Foxtro | [ext-local] |
| 21:03.16 | Foxtro | include => ext-local-custom |
| 21:03.17 | Foxtro | exten => 100,1,Macro(exten-vm,100,100) |
| 21:03.21 | Assid | umm whats a clock skew? |
| 21:03.32 | asterboy | Assid, I have seen that before, what OS? |
| 21:03.34 | docelm0 | Assid, means recompile |
| 21:03.50 | De_Mon | time to buy a new computer |
| 21:03.53 | Assid | debian |
| 21:03.56 | Assid | its a new one |
| 21:03.58 | Assid | amd64 |
| 21:04.05 | [TK]D-Fender | Foxtro : [ext-local] != [local] |
| 21:04.11 | dlynes_ | Assid, It means your rtc or your cmos is buggered |
| 21:04.13 | generalhan | lol |
| 21:04.19 | dlynes_ | Assid, your clock shouldn't be changing |
| 21:04.26 | generalhan | [local] = [local] |
| 21:04.38 | [TK]D-Fender | generalhan : Some people just can't griggen read... |
| 21:04.38 | dlynes_ | Assid, try setting your hwclock and your bios clock and then restarting your machine to see if that fixes it |
| 21:04.42 | [TK]D-Fender | friggen* |
| 21:04.43 | asterboy | Has your computer been burnt in yet? |
| 21:04.45 | De_Mon | dlynes_ my time drifts /skews all the time |
| 21:04.47 | dlynes_ | Assid, if it doesn't, try replacing your cmos battery |
| 21:05.00 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 21:05.02 | dlynes_ | De_Mon, Yeah, but it shouldn't |
| 21:05.07 | asterboy | New computers these days are getting real picky about burn in. |
| 21:05.09 | De_Mon | fair enough :) |
| 21:05.14 | Foxtro | [TK]D-Fender: i have that reset exten => 100 for [local] ? |
| 21:05.28 | dlynes_ | De_Mon, I've only ever had that problem with one computer, and it's because the cmos battery was dead |
| 21:05.31 | Assid | should i just rdate -sa ? and then reboot and then trey again? |
| 21:05.34 | generalhan | well this is crappy... because of digiums 2 failed SSH aatempts, my boss wont let me restart the phone lines anymore today, so im NEVER gonna get this TDM fixed |
| 21:05.35 | Assid | im not local to the box |
| 21:05.36 | [TK]D-Fender | Foxtro : what exacty are you trying to do anyways? |
| 21:05.37 | De_Mon | Foxtro just use 'dial 100@ext-local' |
| 21:05.50 | dlynes_ | Assid, hwclock --set and date --set |
| 21:06.15 | Foxtro | ahhh |
| 21:06.15 | dlynes_ | Assid, if those commands don't fix it, look at getting a new cmos battery |
| 21:06.17 | Foxtro | perfect |
| 21:06.17 | Foxtro | :D |
| 21:06.19 | Foxtro | thanks! |
| 21:06.34 | De_Mon | Foxtro learn to use 'help' and read voip-info.org |
| 21:06.48 | dlynes_ | De_Mon, I already suggested that...didn't seem to help :) |
| 21:06.49 | Foxtro | :( <-- go RTFM |
| 21:06.56 | Cybertoy | anyone have a cisco 7970 that switched to daylight savings time? |
| 21:07.05 | riddlebox | [TK]D-Fender, you got a minute? |
| 21:07.07 | generalhan | hahaha i LOVE living in Arizona |
| 21:07.13 | De_Mon | dlynes_ he's slow. Normally I would talk bad about people when they are in the room, but he doesn't read half our messages anyway |
| 21:07.17 | generalhan | stupid Fall Back Spring Forward garbage |
| 21:07.20 | De_Mon | *wouldn't |
| 21:07.43 | jsharp | You'd think that DST would have been gotten rid of with the development of the electric light. |
| 21:07.51 | generalhan | lol |
| 21:07.54 | Cybertoy | lol |
| 21:08.07 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 21:08.26 | generalhan | see it sux cause even though we dont have to set our clocks back ... we still have to pay attention cuase now we are 3 hours from the east coast instead of 2 and now Cali is on the same time |
| 21:08.39 | jsharp | But hey, who am I to mess with the Old White Protestants in power. |
| 21:09.11 | generalhan | good point ... lets just all lay back and enjoy the fruits of our voting ! pfft |
| 21:09.11 | Katty | [TK]D-Fender: mew? |
| 21:09.13 | Cybertoy | I have family in Switzerland and Brazil ... and Brazil goes to DST during our winter.. |
| 21:09.20 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 21:09.20 | Cybertoy | so it's all messed up |
| 21:09.21 | [TK]D-Fender | riddlebox : shoot |
| 21:09.23 | [TK]D-Fender | Katty: mew. |
| 21:09.30 | riddlebox | I want to write an agi script that will access someone from a mysql database, then call them what is the best way to initiate the call while someone is running the script? |
| 21:09.44 | jsharp | Don't use AGI. Use call files. |
| 21:09.51 | Katty | [TK]D-Fender: there are new pictures :> |
| 21:09.53 | De_Mon | riddlebox you could use cmd MYSQL() |
| 21:09.55 | [TK]D-Fender | riddlebox : HAVE IT CREATE A .CALL FILE |
| 21:09.56 | Katty | [TK]D-Fender: wanna see? |
| 21:09.59 | [TK]D-Fender | damn caps.. |
| 21:10.05 | [TK]D-Fender | Katty : Sure. |
| 21:10.11 | riddlebox | ok thats what I thought |
| 21:10.18 | De_Mon | tell me more about this .call file and how I works with databases |
| 21:10.28 | jsharp | Pictures of Katty causes caps in men? |
| 21:10.37 | De_Mon | oh.. nevermind that's not what I want |
| 21:11.14 | Assid | well i rdated the ntp .. and then ran hwclock --systohc |
| 21:11.17 | Assid | didnt help |
| 21:11.20 | De_Mon | is the MYSQL() command available in odbc or as a postgres command? |
| 21:11.21 | Assid | still giving me problems |
| 21:12.00 | riddlebox | I do need to find more info out on the .call file |
| 21:12.01 | Katty | jsharp: weirdo. |
| 21:12.04 | De_Mon | or should I be doing database queries from aig instead of the dialplan? |
| 21:12.07 | *** join/#asterisk salviadud (n=ralfalfa@201.137.164.110) |
| 21:12.36 | De_Mon | riddlebox http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 21:12.53 | riddlebox | De_Mon, I just want to do my queries from an agi script |
| 21:13.28 | De_Mon | riddlebox I'd rather stay away from agi weridness if at all possible :o |
| 21:13.40 | riddlebox | De_Mon, whys that? |
| 21:14.13 | De_Mon | riddlebox I keep reading about horrible problems with people using agi |
| 21:14.18 | Assid | well.. i gotta wait for someone to change the cmos battery now |
| 21:14.22 | De_Mon | s/with/from/ |
| 21:14.56 | De_Mon | mostly slowdowns and cpu usage |
| 21:16.06 | riddlebox | De_Mon, ahh, well there are only two of us that use this system, so I am not to worried about it |
| 21:16.19 | Foxtro | i can setup a simple pci modem for asterisk ? |
| 21:16.24 | riddlebox | plus it is on a 2.4 ghz machine with 512 ram |
| 21:16.46 | [TK]D-Fender | Foxtro : only a very specific series. Go read up on the X100... though it sorta sucks... |
| 21:17.21 | De_Mon | sorta? it works great for around 14 days! |
| 21:17.27 | pfn | hmm, how well does asterisk run in virtual environments (e.g. in one of those virtual dedicated server hosting environments?) |
| 21:17.37 | pfn | I imagine that it should work very well |
| 21:17.44 | mtaht3 | timing, timing, timing |
| 21:17.59 | De_Mon | pfn horribly |
| 21:18.31 | Foxtro | [TK]D-Fender exten => 200,1,Dial(Modem/ttyI0/1234567:${EXTEN}) ? |
| 21:18.47 | De_Mon | pfn direct access is better than going through a virtulization layer to get to timers |
| 21:19.01 | [TK]D-Fender | Foxtro : uhh... where did you see a reference to a tech setup like that? |
| 21:19.25 | Foxtro | http://www.voip-info.org/wiki/view/Asterisk+Modem+channels |
| 21:19.33 | triple-e | anyone in stlouis |
| 21:19.58 | *** join/#asterisk yellowline (n=yellowli@p54BA28B5.dip0.t-ipconnect.de) |
| 21:20.00 | tainted- | [TK]D-Fender i caught you a delicious bass |
| 21:20.15 | *** join/#asterisk FlyboySR22 (n=rsears@sdtc.ar01.f2-40.host2.1.americanis.net) |
| 21:22.24 | Foxtro | modem.conf: Configure Modem channels (for ISDN, not for modems!) |
| 21:22.25 | Foxtro | :( |
| 21:22.32 | pfn | so is asterisk's timing that poor in a virtual box? |
| 21:22.50 | pfn | what about colinux... isn't that considered virtual and "usable"? |
| 21:23.14 | Foxtro | pfn: virtual? as jailed virtual server? |
| 21:23.27 | Foxtro | or at shell on webhosting server per example |
| 21:23.32 | [TK]D-Fender | tainted- : Sea bass at least? And all I wanted were sharks with friggen lasers on their heads! |
| 21:23.39 | asterboy | yippie...received my damnsmalllinux CD! |
| 21:23.57 | Foxtro | [TK]D-Fender: where i can read for work with simple modem ? |
| 21:24.39 | asterboy | must be hurtin for funds...they just used a fugly marker on the cd...too funny. |
| 21:24.44 | [TK]D-Fender | Foxtro : You DON'T. Asterisk can not work with just any junk modem you can get your hands on? I though I made that clear. only the Intel 537 shipset series were compatible. |
| 21:24.58 | [TK]D-Fender | asterboy : Why not just burn it yourself? |
| 21:25.05 | pfn | foxtro virtual as in jailed or vmware/xen type setup |
| 21:25.06 | pfn | either |
| 21:25.06 | asterboy | support the project of course |
| 21:25.15 | Foxtro | :o i have a intel modem... :....) |
| 21:25.22 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
| 21:25.29 | _AxS_ | Foxtro: INTEL 537? |
| 21:25.58 | asterboy | If I made tons of cash, (the kind Bush gets to spend on behalf of the public), I'd be pumping every Linux project with cash. |
| 21:26.02 | pfn | I'm just curious if the $30/mo vmware/xen images you can get nowadays are sufficient for running asterisk |
| 21:26.29 | pfn | and I imagine if one can run asterisk suitably within colinux, the same should be true of vmware/xen |
| 21:26.30 | _AxS_ | pfn: if asterisk can runn off of a linksys router i think it can run off of those.. |
| 21:26.51 | pfn | axs the difference is in direct access to hardware for timing, etc. vs. virtualized access |
| 21:27.13 | _AxS_ | ah.. ok well vmware iirc is good for that, but i don't know anything about xen |
| 21:27.15 | triple-e | If the phones fail to register with Asterisk but can still make outbound calls, you likely need to adjust the digest realm parameter from the default of PolycomSPIP. If this does not solve the problem, please visit |
| 21:27.30 | triple-e | how do you change the digest realm parameter |
| 21:27.33 | pfn | xen is waycool, but vmware is sufficient, xen is more than sufficient |
| 21:28.00 | pfn | xen is effectively an "opensource" vmware... except cooler ;-) |
| 21:28.03 | _AxS_ | does xen have the same kernel-level hooks that vmware does? |
| 21:28.28 | jaiger | _AxS_, xen needs kernel hooks for linux to run |
| 21:28.30 | tzafrir_laptop | no. Different ones |
| 21:28.31 | pfn | xen has the hardware hooks on new p4 hardware |
| 21:28.43 | _AxS_ | yep, shoudlw rok then |
| 21:28.44 | pfn | jaiger xen guests can run unmodified |
| 21:28.47 | _AxS_ | err, should work |
| 21:28.51 | pfn | in xen 3.0 |
| 21:28.59 | jaiger | pfn, even on old hardware?\ |
| 21:29.10 | pfn | no, not on old hardware |
| 21:29.17 | pfn | but for my purposes, I don't care about old hardware |
| 21:30.14 | [TK]D-Fender | Yay... my iaito is due to arrive tomorrow.. |
| 21:30.19 | pfn | I guess I'll have to try out asterisk on one of my virtual boxes and see if it works well |
| 21:31.12 | pfn | a sword? |
| 21:32.52 | *** join/#asterisk Utah_Dave (n=boucha@12.118.109.86) |
| 21:35.02 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 21:35.35 | ctooley | Where did the Polycom firmware updates go. You stop doing something for a year or so and everything changes. I can't find them anywhere anymore. |
| 21:36.02 | *** join/#asterisk Itto (n=x@host-200-94-104-10.block.alestra.net.mx) |
| 21:36.14 | [TK]D-Fender | pfn : yup. |
| 21:36.37 | Foxtro | _AxS_: yes. i corrobored. my modem is a intel 537 AMI-IA92/IE92 |
| 21:36.39 | Foxtro | :d |
| 21:36.41 | Foxtro | :D |
| 21:38.13 | _AxS_ | Foxtro: cool. my modem's an intel, but unfortunately its not a 537.. no luv for me.. :( |
| 21:38.18 | *** join/#asterisk eauxnguyen (n=eauxnguy@oh-71-53-63-201.dhcp.sprint-hsd.net) |
| 21:38.24 | Foxtro | jojoojoj |
| 21:38.25 | Foxtro | :D |
| 21:38.32 | Foxtro | lucky lucky lucky |
| 21:38.40 | lokkju | http://rafb.net/paste/results/YCtn6M40.html - full log shows answer, then wait, then playing beep, then nothing untill I hangup - hangin on the Playback, obviously, but *why* |
| 21:40.10 | [TK]D-Fender | lokkju : pastebin the CLI of a call |
| 21:40.25 | Foxtro | _AxS_ now problem is how fuck have work with asterisk :( |
| 21:40.35 | [TK]D-Fender | pfn : What I've got coming in : http://www.blades-uk.com/large_pic.php?product_id=576 |
| 21:40.35 | lokkju | [TK]D-Fender, k, hold on a sec |
| 21:41.04 | lokkju | [TK]D-Fender, wait, CLI, or full? don't you mean full? |
| 21:41.56 | dlynes_ | Is there any way to get these annoying messages to stop showing up on the console:? |
| 21:41.58 | dlynes_ | Apr 12 14:40:57 NOTICE[11901]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 21:42.08 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 21:42.15 | dlynes_ | You'd think with verbosity set to 0, notices wouldn't show up |
| 21:42.43 | tzanger | you'd think a lot of things :-) |
| 21:43.07 | asterboy | Anyone with a Dlink DVG-1402S |
| 21:43.08 | dlynes_ | at verbosity 0, only errors should be showing up |
| 21:43.17 | dlynes_ | but that would follow logic |
| 21:43.53 | [TK]D-Fender | lokkju : I want to se the full CLI output of a call from beginning to end... I presume you are trying to dial after then 2nd beep.... |
| 21:44.09 | Romik | anybody can advice about problem of 3 way transfer from the queue ? I getting following loop of error message - chan_zap.c: We're Zap/50-1, not Zap/50-2<ZOMBIE>" or "chan_zap.c: We're Zap/50-1, not SIP/???" |
| 21:46.06 | lokkju | [TK]D-Fender, http://rafb.net/paste/results/FyIZpP55.html |
| 21:46.18 | lokkju | [TK]D-Fender, trying to get it to play the beeps at all |
| 21:46.52 | lokkju | [TK]D-Fender, I sorta hear the first beep, like it is really cut off, but then nothing, as you can see in the nopaste |
| 21:47.04 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 21:47.05 | lokkju | and I wait for almost 30 seconds before I hangup |
| 21:47.18 | *** join/#asterisk heath__ (n=heath__@12-215-32-56.client.mchsi.com) |
| 21:48.04 | heath__ | Anyone know anything about "out of iax2 threads for scheduling" ... neither bugs nor google has anything |
| 21:48.43 | tzanger | heath__: it's the new iax2 threading code... too new for both |
| 21:48.57 | tzanger | basically look at the configs/iax.conf.sample and set the iaxthreadcount higher |
| 21:49.33 | heath__ | ahh, awesome |
| 21:49.39 | heath__ | thanks |
| 21:50.50 | [TK]D-Fender | lokkju : lok at the first line of that incoming call.. its a *GOTO*. I think you "included" context has something in there you really don't want and it overrode what you THOUGHT it would do... |
| 21:51.10 | lokkju | hmm |
| 21:51.21 | lokkju | no |
| 21:51.31 | lokkju | I told it to goto ext-did |
| 21:51.35 | lokkju | which is where it wnt |
| 21:51.41 | [TK]D-Fender | lokkju : See from the code you pasted the first line executed should have been a "Set" but it clearly WASN'T |
| 21:51.52 | [TK]D-Fender | exten => s,1,Set(FROM_DID=s) |
| 21:52.04 | [TK]D-Fender | See that? Looks like line 1 should be a SET <- |
| 21:52.30 | [TK]D-Fender | presuming of course the call comes in under it. |
| 21:53.03 | [TK]D-Fender | Wait a sec.. |
| 21:53.04 | lokkju | no, look again |
| 21:53.08 | [TK]D-Fender | looks a little messed up... |
| 21:53.11 | lokkju | the first line after the GOTO should be a set |
| 21:53.12 | [TK]D-Fender | hmm |
| 21:53.12 | lokkju | and it is |
| 21:53.56 | [TK]D-Fender | So it goto's within context and then you should hear audio but only catcha little of the first beep, and no second beep at all? |
| 21:54.04 | lokkju | correct |
| 21:54.17 | lokkju | and no Wait() ever gets executed wither, as you can see in the paste |
| 21:54.34 | lokkju | as soon as it does the playback, it stops |
| 21:54.40 | lokkju | untill I hang up |
| 21:54.55 | lokkju | (no Wait() after the beep, that is) |
| 21:55.53 | [TK]D-Fender | lokkju : you sure that the running version is the same? (not needing a "reload") |
| 21:56.01 | lokkju | yes |
| 21:56.08 | lokkju | very, very, very, very, very sure |
| 21:56.18 | terrapen | Apr 12 13:17:08 WARNING[18562]: db.c:67 dbinit: Unable to open Asterisk database |
| 21:56.22 | terrapen | wot the hell! |
| 21:56.39 | terrapen | hrmm i wonder what file it is trying to open |
| 21:58.57 | lokkju | [TK]D-Fender, see what I mean, about it behaving very oddly? |
| 22:00.08 | [TK]D-Fender | lokkju : Yeah, that just doesn't add up... period... |
| 22:00.17 | [TK]D-Fender | do a "show dialplan" just to be sure |
| 22:01.35 | *** join/#asterisk Liquid_Ic (n=Liquid_I@ool-4573cc11.dyn.optonline.net) |
| 22:02.44 | *** join/#asterisk RoyK (n=roy@ti211310a080-1734.bb.online.no) |
| 22:02.45 | lokkju | [TK]D-Fender, if you insist |
| 22:03.05 | lokkju | you want the whole thing, or just ext-did (since we can see that it is using ext-did) |
| 22:03.43 | ljam | [TK]D-Fender: you! |
| 22:03.49 | lokkju | http://rafb.net/paste/results/nF6b7466.html |
| 22:05.02 | *** join/#asterisk timscott (n=a@d198-166-221-177.abhsia.telus.net) |
| 22:05.46 | [TK]D-Fender | lokkju : Just look at it yourself, I'm getting the impression you'll notice if something si out of place... |
| 22:05.51 | [TK]D-Fender | ljam : I! |
| 22:06.07 | ljam | [TK]D-Fender: I don't want to know your name, I just want, !, !, ! |
| 22:06.48 | lokkju | [TK]D-Fender, Yeah, I would, and I've looked and looked, without finding anything... so I am fresh out of ideas |
| 22:06.57 | lokkju | but I don't really want to reinstall |
| 22:07.27 | [TK]D-Fender | lokkju : Ummm.. looks like an infinite loop should be formed, not a hangup |
| 22:07.35 | *** part/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 22:07.37 | [TK]D-Fender | #! ;) |
| 22:09.52 | lokkju | I have to initiate the hangup, form my end, otherwise, it does "hang" (hang as in do nothing) |
| 22:09.59 | lokkju | what would cause an infinate loop? |
| 22:11.12 | [TK]D-Fender | lokkju : Sorry, just misread the target context... my bad. |
| 22:11.23 | [TK]D-Fender | I'm just a little too sloppy today... |
| 22:12.29 | lokkju | np |
| 22:19.59 | *** join/#asterisk cicadia (n=ian@S01060013463efeeb.vc.shawcable.net) |
| 22:21.31 | *** join/#asterisk MacDeath (n=davidn@196.202.248.34) |
| 22:21.49 | MacDeath | morning / evening all |
| 22:22.26 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 22:23.03 | MacDeath | i used to have my * machine with both a diginum and hfc cards working |
| 22:23.06 | DoktorGreg | hey all |
| 22:23.07 | MacDeath | the hdd crashed today |
| 22:23.13 | MacDeath | and i had to reinstall |
| 22:23.22 | DoktorGreg | pri question... last think that is not crystal clear to me |
| 22:23.23 | MacDeath | for the life of me, i cant get both cards to work |
| 22:23.33 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 22:23.44 | MacDeath | when i try i get Ouch ... error while writing audio data: : Broken pipe |
| 22:23.58 | DoktorGreg | how does asterisk know what number the outside caller dialed to get into my pri trunk? |
| 22:24.05 | MacDeath | i have looked high and low and see a few people have had the same problem |
| 22:24.11 | |omni| | Dok: the extension |
| 22:24.21 | MacDeath | DoktorGreg : the incoming extention |
| 22:25.18 | MacDeath | DoktorGreg : if you need to display that instead of your cid, you can set a variable to $exten in your extension.conf |
| 22:25.33 | MacDeath | and then set the cid to that variable |
| 22:26.02 | DoktorGreg | ok, so im crystal clear before i go messing with the pri line... |
| 22:26.16 | DoktorGreg | the channel number on pri line means nothing |
| 22:26.27 | MacDeath | nope |
| 22:26.39 | DoktorGreg | the extension contains the number |
| 22:26.54 | DoktorGreg | OIC! |
| 22:27.30 | MacDeath | on a normal zaptel / hfc this would normally be set as "" or s |
| 22:27.44 | MacDeath | on a pri it is the dialled numbe |
| 22:28.16 | DoktorGreg | is there an example extensions.conf around for this? |
| 22:29.29 | *** join/#asterisk UncleKaos (n=spitalfi@CPE00045ad7df79-CM00137188ab96.cpe.net.cable.rogers.com) |
| 22:29.30 | DoktorGreg | oic |
| 22:29.38 | DoktorGreg | Im gonna have extensions call |
| 22:29.50 | DoktorGreg | 18885551212 |
| 22:30.06 | DoktorGreg | so in my incoming context |
| 22:30.18 | DoktorGreg | Ill have alone along the lines of |
| 22:30.51 | DoktorGreg | exten = 18886128242,1,macro(route-the-call,1,1) |
| 22:31.09 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 22:31.25 | |omni| | ya |
| 22:31.40 | DoktorGreg | ok, makes perfect sense now |
| 22:32.52 | DoktorGreg | ok, then on the norstar side |
| 22:33.05 | DoktorGreg | i just make sure I only add DID's |
| 22:33.28 | DoktorGreg | then the phones on the norstar just map to entensions on asterisk |
| 22:33.51 | DoktorGreg | hahahaha I think i Grok it! |
| 22:34.51 | DoktorGreg | just in time to, tomorrow evening is x-day! |
| 22:35.14 | DoktorGreg | wait! |
| 22:35.32 | DoktorGreg | so in theory, I dont really have to change anything right away on norstar |
| 22:35.46 | DoktorGreg | I could just pass call through to it |
| 22:36.03 | DoktorGreg | so my incoming from pri context would be |
| 22:36.38 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-135-44.gdrpmi.dsl-w.verizon.net) |
| 22:36.56 | DoktorGreg | exten => 18886128242,1,dial(correct pri line channels,18886128242) |
| 22:37.15 | DoktorGreg | syntax corrected |
| 22:37.59 | DoktorGreg | and the same thing only going out for calls from the pri line |
| 22:38.06 | DoktorGreg | er norstar to pri |
| 22:38.56 | DoktorGreg | then i can refactor at will |
| 22:39.29 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
| 22:39.46 | DoktorGreg | |omni|, am i groking this correctly? |
| 22:42.04 | |omni| | yup |
| 22:42.30 | |omni| | so just create different extensions for each of your DIDs |
| 22:42.36 | |omni| | or for several to match |
| 22:42.42 | |omni| | each one that you need to do something different with anyway |
| 22:42.49 | *** join/#asterisk RoyK (n=roy@28.80-203-106.nextgentel.com) |
| 22:43.11 | DoktorGreg | well i like the idea of not having to mess around with **CONFIG yet A LOT |
| 22:44.00 | DoktorGreg | that lets me break it down into several steps rather than have a huge one night hussle |
| 22:45.07 | DoktorGreg | one very last question and i am ready to start breaking thigns |
| 22:45.25 | DoktorGreg | if i pass a fax line through that way |
| 22:45.32 | DoktorGreg | will the faxes transmit ok? |
| 22:45.56 | DoktorGreg | I was reading that there is some issue with digium cards and faxes |
| 22:46.23 | *** join/#asterisk Strom_M (n=strom@gateway.digium.com) |
| 22:46.34 | Az_au | i think the problem is more with using codecs via ata's? |
| 22:47.35 | DoktorGreg | kk |
| 22:47.44 | DoktorGreg | oh one very last question off topic |
| 22:48.06 | DoktorGreg | how is it that people demoing cad software make that stuff look a lot easier than it really is? |
| 22:48.35 | Qwell[] | DoktorGreg: because it is easy |
| 22:49.09 | MacDeath | everything is easy when you know how :P |
| 22:49.19 | timscott | : |
| 22:49.19 | timscott | ) |
| 22:51.08 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
| 22:53.24 | *** join/#asterisk frenzy (n=frenzy@196.45.144.41) |
| 22:54.38 | frenzy | any one recommend an LCR Trunking system/software for asteirsk? |
| 22:56.06 | Cybertoy | If you search for LCR asterisk on google you will find one ... I never tried it though |
| 22:58.11 | frenzy | I'm looking for more of a LCR Failover system |
| 22:58.36 | frenzy | if a trunk fails it fails over the the next least route |
| 22:58.44 | frenzy | to the ** |
| 22:59.17 | *** join/#asterisk Nodren (n=nodren@64.193.95.10) |
| 23:00.13 | *** join/#asterisk QbY (i=user@cm-12-146-225-110.dhcp.geo-sc.southerncoastalcable.net) |
| 23:00.32 | QbY | I just got my timer working propertly.. But now I get this: Apr 12 18:59:24 WARNING[7610]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (incoming, 7005420272, 5) |
| 23:00.43 | QbY | Do I have to rebuild Asterisk? |
| 23:00.52 | DoktorGreg | how come fractional PRI lines cost more that full PRI lines? |
| 23:01.54 | Nodren | i have a question, i'm setting up a dialplan, which uses sip phones and zaptel lines to receive/make calls... anyway, my Dial() function is like Dial(Zap/1/number&Zap/2/number&Zap/3/number,30,Ttr) it seems to work fine the first time, anytime after that i get a message saying the call couldnt be completed as dialed, any suggestions? |
| 23:03.06 | Strom_M | Nodren, you're dialing the same number on multiple phone lines?? |
| 23:03.28 | Nodren | i read in the oreilly book that the first line to take the call |
| 23:03.31 | Nodren | causes the others to hang up |
| 23:03.43 | Nodren | and asterisk seems to do just that when i'm watching it in the console |
| 23:03.51 | Strom_M | Nodren, um. use hunt groups :) |
| 23:04.15 | Nodren | no idea what those are, but i'll look into that, thanks |
| 23:04.17 | Strom_M | put zap channels 1, 2, 3, whatever into a group in the zapata.conf |
| 23:04.29 | Strom_M | then you just need to dial Zap/G0/number |
| 23:04.50 | Nodren | ohh! |
| 23:04.51 | Nodren | awesome |
| 23:04.53 | Nodren | thanks. |
| 23:05.18 | Nodren | i bet thats already done... i started with asterisk@home, but my dialplan is really messed, so i'm rewriting it from scratch |
| 23:05.29 | Strom_M | yech, not a@h |
| 23:06.31 | Nodren | it got the job done quick.. even though there was tons of errors |
| 23:06.42 | Nodren | like, calls hanging up randomly.. people being transfered are somehow lost |
| 23:07.03 | Nodren | lots of dumb stuff, which is why i'm setting it up from scratch |
| 23:07.27 | DoktorGreg | refactor, dont rewrite! |
| 23:08.01 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 23:08.26 | QbY | chan_zap.c:9131: error: too few arguments to function `pri_new' ??? any suggestions |
| 23:08.30 | DoktorGreg | simple programming rule to make you a better program in one step |
| 23:08.40 | DoktorGreg | never re-write anything |
| 23:08.47 | DoktorGreg | never start a program from scratch |
| 23:08.54 | DoktorGreg | find existing code and use it |
| 23:09.21 | DoktorGreg | no matter how fugly the existing code is |
| 23:09.23 | Az_au | umm.... i wouldn't agree |
| 23:09.47 | [hC] | DoktorGreg: do you work for microsoft, by chance? :) |
| 23:09.48 | Az_au | you remember things better if you write it yourself |
| 23:10.13 | DoktorGreg | lol, saying I work for mocrosoft would indeed be a fair criticism |
| 23:10.20 | DoktorGreg | while i dont directly work for microsoft |
| 23:10.26 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 23:10.27 | Nodren | ok using the group |
| 23:10.39 | Nodren | still getting a mesage from the phone company saying "were sorry the call did not go through, try again" |
| 23:10.42 | franck | hi, what codecs IAX can use? |
| 23:10.50 | Qwell[] | franck: any |
| 23:10.52 | [hC] | franck: all of em |
| 23:10.56 | Qwell[] | [hC]: well? |
| 23:11.00 | *** join/#asterisk dflow (i=pch@yennefer.sisco.pl) |
| 23:11.06 | DoktorGreg | Az_au, you can remember every detail of your code when your program gets into the 10k line range? |
| 23:11.15 | Az_au | yes |
| 23:11.18 | DoktorGreg | I wish i had memory like that |
| 23:11.20 | [hC] | Qwell[]: astlinux got upgraded last night at home to trunk, so i'll point my phone at it tonight when i get there |
| 23:11.22 | franck | ok... must have an error in my config |
| 23:11.25 | Qwell[] | cool |
| 23:11.25 | [hC] | Qwell[]: you gonna be online tonight? |
| 23:11.35 | Qwell[] | [hC]: yeah, I'll be playing with my new server :D |
| 23:11.39 | [hC] | sunfire? :) |
| 23:11.41 | DoktorGreg | I start losing bits around 3k lines of code |
| 23:11.45 | Qwell[] | indeed |
| 23:11.48 | [hC] | Cool |
| 23:11.55 | [hC] | I havent played with one of those in a while |
| 23:11.58 | franck | if I specify say iLBC to talk to a voip provider but then the local phone is ulaw to asterisk, is ilbc still in use? |
| 23:12.07 | Qwell[] | franck: yes |
| 23:12.10 | Strom_M | franck, yes, asterisk transcodes |
| 23:12.40 | DoktorGreg | I knew a guy who could grok 100k lines programs |
| 23:12.44 | franck | ok cool... |
| 23:12.59 | DoktorGreg | he didnt have a girlfriend though:) |
| 23:13.10 | franck | they slow you down girlfriends... |
| 23:14.09 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 23:14.19 | *** join/#asterisk mgob (n=goldenol@w110.z064003070.lax-ca.dsl.cnc.net) |
| 23:14.25 | mgob | hi |
| 23:14.29 | mgob | any clue why I would get "Ring requested on channel 0/23 already in use on span 1 |
| 23:14.31 | mgob | " |
| 23:14.41 | mgob | when normally the calls cycle through the lines |
| 23:14.47 | DoktorGreg | oh man now that i understand this i am officially itching to install the new digium card |
| 23:14.55 | mgob | but every so often, usually after the PBX has been running for awhile, this will occur |
| 23:14.57 | *** join/#asterisk Rawplayer (i=kevin@ipc31055d2.oom-killer.org) |
| 23:15.00 | Rawplayer | re |
| 23:15.02 | DoktorGreg | but the server is 75 miles away |
| 23:15.08 | Strom_M | DoktorGreg, which digium card didya get? |
| 23:15.13 | DoktorGreg | 405 |
| 23:15.16 | Qwell[] | Strom_M: x100p! |
| 23:15.18 | Qwell[] | :p |
| 23:15.21 | Strom_M | ooh, that's a yummy one ;) |
| 23:15.27 | Strom_M | Qwell, hahaha |
| 23:15.58 | Qwell[] | 75 miles is a walk in the park |
| 23:16.09 | Qwell[] | That'd only take about 30 minutes to get there...3 hours with traffic |
| 23:16.27 | LostFrog | You just need a really long screwdriver. |
| 23:16.32 | *** join/#asterisk pc2 (n=pc@hlydsl1720.marketron.com) |
| 23:16.32 | Az_au | lol |
| 23:16.38 | pc2 | My DISA stopped working. I dial the access number, hit the extension I assigned to it, I get a dial tone but everything I dial does nothing |
| 23:16.46 | pc2 | It eventually just says exited non-zero and times out. |
| 23:16.49 | pc2 | any ideas? |
| 23:16.57 | Qwell[] | pc2: Is it pointing to a valid context still? |
| 23:16.59 | Strom_M | pc2, who is your voip carrier? |
| 23:17.06 | *** part/#asterisk dflow (i=pch@yennefer.sisco.pl) |
| 23:17.07 | Az_au | anyone here familiar with the asterisk management api? |
| 23:17.08 | generalhan | Qwell[]: remember my issue with the TE210 and TDM not working properlly together ? |
| 23:17.11 | pc2 | Strom_M - inbound, sunsaturn, outbound, nufone |
| 23:17.14 | Qwell[] | generalhan: no, but ok |
| 23:17.18 | pc2 | Qwell - I didnt' change anything. I can check. |
| 23:17.25 | pc2 | Qwell - It just, stopped working =) |
| 23:17.32 | mgob | anyone seen a resolution for this... http://www.voip-info.org/tiki-print.php?page=Ring+requested+on+channel ??? |
| 23:17.55 | generalhan | Qwell[]: well i talked with digium and told them that nothing can be stopped or restarted cause it was a production server and they agreed, so i gave them root access and they shut my lines down TWICE ! lol |
| 23:18.03 | generalhan | freaking people .... my boss is all sorts of pissed at me right now ! |
| 23:18.06 | pc2 | Qwell - I can dial out fine. |
| 23:18.15 | pc2 | Qwell - the line is just exten => 8500,1,DISA(no-password|default) |
| 23:18.21 | Qwell[] | EWW! |
| 23:18.40 | Qwell[] | default = BAD |
| 23:18.41 | generalhan | lol |
| 23:18.47 | pc2 | :P |
| 23:18.58 | pc2 | Qwell - It's ok, there's lik $5 on the nufone account =) |
| 23:19.19 | Strom_M | wow, ive been doing DIY DISA all this time - didnt know about the DISA app ;) |
| 23:19.45 | UncleKaos | anyone know how to change the default festival voice to another one of the festival voices? i can't find it in the festival config anywhere |
| 23:20.04 | pc2 | Qwell - Any ideas? :P |
| 23:20.23 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- Asterisk 1.2.7 Released! (April 7, 2006) -=- http://www.asterisk.org/ -=- AMP/FreePBX/Asterisk@Home users should join #freepbx for support |
| 23:20.43 | Qwell[] | April 7th? |
| 23:20.49 | Qwell[] | damn, that's a late announcement |
| 23:20.57 | [hC] | holy crap its the past! |
| 23:21.10 | Qwell[] | russellb: :p |
| 23:21.10 | frenzy | more like the past |
| 23:21.13 | frenzy | in the future |
| 23:21.25 | pc2 | Anyone? Disa functionality help please? :) |
| 23:21.38 | lokkju | hmm |
| 23:21.41 | russellb | d'oh |
| 23:21.47 | Strom_M | pc2, replace DISA with a menu or something and see if the touchtones are even reaching the box |
| 23:21.54 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- Asterisk 1.2.7 Released! (April 12, 2006) -=- http://www.asterisk.org/ -=- AMP/FreePBX/Asterisk@Home users should join #freepbx for support |
| 23:21.59 | pc2 | Strom_M - they are, because I have to dial extesnoin 8500 to get disa |
| 23:22.13 | lokkju | with DISA, can it specify a toally different context then normal? I am thinking it would be an interesting way to do a per-user auth conference calling system |
| 23:22.16 | DoktorGreg | oh where oh were did i go wrong, my sone wears black turtle necks.... |
| 23:22.46 | Qwell[] | DoktorGreg: About 15? |
| 23:23.02 | DoktorGreg | lol, about 5 |
| 23:23.06 | Qwell[] | oh boy |
| 23:23.32 | LostFrog | Am I missing something with respect to black turtle necks? |
| 23:23.36 | DoktorGreg | maybe he goes through the whole emo thing before he is teenager? |
| 23:23.42 | Strom_M | what's wrong with black turtle necks? |
| 23:23.57 | DoktorGreg | I want him to be happy, not emo kid |
| 23:24.08 | Qwell[] | DoktorGreg: For the record...somebody picked it out for him :p |
| 23:24.09 | LostFrog | Other than the fact that AV geeks and Drama grips wear them? |
| 23:24.32 | Nodren | heres a question... i got the zap group configured, i'm setting up the dialplan so you simply dial on the phone, and it rings out to the Zap group. this works fine for the first call, i can even watch the console and see the number i dialed. but every other call after that fails, unless i add a 9 infront of it.. any ideas why? |
| 23:24.59 | Strom_M | Nodren, pastebin your extensions.conf |
| 23:25.03 | Strom_M | ~pastebin |
| 23:25.09 | jbot | i guess pastebin 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 |
| 23:25.35 | Nodren | http://pastebin.com/656731 |
| 23:26.09 | Nodren | the [from-internal] group is where it starts |
| 23:26.30 | Strom_M | yeah, im looking.... |
| 23:26.49 | Strom_M | you know, you really should use more specific pattern matching than _X. |
| 23:27.13 | frenzy | dont see changelog for * 1.2.7 |
| 23:27.24 | Qwell[] | frenzy: Give it a minute, geez :p |
| 23:28.10 | frenzy | Qwell[]: hmm so they make the cake, put out and then add the icing ? |
| 23:28.16 | frenzy | bake* |
| 23:28.19 | Qwell[] | frenzy: yes, then they add candles |
| 23:28.21 | Nodren | Strom_M: i'll get to that later, this is just basic beginner stuff |
| 23:28.24 | Strom_M | Nodren, comment out the macro and use the following line temporarily: |
| 23:28.40 | frenzy | Qwell[]: hmm |
| 23:28.45 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 23:28.51 | Qwell[] | Then, if you're lucky, they'll light them |
| 23:28.55 | Strom_M | exten => _X.,1,Dial(ZAP/G0/${EXTEN}) |
| 23:29.10 | frenzy | Qwell[]: nea.. i'll settle for coffee |
| 23:29.16 | Qwell[] | coffee cake? |
| 23:29.18 | Strom_M | then do an extensions reload and see if you still have the same problem |
| 23:29.28 | frenzy | Qwell[]: coffee coffee ;) |
| 23:31.00 | Nodren | Strom_M: same thing |
| 23:31.01 | timscott | :) |
| 23:31.06 | *** part/#asterisk QbY (i=user@cm-12-146-225-110.dhcp.geo-sc.southerncoastalcable.net) |
| 23:31.09 | Nodren | its showing the entire number being sent to the group |
| 23:31.14 | Nodren | showing zap/3 picking up |
| 23:31.28 | Nodren | then the same message from our phone company telling us no go |
| 23:31.31 | Strom_M | Nodren, you have three analog lines coming into the TDM400 right? |
| 23:31.36 | Nodren | yes |
| 23:31.39 | Strom_M | are the lines in a centrex group? |
| 23:31.47 | Nodren | whats a centrex group? |
| 23:31.54 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 23:32.00 | Strom_M | do this: |
| 23:32.08 | Strom_M | plug a phone into the phone line directly |
| 23:32.16 | Nodren | i've done that |
| 23:32.20 | Strom_M | a real phone |
| 23:32.23 | Nodren | yep |
| 23:32.25 | Strom_M | brb phone |
| 23:32.29 | Az_au | lol |
| 23:32.33 | Qwell[] | What's a brb phone? |
| 23:32.42 | [hC] | Its like a budgetone |
| 23:32.45 | [hC] | only shittier. |
| 23:32.45 | LostFrog | lol.. I knew that was comng. |
| 23:32.46 | Qwell[] | ahh |
| 23:32.53 | Strom_M | ok |
| 23:32.54 | Strom_M | back |
| 23:33.01 | Strom_M | see if you have trouble dialing on all the lines |
| 23:33.04 | Nodren | i've plugged a real phone into each of the 3 lines |
| 23:33.08 | Strom_M | see if the lines require you to dial 9 first |
| 23:33.10 | Nodren | and been able to successfully make a call |
| 23:33.21 | Nodren | used a cheepy 5$ phone from walmart |
| 23:33.21 | Strom_M | Nodren, do you have a buttset handy? |
| 23:33.22 | Nodren | worked fine |
| 23:33.27 | Nodren | buttset? |
| 23:33.34 | LostFrog | Lineman's phone. |
| 23:33.36 | Strom_M | phone technician's test set |
| 23:33.40 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
| 23:33.57 | Strom_M | Harris TS-22A or somesuch ;) |
| 23:33.59 | Nodren | never heard of one before, so most definately no |
| 23:34.09 | Qwell[] | grab your wire cutters... |
| 23:34.24 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 23:34.24 | Strom_M | Nodren, what is the exact error from the telco? |
| 23:34.50 | Az_au | what do your logs say when the call is made in asterisk? |
| 23:34.54 | Az_au | like Executed Dial blah blah? |
| 23:35.06 | Nodren | hang on |
| 23:35.08 | Nodren | lemme try again |
| 23:35.32 | Nodren | heh and now it works |
| 23:35.34 | Gamercjm | Whats agood codec for SIP? |
| 23:35.38 | Gamercjm | g711u? |
| 23:35.40 | Strom_M | Gamercjm, I like ulaw |
| 23:35.42 | Qwell[] | Gamercjm: depends |
| 23:35.42 | Az_au | depends on your seutp |
| 23:35.43 | *** join/#asterisk Ciber311 (n=Ciber@user-1087e94.cable.mindspring.com) |
| 23:35.45 | Nodren | hang on lemme try again |
| 23:35.48 | Az_au | low bandwidth g729 |
| 23:35.52 | Az_au | lan 711u |
| 23:36.00 | LostFrog | Depends on how much bandwidth you can afford. |
| 23:36.02 | Nodren | wierd |
| 23:36.04 | Gamercjm | I just added ulaw only and i cant send audio out of sip |
| 23:36.07 | Nodren | when i unplugged that line to test |
| 23:36.11 | Gamercjm | on my softphone x-lite |
| 23:36.12 | Nodren | with a reg phone and plugged back in |
| 23:36.15 | Nodren | it all the sudden works |
| 23:36.15 | Nodren | bleh. |
| 23:36.24 | Qwell[] | all the sudden? |
| 23:36.31 | Strom_M | Nodren, hmm, odd - perhaps you didnt have the line plugged in securely? |
| 23:36.36 | LostFrog | Loose screw |
| 23:36.37 | LostFrog | ? |
| 23:36.41 | Nodren | possibly |
| 23:36.49 | Nodren | that would explain missing a few dialed numbers |
| 23:37.15 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 23:37.15 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:37.22 | Qwell[] | another symptom of is "losing" digits, is dialing random local numbers |
| 23:37.45 | Nodren | heh i'm ALMOST getting tired of my zelda ringtone from calling my cell so much :P |
| 23:38.02 | Strom_M | Nodren, oh blah, call the local time announcement number |
| 23:38.07 | Strom_M | less irritating |
| 23:38.08 | Nodren | popcorn! |
| 23:38.10 | Nodren | haha |
| 23:38.15 | Strom_M | oh, you're in northern california |
| 23:38.19 | Nodren | yes |
| 23:38.24 | Nodren | heh |
| 23:38.26 | Strom_M | good old Weatherchron machine in san francisco |
| 23:38.28 | Qwell[] | Strom_M: no.. |
| 23:38.30 | Gamercjm | eww yah i just added ulaw,g77a,g77u and like from sip-> cell its all staticy |
| 23:38.32 | Nodren | they have a number local to my area |
| 23:38.35 | Nodren | thats 3334444 |
| 23:38.41 | Nodren | anyway it calls a very umm |
| 23:38.44 | Strom_M | "good afternoon. At the tone, pacific daylight time will be..." |
| 23:38.45 | Qwell[] | erm |
| 23:38.47 | Nodren | non-appropriate popcorn lady |
| 23:38.50 | Qwell[] | silly highlighting |
| 23:39.09 | Strom_M | Nodren, the "popcorn lady" is named Sharon Daniels, btw |
| 23:39.31 | Nodren | wierd |
| 23:39.37 | Nodren | well my boss's cell doesnt work, but mine does |
| 23:39.39 | Nodren | and popcorn lady does |
| 23:39.50 | timscott | Hello there, Strom. |
| 23:40.06 | Strom_M | Nodren, it would be extremely helpful if you had a buttset so you could actually listen to the pbx dialing |
| 23:40.10 | Strom_M | hello tim |
| 23:40.14 | Nodren | yeah |
| 23:40.17 | Nodren | ohh well |
| 23:40.30 | Nodren | maybee if i rearrange my zapata group |
| 23:40.36 | Nodren | it'll use line 1 more often for calls |
| 23:40.36 | DoktorGreg | omg i know ive done this |
| 23:40.48 | Ciber311 | anyone using a spa-942? |
| 23:40.49 | Strom_M | Nodren, what makes line 1 more reliable than the others? |
| 23:40.50 | DoktorGreg | I know this is wildly off topic |
| 23:41.02 | DoktorGreg | how do i make a new layer from selection in photoshop? |
| 23:41.06 | Qwell[] | different areacode or prefix? |
| 23:41.20 | Nodren | same area code and prefix |
| 23:41.26 | Az_au | DoktorGreg: in edit, paste maybe? |
| 23:41.30 | Strom_M | Nodren, is line 1 still more reliable if you swap the phone lines around on the zap card? |
| 23:41.32 | Nodren | well, these lines were at one time going into a phone system |
| 23:41.37 | Nodren | we did a very ammature job |
| 23:41.42 | Nodren | and re terminated the lines |
| 23:41.46 | Nodren | into where 3 of our stations were |
| 23:41.47 | Strom_M | Nodren, they're lines from SBC, right? |
| 23:41.50 | Nodren | yes |
| 23:41.54 | Az_au | DoktorGreg: i think it's paste as new layer or something |
| 23:42.04 | Nodren | needless to say the line connections arnt done professionally |
| 23:42.08 | Nodren | from inside our office |
| 23:42.19 | Strom_M | Nodren, if I were you I'd check your wiring and make sure you're not introducing noise or hum |
| 23:42.36 | Strom_M | call a silent termination test |
| 23:42.47 | Gamercjm | when i enter the codec how do i do it? allow=g711a? |
| 23:42.55 | Qwell[] | Gamercjm: ulaw |
| 23:43.00 | Qwell[] | erm, alaw |
| 23:43.02 | Strom_M | Gamercjm, allow=alaw |
| 23:43.07 | Gamercjm | oh |
| 23:43.10 | Nodren | well that costs money, be it for a technitian, man hours, or equipment.. which is why it probably wont happen |
| 23:43.16 | Nodren | you know how some offices are, boss wants to save every penny |
| 23:43.25 | Nodren | which is why we did the job ourselves |
| 23:43.32 | Gamercjm | is ilbc low band? |
| 23:43.33 | Strom_M | Nodren, IIRC the silent term number is NXX-0040 but dont hold me to that |
| 23:44.10 | Nodren | Strom_M: thanks |
| 23:44.23 | Strom_M | been a while since I lived in the bay area ;) |
| 23:44.26 | Nodren | seems to be working better now, so i guess it was the line |
| 23:44.32 | Nodren | thanks for the help |
| 23:44.37 | Strom_M | Nodren, I'd check your wiring |
| 23:44.57 | Nodren | naw i'll let it torment everyone in the office until my boss breaks down and hires a professional tech |
| 23:45.00 | Strom_M | yes, it may cost money to do it right, but it's a small price to pay for having reliable telephones |
| 23:45.06 | Nodren | working 4/5 times is good enough |
| 23:45.17 | Strom_M | Nodren, that attitude horrifies me |
| 23:45.25 | Nodren | it horrifies me too |
| 23:45.32 | Nodren | i've already wanted to hire a pro to set up asterisk |
| 23:45.39 | Nodren | sadly it hasnt happened |
| 23:45.44 | Nodren | so i gota make due with what i got |
| 23:45.56 | Strom_M | "make do" is the phrase, btw |
| 23:46.09 | Qwell[] | and "all of a sudden" ;/ |
| 23:46.12 | LostFrog | 20% failure? |
| 23:46.25 | LostFrog | I'd be beat up by everyone in my office. |
| 23:46.28 | Strom_M | no nines of reliability! |
| 23:46.36 | Nodren | heh |
| 23:46.37 | Qwell[] | Strom_M: 9 8's? |
| 23:46.45 | LostFrog | I do about 95%. |
| 23:46.56 | Nodren | theres only 12 people in my office |
| 23:47.03 | Nodren | and i can redirect their complaints to my boss |
| 23:47.04 | Qwell[] | < 99.999 is unacceptable |
| 23:47.57 | LostFrog | Qwell[]: That will happen as soon as we get two T1s and a E1 in India. |
| 23:48.12 | Qwell[] | t1 AND e1? |
| 23:48.13 | Qwell[] | why? |
| 23:48.23 | LostFrog | Two T1s in the US. |
| 23:48.27 | LostFrog | Data+telecom. |
| 23:48.30 | Qwell[] | right |
| 23:48.38 | LostFrog | E1 for data in India. |
| 23:50.44 | Nodren | thanks for all the help |
| 23:51.08 | key2 | !seen mark |
| 23:51.22 | key2 | !seen kram |
| 23:51.43 | key2 | the bot is not very talkative |
| 23:52.17 | generalhan | ~seen kram |
| 23:52.20 | jbot | kram <n=mark@pdpc/sponsor/digium/kram> was last seen on IRC in channel #asterisk, 9d 20h 45m 41s ago, saying: 'oh most certainly :)'. |
| 23:53.00 | frenzy | ~seen frenzy |
| 23:53.02 | jbot | frenzy is currently on #asterisk (59m 38s). Has said a total of 13 messages. Is idling for 2s, last said: '~seen frenzy'. |
| 23:53.05 | generalhan | lol |
| 23:53.55 | *** join/#asterisk dextro (n=dextro@cpe-70-116-10-201.austin.res.rr.com) |
| 23:57.05 | [hC] | Soo what is the main difference for polycom 3/5/600 and the 3/5/601 counterparts? |
| 23:57.08 | [hC] | just where they are sold in the world? |
| 23:57.24 | MstlyHrmls | 301 and 501 have 4 megs of flash |
| 23:57.33 | MstlyHrmls | vs 2 Megs in the 300 and 500 |
| 23:57.38 | MstlyHrmls | 601 has EM support |
| 23:57.58 | [hC] | What gets stored in the flash besides firmware? I mean how would that affect a consumer? |
| 23:59.12 | [hC] | and hte 301 doesnt have speakerphone, or is it listen only? |
