| 00:03.17 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 00:05.06 | Vorondil | CrashSys: Of course, you can use a custom player for moh, so it doesn't really matter if mpg321 can or not. :-) |
| 00:12.00 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 00:19.40 | `Sean | hehe |
| 00:19.41 | `Sean | anyone around? |
| 00:19.45 | `Sean | s/hehe/heh/ |
| 00:19.51 | `Sean | jbot wakeup |
| 00:19.58 | `Sean | s/heh/hehw/ |
| 00:20.00 | `Sean | s/heh/hehe/ |
| 00:20.10 | `Sean | bl0berscope |
| 00:20.18 | `Sean | meh im getting boreed |
| 00:20.29 | `Sean | s/bored/boreed/ |
| 00:20.32 | `Sean | s/boreed/bored/ |
| 00:20.37 | `Sean | there we go |
| 00:21.16 | Nuitari | Hi `Sean |
| 00:21.31 | `Sean | Hey Nugget |
| 00:21.33 | `Sean | err |
| 00:21.34 | `Sean | Nuitari |
| 00:21.43 | Nuitari | :) |
| 00:21.53 | Nuitari | Just found out how to set custom device states using the manager interface |
| 00:22.23 | `Sean | Who Owns Jbot again |
| 00:23.46 | *** join/#asterisk qartis (n=qartis@s207-6-25-110.bc.hsia.telus.net) |
| 00:24.08 | Nuitari | hi qartis |
| 00:24.28 | qartis | hi hi |
| 00:24.40 | qartis | somebody is supposed to be showing me how fantastic your ircbot is |
| 00:24.58 | `Sean | heh, qartis fantastic!? sigh nvm dude |
| 00:25.02 | `Sean | s/nvm/nevermind/ |
| 00:25.34 | qartis | `Sean: I'm not sure how noisy this channel gets, but if you have 0.5 - 1 message per second, that would get *way* too noisy |
| 00:25.54 | qartis | I apparently did understand what you meant |
| 00:25.58 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 00:26.03 | `Sean | qartis would only get noisy, if the bot was doing autocorrections |
| 00:26.05 | `Sean | its not tho :) |
| 00:26.07 | qartis | that kind of bot is fairly easy to code |
| 00:26.19 | `Sean | Perl?TCL? |
| 00:26.27 | qartis | `Sean: okay, and what's wrong with you saying "s/nvm/nevermind/" and people parsing it in their head? |
| 00:26.36 | qartis | `Sean: that would turn 3 lines into 2 |
| 00:26.56 | `Sean | qartis nevermind lets drop it |
| 00:27.01 | qartis | `Sean: there are irc bots written in basically every language: shell script, php, perl, python, ruby, |
| 00:27.01 | rob0 | ~jbot |
| 00:27.03 | jbot | jbot is probably a hack!, or known to have only said one useful thing. |
| 00:27.11 | `Sean | hah |
| 00:27.17 | qartis | `Sean: I can name you some popular bots if you'd like |
| 00:27.19 | `Sean | ~tfot |
| 00:27.21 | jbot | tfot is probably "The Future of Telephony", a book about Asterisk from O'Reilly Publishing, ISBN: 0-596-00962-3, click http://www.oreilly.com/catalog/asterisk/ for more details |
| 00:27.30 | `Sean | qartis its ok :) |
| 00:27.43 | qartis | `Sean: #amarok also has a channel bot, that can play games and do all kinds of things |
| 00:28.13 | `Sean | brb |
| 00:28.21 | qartis | but forcefully interpreting s/didnt/didn't/ inline sed code seems very.. unnecessarily noisy |
| 00:28.38 | Nuitari | I wonder |
| 00:28.43 | Nuitari | s/wonder/wander |
| 00:28.46 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net) |
| 00:29.02 | Nuitari | I'm testing |
| 00:29.03 | qartis | s|but forcefully interpreting s/didnt/didn't/ inline sed code seems very.. unnecessarily noisy|to say that jbot is pointless| |
| 00:29.06 | Nuitari | s/testing/sigh/ |
| 00:29.11 | qartis | aww, dang |
| 00:29.13 | Nuitari | hum |
| 00:29.23 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 00:29.24 | Sycofant | Still looking for insight on this: [Jun 14 12:30:35] NOTICE[2536] chan_sip.c: Registration from '<sip:201@asterisk.bunker>' failed for '192.168.1.51' - Not a local domain |
| 00:29.33 | Nuitari | Sycofant: read sip.conf |
| 00:29.54 | Nuitari | s/read|sip/interpret/ |
| 00:29.59 | Sycofant | Have done - seems like it shoudl work. |
| 00:30.06 | Nuitari | did you set domains ? |
| 00:30.19 | Sycofant | domain = bunkermedia.co.nz, asterisk.bunker, 192.168.1.112 |
| 00:30.28 | Nuitari | did you set localnet? |
| 00:30.31 | Sycofant | autodomain = yes |
| 00:30.36 | Nuitari | domain restricts the ips someone can connect to |
| 00:31.11 | Sycofant | Ahh, okay... I'll look into Localnet. |
| 00:31.20 | Nuitari | Action: Logoff |
| 00:31.20 | Nuitari | Response: Goodbye |
| 00:31.20 | Nuitari | Message: Thanks for all the fish. |
| 00:31.21 | Nuitari | :) |
| 00:32.37 | Sycofant | localnet = 192.168.0.0/255.255.0.0 |
| 00:32.41 | Sycofant | Still no happy. |
| 00:33.02 | Nuitari | hum, that was what solved it for me |
| 00:34.00 | Sycofant | This is AsteriskNow, so I am looking at the config and trying to find the settings in the GUI |
| 00:34.57 | Nuitari | I don't know much about asterisknow |
| 00:35.08 | Sycofant | [Jun 14 12:35:41] NOTICE[2536] chan_sip.c: Can't add wildcard IP address to domain list, please add IP address to domain manually. |
| 00:36.45 | Sycofant | Got rid of that, but still not playing nice. |
| 00:36.58 | *** part/#asterisk qartis (n=qartis@s207-6-25-110.bc.hsia.telus.net) |
| 00:38.49 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:44.12 | *** part/#asterisk grey (n=grey@bas3-sudbury98-1168048322.dsl.bell.ca) |
| 00:46.14 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 00:46.14 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:48.30 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 00:48.49 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.67) |
| 00:49.17 | *** join/#asterisk boch (n=fran@190.48.242.132) |
| 01:06.22 | *** join/#asterisk JSabines (n=alancast@189.158.199.236) |
| 01:14.51 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 01:20.47 | *** join/#asterisk n00dle (n=ccraft@hillel.springsips.com) |
| 01:21.37 | n00dle | I could use some help with bugs.digium.com... I need to revise my patch as it breaks things, but the system says access denied when I try to delete the old one. |
| 01:22.04 | blitzrage | n00dle: I can delete that for ya |
| 01:22.06 | blitzrage | which bug number? |
| 01:22.33 | *** join/#asterisk perf3kt (n=perf3kt@adsl-68-77-93-206.dsl.ipltin.ameritech.net) |
| 01:23.26 | n00dle | bug 9973, the only file attached to it. |
| 01:24.22 | n00dle | <TongueInCheek> Asterisk doesn't like it very well when you reference a pointer after it's been free'd. </TongueInCheek> Oops! |
| 01:25.59 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 01:26.03 | blitzrage | n00dle: done |
| 01:26.07 | n00dle | Thanks. |
| 01:26.29 | blitzrage | np! |
| 01:26.31 | n00dle | Why wouldn't it let me delete it? |
| 01:27.09 | russellb | because it only lets people with elevated privilidges do it |
| 01:27.53 | russellb | n00dle: so other than this result var thing, is it working for you now? |
| 01:31.06 | *** join/#asterisk karleeto (i=karl@gentoo.karlhaines.com) |
| 01:32.34 | karleeto | what up folks ? |
| 01:32.57 | karleeto | i'm really loving having our own asterisk setup at work now! |
| 01:34.09 | karleeto | we've got 3 around town that we put in, but just now got our lines swapped over from our T1, so we're running our own VOIP setup at the office now |
| 01:34.28 | karleeto | i've learned quite a bit more, having my own setup to play with |
| 01:34.45 | russellb | karleeto: it's nice to hear from the happy users |
| 01:35.22 | karleeto | russellb: yeah, i bet you get mostly people with a problem, huh? |
| 01:35.27 | n00dle | Gotcha. |
| 01:35.31 | Qwell | karleeto: You don't even know :) |
| 01:35.47 | n00dle | Yeah, russellb, it's working with my (now FULLY tested) hack. |
| 01:35.50 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 01:36.03 | n00dle | Uploading the fixed patch now... |
| 01:36.35 | karleeto | Qwell: i see it most of the time i'm here.. i'm usually 'karlhaines', but i just finally decided on a new handle, and got my reverse ip stuff setup wth my ISP, so i'm now using my new identity ;) |
| 01:36.36 | russellb | karleeto: exactly :) all day every day .. |
| 01:36.45 | russellb | n00dle: well cool, i haven't had much feedback on that code ... |
| 01:37.23 | n00dle | Well, I was able to follow it, and figured out all by myself that it helps to reference a pointer to an object BEFORE it is destroyed! :O |
| 01:38.11 | russellb | yes, that is important :) |
| 01:39.07 | n00dle | Otherwise, it's a sure-fire SEGV and a huge ker-duh! |
| 01:39.47 | n00dle | Ok, now I get to go home... it's been a 10.75 hour day (normally 9), and then the commute. yay |
| 01:40.01 | blitzrage | commutes suck |
| 01:40.08 | n00dle | Ciao, alle, and thanks again! |
| 01:40.54 | karleeto | blitzrage: whats a DTMF frame? seems like i've heard that before |
| 01:41.15 | blitzrage | Dual-Tone Multi-Frequency (i.e. Touch-Tone(tm)) |
| 01:45.15 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:46.31 | russellb | blitzrage: i still hate dtmf |
| 01:47.45 | DTMF | <3 russellb |
| 01:47.55 | *** mode/#asterisk [+b %DTMF!*@*] by russellb |
| 01:47.58 | Qwell | :p |
| 01:48.01 | russellb | hehe |
| 01:48.09 | Qwell | nice slide earlier, BTW :D |
| 01:48.09 | karleeto | lol |
| 01:48.13 | tengulre | LOL.. |
| 01:48.15 | *** mode/#asterisk [-b %DTMF!*@*] by russellb |
| 01:48.24 | Qwell | russell slid into first, to avoid an out, heh |
| 01:48.33 | Qwell | way to take one for the team! |
| 01:48.38 | russellb | thanks :-D |
| 01:48.44 | russellb | it was a fun game ... but now my body hurts |
| 01:49.02 | Qwell | I think Dwayne got like 8 RBIs :P |
| 01:49.06 | russellb | i know, he pwns |
| 01:49.20 | Qwell | he was saying he has to kick ass this game, because of his strikeout last game |
| 01:49.59 | russellb | (we're talking about the Digium softball team, btw) :) |
| 01:50.07 | Qwell | We need ringers. If any of you guys can code and play softball, you should TOTALLY apply to Digium. :P |
| 01:50.42 | Qwell | You know...I may actually...sign up for the second half |
| 01:50.56 | file | Qwell: *GASP* |
| 01:50.57 | russellb | you should! |
| 01:50.58 | [TK]D-Fender | 10 GOTO FIRST-BASE 20... *segfault* |
| 01:51.04 | Qwell | I should. I'm there anyways. |
| 01:51.07 | russellb | yup |
| 01:51.14 | russellb | and we're always so close to not having enough people ... |
| 01:51.18 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-106-85-185.hag.east.verizon.net) |
| 01:51.19 | Qwell | I'll ask Lauren about it tomorrow |
| 01:51.22 | file | dooooo it |
| 01:51.36 | Qwell | file you must too |
| 01:51.49 | file | eep |
| 01:52.10 | file | I refuse |
| 01:52.39 | russellb | fclose(file); |
| 01:52.39 | Qwell | too late, I already signed you up |
| 01:52.41 | russellb | take that. |
| 01:52.46 | Qwell | Any time you're here, you must play. |
| 01:52.50 | Qwell | OR cheer |
| 01:53.07 | file | k! |
| 01:53.07 | russellb | ooh, cheerleader |
| 01:53.15 | Qwell | pom-poms and all |
| 01:53.35 | file | eep |
| 01:53.54 | russellb | i am so glad that asterisk doesn't take long to compile |
| 01:54.07 | blitzrage | lies |
| 01:54.08 | Qwell | russellb: yeah, I could never do openoffice/qt dev |
| 01:54.14 | blitzrage | Qwell: ewwww! |
| 01:54.14 | file | Digium Digium go go go! analog signalling? no no no! |
| 01:54.26 | Qwell | file: there you go |
| 01:54.26 | file | gooooooo Digium! |
| 01:54.40 | russellb | i have a patch i need to submit, but it's going to be another hour before i can verify the fix because i have to wait for it to build, gah |
| 01:54.53 | Qwell | next time russellb is up to bat, I'm gonna shout "Pretend the ball is VLDTMF!' |
| 01:55.13 | russellb | it's not VLDTMF, really ... it's just ... dtmf processing in general |
| 01:55.18 | *** join/#asterisk kolian123 (n=kvirc@124.107.63.223) |
| 01:55.20 | Qwell | whichever |
| 01:55.27 | russellb | it seems like it should be so simple |
| 01:55.29 | kolian123 | Hi TK |
| 01:55.30 | file | russellb: way to spoil a comment there |
| 01:55.32 | russellb | but it's so damn complicated |
| 01:55.35 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:55.44 | russellb | file: thanks. |
| 01:56.12 | [TK]D-Fender | kolian123, 3/4 of Digium is online now... go ask them :) |
| 01:56.23 | kolian123 | really? |
| 01:56.29 | kolian123 | what channel? |
| 01:56.31 | Qwell | bbl |
| 01:56.36 | [TK]D-Fender | lol |
| 01:56.41 | kolian123 | Anyway i got the card workiing:) |
| 01:56.50 | kolian123 | One span! |
| 01:56.53 | [TK]D-Fender | ok guys, its safe to return! |
| 01:57.01 | Qwell | You know, I'm surprised nobody has commented on the new pcie card on the lists |
| 01:57.07 | Qwell | I expected people to troll.. I really did. |
| 01:57.08 | kolian123 | But the bad news! |
| 01:57.11 | Qwell | bbl |
| 01:57.12 | russellb | Qwell: heh .. |
| 01:57.17 | russellb | bad news? |
| 01:57.29 | kolian123 | only span 3 is working the rest failing loopback test |
| 01:57.30 | Qwell | cards* |
| 01:57.36 | Qwell | kolian123: digital card? |
| 01:57.41 | Qwell | te4xxp? |
| 01:57.44 | kolian123 | te405p |
| 01:57.49 | Qwell | that's funky |
| 01:57.56 | Qwell | I'd call support tomorrow, honestly |
| 01:58.01 | Qwell | it's a new card, right? |
| 01:58.04 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 01:58.11 | kolian123 | Nope it's used on from ebay |
| 01:58.13 | Qwell | oh |
| 01:58.37 | kolian123 | would you say the card is dead if loopback isn't working? |
| 01:58.39 | Qwell | I'm not sure if the install support is offered if you buy a card used |
| 01:58.47 | kolian123 | or still PCI/IRQ issues? |
| 01:58.52 | Qwell | kolian123: if it's working on one port, and not the others...yeah, probably :( |
| 01:59.44 | kolian123 | :( |
| 02:00.06 | kolian123 | the card has lights and comes up in driver but doesn't sync PRI |
| 02:00.07 | Qwell | russellb: any idea if install support is still offered on resale, assuming it's in the 2 year warranty period? |
| 02:00.14 | Qwell | or, actually, you know...that wouldn't even be install support |
| 02:00.20 | Qwell | any idea how old the card actually is? |
| 02:00.25 | kolian123 | one year |
| 02:00.27 | russellb | no clue |
| 02:00.37 | Qwell | call support tomorrow, explain you bought it used, see what they can do |
| 02:00.56 | kolian123 | thanks will call them up, see if they can help with something |
| 02:00.58 | Qwell | if it's in the warranty period, it really shouldn't matter |
| 02:01.19 | kolian123 | Do cards have 2 years now? |
| 02:01.24 | Qwell | don't they? |
| 02:01.31 | kolian123 | let me check |
| 02:01.36 | Qwell | I don't know - I'm just a developer |
| 02:01.41 | russellb | heh |
| 02:02.17 | kolian123 | hehe! |
| 02:02.29 | kolian123 | are you working for digium? |
| 02:02.40 | Qwell | yeah |
| 02:02.49 | Corydon76-home | Right now, he's working for his wife |
| 02:04.02 | russellb | zing! |
| 02:04.16 | file | russellb: how is Chateau Bryant? |
| 02:04.27 | russellb | it's houseish |
| 02:04.30 | Qwell | "Such notice may be given by facsimile transmission, or other reliable means" haha |
| 02:04.43 | Qwell | I don't know why I think that's so funny |
| 02:05.23 | kolian123 | seems like 2 years! |
| 02:05.45 | kolian123 | your wife's money is her money |
| 02:05.50 | kolian123 | and your money is her money |
| 02:06.08 | karleeto | so, my prouction systems around town are using TB, and i've noticed options for FAX in there, does asterisk handle fax'es ? if so, why have i come accross services that you can port your fax number over to them, and they'll email faxes to you?? |
| 02:06.30 | karleeto | kolian123: is this card you're speaking of with 4 ports a modular card? |
| 02:06.37 | Qwell | efax and the like are incredibly expensive |
| 02:06.50 | Qwell | I think it's in excess of $.10/page |
| 02:06.54 | kolian123 | karleeto, it's te405p with a echo module |
| 02:07.15 | kolian123 | it's PRI a card |
| 02:07.41 | karleeto | kolian123: but the ports each have modular cards is what i'm asking, like the TDM cards? |
| 02:07.54 | Qwell | karleeto: no |
| 02:07.55 | kolian123 | no its just build into the card |
| 02:08.02 | Qwell | the echo can is a module though |
| 02:08.05 | kolian123 | the module is echo canceller |
| 02:08.11 | karleeto | OiC, that sucks! |
| 02:08.26 | Qwell | karleeto: it would be non-trivial to do |
| 02:08.36 | Qwell | at least, if you wanted analog and PRI on the same card |
| 02:08.39 | kolian123 | Qwell, when i call support they need serial number or receipt to trace how old is the card. Would you know? |
| 02:08.49 | Qwell | serial number, it should be ont hec ard |
| 02:08.50 | karleeto | i've heard a lot about people having cards with no working parts.. this is why when i pay that much money for a damn PCI card, they better have modular parts |
| 02:08.52 | Qwell | ...on the card |
| 02:09.32 | kolian123 | Thanks hopefully it's still on the warranty. would be nice to get a replacement |
| 02:10.47 | karleeto | Qwell: i have a question for you!! so, i've been playing with a "Clone" X100P card (ambient chip) at the shop.. it works pretty great, but the only issue is, when zaptel is reloaded, or unloaded (like on a reboot), my kernel panics out, therefor i never get a successful clean reboot |
| 02:11.23 | Qwell | karleeto: there is an issue with now with the init scripts.. I think it causes a panic |
| 02:11.33 | Qwell | remove the ztcfg -s from the unload part of the script |
| 02:11.44 | karleeto | Qwell: i've ordered 2 cards of Ebay, voxcom cards, that claim to be "Genuine", and even when zaptel loads them, they even say "X100P", no Clone |
| 02:11.56 | Qwell | they aren't genuine :p |
| 02:12.02 | Qwell | Digium hasn't sold those cards in...forever |
| 02:12.09 | karleeto | Qwell: well, obviously |
| 02:12.24 | karleeto | Qwell: but they must be better than using an old-ass ambient chip card |
| 02:12.35 | Qwell | they're pretty much all junk... |
| 02:12.43 | karleeto | Qwell: hmmm.. shit ;) |
| 02:13.08 | karleeto | Qwell: well, hopefully the two new ones will at least be more reliable than the single ambient card i've got now |
| 02:13.47 | karleeto | Qwell: i need em!! cause i have two phone lines, one that rolls to the other, and only one card ATM, so when i'm on the phone people get rolled to the second line, and it just rings forever |
| 02:14.09 | Qwell | well, you could just get a single tdm400p, with 2 fxo modules |
| 02:14.19 | karleeto | Qwell: so you think that removing -s in the init script might solve my current problem for the time being? |
| 02:14.28 | Qwell | karleeto: remove the whole line |
| 02:14.50 | *** join/#asterisk guillote_GNU (n=guillote@190.7.30.134) |
| 02:14.59 | karleeto | Qwell: i have 3 businesses in town that i have TDM400 cards in, I LOVE THEM! |
| 02:16.14 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 02:16.57 | karleeto | Qwell: and i certainly plan on buying one for our shop once i can afford it |
| 02:17.15 | kolian123 | Hi Qwell, another question |
| 02:17.33 | karleeto | Qwell: its just that we spent a lot of money getting our box built and our lines switched back to normal lines, from our T1 |
| 02:17.43 | kolian123 | Do you know how the quality of older t405p |
| 02:17.55 | Qwell | why not have the T1 in one office, and voip the rest? |
| 02:18.30 | karleeto | Qwell: cause we are canceling the T1 service, got MORE speed from a cable modem, and are gonna save $500 dollars a month |
| 02:19.04 | karleeto | Qwell: so in a month or two, when we actually start to see that money we are saving, thats when we'll buy a TDM400 |
| 02:19.09 | kolian123 | for PRI termination |
| 02:19.11 | karleeto | and two FXO modules for it |
| 02:19.44 | Qwell | karleeto: ahh |
| 02:19.59 | karleeto | Qwell: untill then we're gonna duke it out with two X100p cards for our phone lines |
| 02:20.39 | karleeto | Qwell: and we're gonna sell all of the old panasonic pbx equipment on ebay too, probably get a nice little chuck for that stuff as well |
| 02:20.40 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 02:22.33 | *** join/#asterisk JT_ (n=jon@unaffiliated/jt) |
| 02:22.54 | rue_mohr | I'v solved the second most proplexing problem with my system |
| 02:23.04 | rue_mohr | where to put the hardware |
| 02:23.20 | rue_mohr | turn out the answer is simple, because the channelbank just needs a T1, the computer can go anywhere |
| 02:23.29 | rue_mohr | which means it will fit in the closet |
| 02:23.53 | rue_mohr | as long as I dont use connectors that are too big |
| 02:24.08 | rue_mohr | and i oust the doorbell speakers |
| 02:26.28 | russellb | victory! |
| 02:26.32 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:29.32 | rue_mohr | quite |
| 02:29.36 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 02:29.50 | rue_mohr | now, I have to battle paint, so much less forgiving |
| 02:31.21 | rue_mohr | ...and I need to find a chunk of plywood to mount this on... |
| 02:31.22 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 02:31.55 | mmlj4 | the linksys ATAs, i want one that's $foo-NA, or non -NA? |
| 02:32.27 | [TK]D-Fender | mmlj4, SPA-2102 <- |
| 02:32.47 | mmlj4 | i want FXO, sorry |
| 02:33.00 | mmlj4 | the 3000 does both, which is acceptable |
| 02:33.13 | mmlj4 | but it's -NA |
| 02:33.19 | [TK]D-Fender | mmlj4, spa-3102 |
| 02:33.19 | mmlj4 | that means it's locked? |
| 02:34.28 | mmlj4 | Linksys SPA3102 NA 1FXS / 1FXO Analog VoIP Gateway # http://www.voipsupply.com/product_info.php?products_id=1646 |
| 02:35.22 | [TK]D-Fender | mmlj4, http://www.telephonydepot.com/product_p/105-054-312.htm |
| 02:36.06 | mmlj4 | same thing, only sipura: http://www.sipura.com/products/spa3000.htm |
| 02:36.29 | [TK]D-Fender | mmlj4, Sipura was bought out by Linksys YEARS ago. Sipura no longer EXISTS |
| 02:36.29 | mmlj4 | ok, the link you gave me is an -NA |
| 02:37.00 | mmlj4 | there used to be an issue where some where locked in some way... was in -NA or not, do you remember? |
| 02:37.53 | [TK]D-Fender | -NA = not locked, and thats the PAP2 garbage |
| 02:37.57 | [TK]D-Fender | AVOID |
| 02:38.11 | mmlj4 | avoid PAP2? ah. |
| 02:38.59 | [TK]D-Fender | mmlj4, it has a weaker CPU and doesn't support T.38 |
| 02:39.09 | mmlj4 | ah. |
| 02:39.18 | [TK]D-Fender | mmlj4, SPA-3102 has a bigger CPU than the 3000 and can act as a router as well. |
| 02:39.38 | mmlj4 | fair enough, thanks :-) |
| 02:40.50 | mmlj4 | so telephony depot is a reputable web store? I've bought from voipsupply (i think) before |
| 02:41.19 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 02:44.32 | [TK]D-Fender | mmlj4, I've bought from both |
| 02:44.56 | [TK]D-Fender | mmlj4, VS is overpriced for nothing. Their service is decent, but not justifiable. |
| 02:45.43 | mmlj4 | k |
| 02:46.10 | file | has it really been years since they were bought out? |
| 02:46.23 | *** join/#asterisk ACiDV (n=dan@97-147.dr.cgocable.ca) |
| 02:46.39 | mmlj4 | maybe... i bought 2 right as Katrina hit, they have Sipura printed on the cases |
| 02:46.49 | mmlj4 | so that's 2 years ago |
| 02:46.56 | *** join/#asterisk guillote_GNU (n=guillote@190.7.30.134) |
| 02:47.34 | mmlj4 | incidentally, VS treated me very well, considering I lived in the hurricane's destruction area |
| 02:48.23 | mmlj4 | s/lived/live/ |
| 02:48.36 | mmlj4 | heh, neat feature |
| 02:49.31 | russellb | <PROTECTED> |
| 02:49.41 | file | russellb: nope! |
| 02:49.45 | russellb | d'oh. |
| 03:01.59 | kolian123 | Hi Russell |
| 03:02.07 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.us.warpdriveonline.com) |
| 03:02.17 | kolian123 | Seems like my machine got e1000 on board |
| 03:02.20 | russellb | greetings |
| 03:02.28 | kolian123 | Still up? |
| 03:02.43 | russellb | not really, no. |
| 03:02.52 | kolian123 | hehe:) |
| 03:03.21 | kolian123 | do digital cards not play well with e1000 driver? |
| 03:03.28 | russellb | i have no idea. |
| 03:03.37 | Qwell | used to have problems, I think |
| 03:03.45 | kolian123 | Hi Qwell:) |
| 03:03.55 | karleeto | Qwell: so you think that removing -s in the init script might solve my current problem for the time being? |
| 03:04.27 | SirThomas_Home | I have two "normal" cordless phones hooked up to FXS ports on a rhino card. I used to be able to use "#" to transfer a call from them when they were hooked into a digium card... now that no longer works. Any ideas/pointers? |
| 03:04.42 | kolian123 | Was it a driver problem or hardware? |
| 03:06.12 | blitzrage | file: 1.0459 |
| 03:08.11 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 03:13.29 | karleeto | Qwell: sorry about the dupe paste |
| 03:15.30 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.67) |
| 03:16.01 | karleeto | Qwell: thanks for your help! |
| 03:16.04 | *** part/#asterisk karleeto (i=karl@gentoo.karlhaines.com) |
| 03:16.25 | *** join/#asterisk bonderponder (n=test@201.199.68.150) |
| 03:17.29 | bonderponder | hello, anybody knows why when I hit the transfer bottom of a GrandStream GXP2000 or BT200 it will go in blind transfer ? how can I change this ? |
| 03:24.09 | *** join/#asterisk ivanfm (n=ivanfm@c934f322.virtua.com.br) |
| 03:25.57 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:26.33 | *** join/#asterisk n00dle (n=ccraft@ip-249-27.springsips.com) |
| 03:30.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:32.32 | *** join/#asterisk waptaxi (n=cahe@stat-5-160.e-sky.ru) |
| 03:34.50 | *** join/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 03:35.43 | nowork | hi, how many line limitaion for dial-plan in context |
| 03:35.52 | nowork | will 100lines be okay? |
| 03:36.51 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 03:39.07 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 03:40.44 | russellb | 100 lines is fine |
| 03:42.20 | *** join/#asterisk waptaxi (n=waptaxi@stat-5-160.e-sky.ru) |
| 03:49.40 | *** join/#asterisk saftsack (n=oliver@p54A7C383.dip0.t-ipconnect.de) |
| 03:50.48 | Nuitari | Does asterisk manager uses \n or \r\n? |
| 03:52.01 | russellb | \r\n |
| 03:52.05 | Nuitari | thanks |
| 03:52.06 | *** join/#asterisk bmg505 (n=leon@196.209.181.170) |
| 03:52.11 | russellb | you're welcome |
| 03:52.30 | Nuitari | thanks for devstate btw |
| 03:52.39 | russellb | you're welcome, glad you like it |
| 03:53.40 | Nuitari | I'm writing a script to have device states across pbxes |
| 03:53.48 | russellb | interesting |
| 03:53.56 | russellb | i have been looking at ways to do that within asterisk |
| 03:53.56 | *** join/#asterisk Greenbox (n=Brett@user-24-214-124-177.knology.net) |
| 03:53.57 | jql | yeah, that's fun |
| 03:54.16 | jql | I use the agi:// system for that |
| 03:57.07 | Nuitari | jql: how does your works? |
| 03:58.19 | jql | I have a bank of asterisk servers running a dialplan which has AGI(agi://10.0.0.1/track) calls wrapped around every call/hangup |
| 03:58.26 | russellb | hopefully i'll have that solved within asterisk by 1.6 |
| 03:58.34 | russellb | whenever i can find time away from bugs |
| 03:58.46 | jql | that agi script reads a bunch of variables from the server and stores the status of that phone in a db |
| 03:59.08 | russellb | jql: nice |
| 03:59.34 | jql | and then the agi script will send a GOTO to the server to control the call-flow state machine |
| 04:00.12 | Nuitari | I'm looking to do it through the manager, to both look at events and set the devstate, in a db less way |
| 04:00.18 | jql | such as "that phone is off the hook; enjoy some hold music" |
| 04:04.00 | Sweeper | hey, is there a list of asterisk consultants somewhere I can put my name on? |
| 04:05.12 | jql | I can't vouch for it, but I have http://www.asterisk-jobs.com/ in my bookmarks |
| 04:06.54 | Sweeper | worth a shot |
| 05:38.23 | *** join/#asterisk jbot (n=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 05:38.23 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.18, 1.4.3 (June 8, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 05:38.33 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 05:39.33 | *** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net) |
| 05:42.12 | *** join/#asterisk A-Data (n=asd@196.218.172.170) |
| 05:42.33 | A-Data | hello all ... any one know Java SIP client |
| 05:42.42 | A-Data | webbased java |
| 05:42.53 | jql | cisco phones |
| 05:43.00 | jql | but their somewhat embedded |
| 05:43.13 | A-Data | yes i need an embedded one |
| 05:43.20 | tzafrir_laptop | for that many lines a channel bank is an overkill |
| 05:43.33 | tzafrir_laptop | get an analog card (e.g.: TDM400P) |
| 05:44.39 | Sycofant | Anyone used a Cisco 7912G with Asterisk? |
| 05:47.11 | A-Data | Does any one know if patton smartnode 4110 can be used with * |
| 05:47.17 | A-Data | http://www.patton.com/products/pe_products.asp?category=51&MiDAS_SessionID=f120972672a7459e8bf777d234945b5a |
| 05:48.41 | *** join/#asterisk iBuMp (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 05:49.26 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 05:51.52 | Zion800 | Has anyone here used page.agi before? its used for intercom |
| 05:52.46 | Zion800 | I need help editing it to work with Asterisk 1.4 |
| 05:53.22 | Zion800 | Its a very small change, but I don't know Perl. The output of "asterisk -rx "show hints" has changed in Asterisk 1.4, and the script requires it to work properly. |
| 05:53.23 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 05:54.31 | [[blah]asfd | trying to do exten => _6XXX,hint,${EXTEN} doesnt work. is there a way to make it work so that I dont have to do 6000,hint,SIP/6000 for every extensions? |
| 05:54.40 | [[blah]asfd | can hit be used with pattern matching? |
| 05:55.35 | Nuitari | dunno |
| 05:55.42 | Zion800 | I dunno...i set up a hint for every individual extension :/ |
| 05:55.50 | Nuitari | but at the very least, exten => _6XXX,hint,SIP/${EXTEN} |
| 05:56.16 | Nuitari | you can try and see |
| 05:56.44 | [[blah]asfd | yeah.. it doesnt work.. .thats why i am asking. |
| 05:56.47 | *** join/#asterisk saftsack (n=oliver@p54A7D7CD.dip.t-dialin.net) |
| 05:56.54 | Nuitari | with the SIP/ ? |
| 05:57.20 | [[blah]asfd | yeah... sorry, I typed it wrong here. i am doing it with sip in the dialplan |
| 05:58.09 | Nuitari | looks like it doesn't work, I tried the same way back |
| 05:59.30 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:00.17 | Zion800 | So does no one wanna help me out with a little perl script? :) |
| 06:01.04 | jql | I like perl |
| 06:01.18 | Zion800 | :) wanna help me out |
| 06:01.25 | jql | whatsup? |
| 06:02.14 | Zion800 | a script (page.agi) was written for Asterisk 1.2 that looks at hints by executing "asterisk -rx "show hints"" and determining what channels are available to page |
| 06:02.28 | Zion800 | well, the output of "show hints" has changed in asterisk 1.4 |
| 06:02.35 | Zion800 | so now, the script doesnt work properly |
| 06:02.47 | jql | interesting |
| 06:02.57 | Zion800 | Heres the script if you wanna see: http://www.voip-info.org/wiki/view/Script+to+page+mixed+SIP+%252F+SCCP+system |
| 06:04.03 | jql | that's pretty simple |
| 06:04.03 | jql | you probably only need to change a number |
| 06:04.04 | Zion800 | hehe |
| 06:04.05 | jql | paste an example 'show hints' to pastebin.ca or something |
| 06:04.10 | Zion800 | ok |
| 06:04.25 | Zion800 | http://www.pastebin.ca/564597 |
| 06:04.33 | Zion800 | has the script, and show hints |
| 06:04.45 | Zion800 | they've actually changed it to "core show hints" in asterisk 1.4 |
| 06:04.54 | Zion800 | "show hints" is deprecated now |
| 06:05.20 | jql | I see |
| 06:06.57 | jql | okay, I think I see |
| 06:07.09 | Zion800 | yay :p |
| 06:08.24 | jql | change this: @sips = grep(/^\s+\d+\s.*/, sterisk -rx "show hints"; |
| 06:08.30 | jql | to this: @sips = grep(/^\s+\d+.*/, sterisk -rx "show hints"; |
| 06:08.39 | jql | err |
| 06:08.46 | jql | ignore the lost ` stuff |
| 06:08.59 | jql | my cut&paste is sensitive to it. :) |
| 06:09.06 | Zion800 | what do you mean? |
| 06:09.19 | jql | that actually says `asterisk, but I pasted just sterisk |
| 06:09.22 | jql | see? |
| 06:09.26 | Zion800 | ohh |
| 06:09.26 | Zion800 | ok |
| 06:09.29 | Zion800 | haha |
| 06:09.37 | Zion800 | lemme try it] |
| 06:09.41 | jql | ignore my paste, do the right thing, just remove the letters \s where I did |
| 06:09.53 | Zion800 | ok |
| 06:10.01 | jql | I think that was your only problem |
| 06:12.58 | Zion800 | cool! |
| 06:13.04 | Zion800 | it worked like a charm! |
| 06:13.08 | jql | cheers |
| 06:13.13 | Zion800 | thanks a lot |
| 06:13.16 | Zion800 | that was a huge help |
| 06:13.28 | jql | be kind; update the wiki. :) |
| 06:13.33 | Zion800 | i was just about to! |
| 06:20.26 | *** join/#asterisk saftsack (n=oliver@p54A7F6D7.dip.t-dialin.net) |
| 06:23.07 | *** join/#asterisk gardo (n=gardo@121.97.211.111) |
| 06:25.15 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 06:26.32 | creativx | weehaa |
| 06:26.45 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:28.21 | A-Data | does any one know a java or ocx SIP client that can run from web page |
| 06:29.06 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 06:30.48 | A-Data | does any one know a java or ocx SIP client that can run from web page |
| 06:31.12 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:31.47 | Nuitari | A-Data: Look on voip-info.org, I know there is one there under Soft Phones |
| 06:32.15 | kaldemar | google is also very helpful. |
| 06:34.40 | A-Data | kaldemar 7 hours searching in google nothing found |
| 06:38.19 | kaldemar | oh. i put "java applet sip phone" and got jain as the first result. |
| 06:39.52 | A-Data | Nuitari i found a normal java but i need a java applet one so that it can run from web site |
| 06:39.55 | A-Data | :( |
| 06:39.59 | *** join/#asterisk bkw__ (n=brian@adsl-64-149-33-245.dsl.tul2ok.sbcglobal.net) |
| 06:44.56 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:48.05 | *** join/#asterisk errr_ (n=errr@fedora/errr) |
| 06:49.54 | *** join/#asterisk Marshall-Laptop (n=eman0n@cpe-76-181-117-192.columbus.res.rr.com) |
| 06:51.14 | *** join/#asterisk fbffff (n=fbffff@dsl092-129-089.chi1.dsl.speakeasy.net) |
| 06:59.18 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:06.30 | Nuitari | hum, to select or to thread, that is the question |
| 07:10.36 | creativx | thread |
| 07:15.33 | Nuitari | of course, but that won't happen tonight then |
| 07:15.41 | Nuitari | ups will be coming and I still need to get some sleep |
| 07:16.15 | Nuitari | anyways threads will mean a bunch of IPC |
| 07:19.22 | creativx | well Nugget |
| 07:19.23 | creativx | err Nuitari |
| 07:19.28 | creativx | i ahve no idea what you are talking about. |
| 07:19.29 | creativx | but go on. |
| 07:20.10 | Nuitari | I'm bashing out a script to connect to some Asterisk managers to track devices and set device states across them |
| 07:20.25 | Nuitari | so that you can see the presence on someone not on your pbx |
| 07:21.25 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:22.24 | *** join/#asterisk mirco (n=mirco@87.139.74.16) |
| 07:26.01 | *** join/#asterisk DTE_it (n=pier@85-18-112-194.ip.fastwebnet.it) |
| 07:26.11 | DTE_it | morning |
| 07:26.33 | DTE_it | i've a problem with asterisk and asterisk-gui |
| 07:26.51 | DTE_it | i've installed asterisk on debian etch and everything is ok |
| 07:27.10 | DTE_it | i've installed asterisk gui...and when i try to login i get in the log |
| 07:27.19 | creativx | Nuitari: cool |
| 07:27.22 | DTE_it | <PROTECTED> |
| 07:27.23 | creativx | presence is popular these days |
| 07:29.09 | Nuitari | and useful |
| 07:29.41 | Nuitari | I'll just write with select and if it bogs down too much I'll use threads |
| 07:31.23 | *** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl) |
| 07:32.26 | creativx | yeah |
| 07:32.38 | creativx | im just implementing it myself these days |
| 07:35.28 | *** join/#asterisk yassaccan (n=yassacca@admin237.hgo.se) |
| 07:36.01 | Nuitari | how are you doing it? |
| 07:36.40 | creativx | well |
| 07:36.53 | creativx | we run the call logic off a different server |
| 07:37.09 | creativx | but basically people can change their statuses |
| 07:37.11 | creativx | like in msn |
| 07:37.23 | creativx | and it affects what happens to the telephony, and everybody can see everybody else |
| 07:37.35 | creativx | and their current user/presence statuses, and extension statuses |
| 07:37.49 | Nuitari | I have that on the polycom phones, but I can't get the pbx to see it |
| 07:38.17 | creativx | i dont use the phones |
| 07:38.20 | creativx | i use asterisk for it |
| 07:38.29 | creativx | so you can use any sip phone you want basically |
| 07:38.49 | creativx | its all implemented in our inhouse crm app |
| 07:39.01 | Nuitari | ok so it's not on the phone itself? |
| 07:39.23 | Nuitari | eg not using the busy line field? |
| 07:40.33 | gardo | anyone knows what version of asterisk the business class edition is? |
| 07:45.43 | creativx | Nugget: correct |
| 07:45.56 | creativx | Nugget: i let asterisk tell me everything by listening to the ami |
| 07:46.20 | creativx | and we have a internal policy that the phones should always be accessible to dial internally |
| 07:46.36 | creativx | but if you change your status it will affect your queue memberships and if DID calls reach your phone |
| 07:47.26 | Nuitari | What I'm trying to accomplish is that if someone is on the phone on pbx1 that it also shows on pbx2 3 and 4 to the other clients on them |
| 07:47.27 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 07:47.45 | creativx | yeah I understand |
| 07:47.53 | creativx | what about hints? |
| 07:48.01 | Nuitari | hints only work on the local pbx |
| 07:48.13 | creativx | ive never tried monitoring more than 1 pbx so i dunno |
| 07:48.13 | Nuitari | I'm using them and devstate to set some custom hints |
| 07:48.14 | creativx | ah |
| 07:48.30 | Nuitari | though func_devstate is only in trunk |
| 07:48.38 | creativx | but how does users at pbx1 know which users is at pbx2? |
| 07:49.10 | Nuitari | probably the admin would know |
| 07:49.23 | creativx | is this two pbxes for the same company? |
| 07:49.24 | Nuitari | it's assumed that it's not in the wild pbxes |
| 07:49.27 | Nuitari | yeah |
| 07:49.43 | creativx | yeah so if im at pbx1 and want to call george @ pbx2 i would have to know his extension there |
| 07:50.00 | Nuitari | though it's going to generate custom names like host_SIP/4000 and then someone can just set the hints he needs |
| 07:50.22 | creativx | yeah i see |
| 07:50.29 | creativx | so you are basically using phones for dialling |
| 07:50.38 | Nuitari | That can be done in anyway |
| 07:50.39 | creativx | here its the other way around, I smacked it all into the screen |
| 07:50.48 | Nuitari | what matters is devices and the blf list |
| 07:51.43 | Nuitari | any dialing should work |
| 07:52.04 | *** part/#asterisk DTE_it (n=pier@85-18-112-194.ip.fastwebnet.it) |
| 07:56.46 | creativx | mkay |
| 07:57.10 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 07:57.10 | creativx | so many approaches! |
| 07:57.22 | Nuitari | yeah |
| 07:57.26 | skirmisha | guys can host in sip.conf to accept masks also |
| 07:58.11 | skirmisha | ? |
| 07:58.19 | skirmisha | like 192.168.1.0/24 |
| 07:59.01 | skirmisha | or i have to use dynamic + allow option |
| 08:00.20 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:00.52 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:02.26 | walhala | hi |
| 08:02.59 | walhala | thanks for work all asterisk is really a good project ! |
| 08:04.14 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 08:04.15 | zeeesh | hi |
| 08:06.32 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 08:10.53 | creativx | this is so fun.. why is it my ip10s thinks that volume 1 should be the same as volume 6! |
| 08:18.01 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-161-5.columbus.res.rr.com) |
| 08:24.49 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
| 08:25.53 | tanacsdavid | Good Morning! |
| 08:26.02 | Nuitari | sigh, morning already |
| 08:26.25 | tanacsdavid | I have another great question... |
| 08:27.41 | *** join/#asterisk kova (n=kova@tech.quentris.com) |
| 08:27.58 | tanacsdavid | How can I sort the incoming calls? I hava multiple accounts at the same provider. I defined multiple extensions, but every inbound calls land at the last section is my sip.conf |
| 08:28.04 | tanacsdavid | Is there any solution? |
| 08:28.23 | tanacsdavid | Seems like Asterisk is sorting the SIP calls according to domains, not accounts |
| 08:28.49 | *** join/#asterisk friedrich| (n=friedric@e177253033.adsl.alicedsl.de) |
| 08:28.57 | Nuitari | It depends on your config at the provider and to which context the calls are supposed to go to |
| 08:29.03 | *** join/#asterisk Marshall-Laptop (n=eman0n@cpe-76-181-161-5.columbus.res.rr.com) |
| 08:29.03 | Nuitari | though I can't really help you much more then that |
| 08:29.16 | tanacsdavid | OK, thank You. |
| 08:29.28 | tanacsdavid | I'll start my walk on that way. |
| 08:32.18 | *** join/#asterisk yannj_fr (n=yannj@APuteaux-152-1-24-247.w82-120.abo.wanadoo.fr) |
| 08:35.01 | *** join/#asterisk matsk (n=mk@194.68.102.174) |
| 08:39.06 | *** join/#asterisk angryuser (n=Miranda@df01t2-212-195-106-226.d4.club-internet.fr) |
| 08:39.59 | angryuser | good day, the echocancel=yes must be set for each channel or only after [channels] once in zapata.conf? |
| 08:43.24 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:47.57 | penguinFunk | you can set it once for all channels by putting it before all the channel definitions |
| 08:48.15 | penguinFunk | (only once) |
| 08:55.22 | *** join/#asterisk HarryR (n=HarryR@host-83-146-53-46.bulldogdsl.com) |
| 08:55.41 | Nuitari | w00tness!!!!!! |
| 08:55.51 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 08:57.17 | creativx | wuff wuff |
| 08:58.05 | Nuitari | it works |
| 08:58.49 | Nuitari | I'll get some sleep, then I'll cleanup the code and post it online |
| 08:59.31 | darkskiez | Nuitari: for what? |
| 09:04.12 | *** join/#asterisk saftsack (n=saftsack@pD9E044CC.dip.t-dialin.net) |
| 09:05.00 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 09:05.10 | *** part/#asterisk saftsack (n=saftsack@pD9E044CC.dip.t-dialin.net) |
| 09:05.30 | *** join/#asterisk saftsack (n=saftsack@pD9E044CC.dip.t-dialin.net) |
| 09:10.47 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 09:11.32 | *** join/#asterisk denke (n=denke@mehess.adsl.datanet.hu) |
| 09:11.47 | kova | Does anyone here use Asterisk with amr codec? |
| 09:16.21 | A-Data | any one know what is the different between proxyaddress and serveraddress????? sure in * case :> |
| 09:17.14 | *** join/#asterisk ChrisTSIS (n=killa666@24.182.21.208) |
| 09:19.25 | angryuser | A-Data: if yoy have only one * put the same thing, to know the difference read about sip gates |
| 09:20.22 | angryuser | penguinFunk: thx |
| 09:21.37 | penguinFunk | np :] |
| 09:21.44 | *** join/#asterisk HarryR (n=Administ@host-83-146-53-46.bulldogdsl.com) |
| 09:36.01 | *** join/#asterisk Corydon76-home (i=two@pdpc/supporter/sustaining/Corydon76-home) |
| 09:36.01 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 09:43.58 | *** join/#asterisk dikdust (n=dikdust@gandalf.ipv6.adfacom.it) |
| 09:47.31 | *** join/#asterisk querix_support (n=tony_que@217.147.84.34) |
| 09:48.47 | querix_support | Hi there. Just a quick question as im a complete noob on this subject. Can asterisk support an ip fax system? |
| 09:52.35 | dikdust | querix_support, yes asterfax I guess |
| 09:57.42 | *** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 09:57.42 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.18, 1.4.3 (June 8, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 09:57.46 | *** join/#asterisk qdk (n=qdk@86.Red-81-44-151.dynamicIP.rima-tde.net) |
| 09:57.54 | *** part/#asterisk querix_support (n=tony_que@217.147.84.34) |
| 10:07.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:10.55 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 10:15.58 | *** join/#asterisk angryuser (n=Miranda@df01t2-212-195-106-226.d4.club-internet.fr) |
| 10:16.44 | angryuser | does * include by default QoS tags? (VPT or DSCP) ? |
| 10:18.05 | *** join/#asterisk Corydon76-home (i=black@pdpc/supporter/sustaining/Corydon76-home) |
| 10:18.05 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 10:20.24 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 10:22.11 | skirmisha | guys is it possible in asterisk to set account to accept many concurrent calls |
| 10:22.48 | tzafrir | skirmisha, yes. |
| 10:22.58 | tzafrir | If the other party supports that as well |
| 10:23.22 | skirmisha | how? |
| 10:23.30 | skirmisha | what do u need to set in asterisk config |
| 10:23.50 | skirmisha | i am talking about user that is registered with asterisk |
| 10:23.55 | skirmisha | not just forward calls |
| 10:25.33 | skirmisha | i tried with follow me like function |
| 10:25.47 | skirmisha | it works but caller id is not send in that case |
| 10:25.56 | skirmisha | as remote site is pbx system |
| 10:26.03 | *** join/#asterisk Dovid (n=Dovid@bzq-88-155-87-253.red.bezeqint.net) |
| 10:26.07 | skirmisha | pbx like panasonic |
| 10:26.15 | Dovid | hello ev1 |
| 10:26.28 | skirmisha | so i want to send to 1 port to pbx and pbx to do this "follow mw" |
| 10:26.53 | Dovid | anyone know of a solution for SIP -> H323 (not using asterisk).... |
| 10:27.11 | creativx | Dovid: #asterisk ?? |
| 10:27.43 | *** join/#asterisk Sycofant (n=Dylan@ip-58-28-134-73.ubs-dsl.xnet.co.nz) |
| 10:28.19 | skirmisha | Dovid yes i know |
| 10:28.26 | skirmisha | how much are u willing to pay for it |
| 10:31.10 | Dovid | depends |
| 10:32.18 | Dovid | brb phone |
| 10:36.35 | skirmisha | any ideas |
| 10:38.38 | mbranca | any g.729 license expert here ? I'm in trouble upgrading license count on digium codec :/ |
| 10:38.57 | Dovid | mbranca: what issues r u having ? |
| 10:39.18 | *** join/#asterisk alin` (n=user@193.226.173.50) |
| 10:39.22 | Dovid | skimicha: what solutions do u of ? |
| 10:40.07 | mbranca | I had a previous license with 30 channels. want to upgrade to 60 and bought an additional 30 channels license. now I have 2 license files in /var/lib/asterisk/license but the codec (latest version, asterisk 1.2.13) only shows up 30 channels |
| 10:40.20 | mbranca | instead of 60 |
| 10:40.37 | Dovid | mbranaca: it says that on the digium site. you can't put in and then do more - u need to call digium up for it |
| 10:43.06 | *** join/#asterisk achu (n=achu@61.17.217.185) |
| 10:43.07 | e-ddie | hrm |
| 10:43.16 | mbranca | Dovid, I don't agree |
| 10:43.17 | e-ddie | you should be able to add the license key to the other file... |
| 10:43.25 | mbranca | digium says is ok having multiple keys |
| 10:43.28 | mbranca | "Multiple G.729 keys may be registered to the same Asterisk server. This will allow |
| 10:43.28 | mbranca | <PROTECTED> |
| 10:43.28 | mbranca | <PROTECTED> |
| 10:43.38 | mbranca | on the kb : http://kb.digium.com/entry/30/5/ |
| 10:43.40 | e-ddie | hrm |
| 10:43.48 | Dovid | i stand to be corrected - i had issues with it b4 and digium helped me |
| 10:43.52 | e-ddie | we got two licenses on top of eachother, in different files |
| 10:44.00 | achu | When I get call from outside number the caller ID is showing like "unknown" |
| 10:44.17 | achu | it is happening from 3 days before and I don't do any changes to the system |
| 10:44.32 | Dovid | achu: it can be ur carrier |
| 10:44.39 | Dovid | is this inbound or outbound ? |
| 10:44.49 | achu | its inbound |
| 10:45.03 | achu | but I have connected two asterisk server using iax |
| 10:45.29 | achu | and when I call from the other server's extension it also shows the same "unknown" Caller ID |
| 10:46.30 | Dovid | and is the CID coming in on the first server ? |
| 10:46.38 | achu | the log shows like this : dialparties.agi: Caller ID name is 'est' number is 'unknown' |
| 10:46.47 | achu | its working good |
| 10:47.02 | Dovid | no upgrades, changes to anything ? |
| 10:47.07 | achu | nothing |
| 10:47.43 | achu | also I was facing with the issue of can't hear anything on my broadvoice sip lines |
| 10:47.45 | Dovid | highly unlikely that you did Nothing at all |
| 10:47.49 | achu | both incoming and outgoing |
| 10:47.54 | Dovid | ROFL |
| 10:48.00 | Dovid | ~YGWYPF |
| 10:48.08 | jbot | [ygwypf] You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 10:48.25 | Dovid | broadvoice is horrible - i had it for a month - i will never touch it again with a 10 foot pole |
| 10:48.39 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:48.57 | tzafrir | ~seen jbot |
| 10:49.14 | jbot | jbot <i=ibot@pdpc/supporter/active/TimRiker/bot/apt> was last seen on IRC in channel #debian, 601d 19h 33m 7s ago, saying: 'rumour has it, sarge is Ten-HUT! Fall in! Sarge is the code name for the current stable Debian release, version 3.1, released on June 6th, 2005. Ask me about <install debian>, or i guess sarge is the biggest lump of free as in ... |
| 10:49.14 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 10:49.33 | achu | Dovid, can you please tell me what are the things I have to check in the configurations ? |
| 10:50.21 | achu | I have checked zapata.conf and its looks like the default conifguration |
| 10:52.02 | achu | If I try an extension from the second server to the first server it shows the iax trunk name |
| 10:52.16 | achu | but beneath to it it shows unknown |
| 10:53.48 | achu | I think I had a kernel upgrade previous month |
| 10:54.00 | achu | it will affect it ? |
| 10:56.12 | achu | also when I try to run the command modprobe wctdm |
| 10:56.18 | achu | it returns errors |
| 10:56.31 | Dovid | achu: do u have a zaptel device ? |
| 10:56.37 | Dovid | or ztdummy ? |
| 10:56.37 | achu | no |
| 10:56.43 | achu | ztdummy |
| 10:56.51 | achu | after rebooting it |
| 10:56.54 | Dovid | also what version of zaptel r u using ? (this should not affect the CID) |
| 10:56.56 | achu | now there is no error |
| 10:56.56 | tzafrir | so zapata.conf is irrelevant |
| 10:57.02 | Dovid | should be |
| 10:57.09 | Dovid | but dosent hurt to load it properly |
| 10:57.12 | Dovid | and not wctdm |
| 10:57.17 | achu | k |
| 10:57.21 | Dovid | you need to also get the latest kernel headers |
| 10:57.32 | Dovid | and then rebuild latest version of zaptel |
| 10:57.57 | achu | any way I think I have to recompile everything |
| 10:57.59 | achu | hmm |
| 10:58.39 | achu | which files are affecting the caller ID ? |
| 11:00.47 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 11:00.52 | Dovid | should not be zaptel |
| 11:00.59 | Dovid | but like i said it dosent hurt to fix it |
| 11:01.06 | Dovid | what distro r u using ? |
| 11:01.12 | achu | centos 4.4 |
| 11:01.16 | Dovid | logout |
| 11:01.18 | Dovid | oops |
| 11:11.24 | alin` | how could I find the structure of a NOTIFY packet that is received by asterisk as response to a SUBSCRIBE? |
| 11:12.02 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:12.30 | Hymie | hmm, does anyone have the polycom working with the microbrowser? I have an url in the appropriate place, but hitting 'services' does nothing. Is there something else to enable? |
| 11:16.49 | alin` | how can I make asterisk to receive a NOTIFY message? |
| 11:17.40 | Hymie | asterisk, or a phone?? |
| 11:17.49 | Hymie | asterisk should just receive it from the phone automatically... |
| 11:19.42 | Hymie | sip notify 'something' extension |
| 11:19.52 | Hymie | if you want to send a notify of a type fo a phone |
| 11:19.56 | Hymie | but, what type of notify? |
| 11:20.08 | Hymie | I use it to reboot my phones |
| 11:20.15 | Hymie | via a script and te asterisk command interface |
| 11:21.32 | Hymie | and goes back tow ork fo ra bit |
| 11:21.55 | *** join/#asterisk bapril (n=bapril@pool-70-109-158-237.cncdnh.east.verizon.net) |
| 11:22.02 | alin` | Hymie: I want to debug a RECEIVED NOTIFY message |
| 11:22.22 | alin` | Hymie: a NOTIFY received after a SUBSCRIBE |
| 11:22.53 | alin` | Hymie: No I was reading on a book... |
| 11:22.57 | Hymie | alin`: not sure what you want to do.. |
| 11:23.15 | alin` | Hymie: I was not regarding the screen |
| 11:23.19 | Hymie | alin`: anyhow, notifies to asterisk usualyl come from external sip peers |
| 11:23.30 | Hymie | alin`: I don't know if you can send a fake one via the console |
| 11:23.39 | alin` | Hymie: my problem is to CAPTURE a notify message as reponse to a SUBSCRIBE |
| 11:23.41 | Hymie | alin`: you could probably set up two asterisk servers |
| 11:23.44 | Hymie | oh |
| 11:23.46 | Hymie | sip debug |
| 11:23.51 | Hymie | in the console |
| 11:24.02 | Hymie | also, you could use ngrep to see the packets directly |
| 11:24.19 | Hymie | ngrep is a linux utility |
| 11:24.22 | alin` | In this moment my asterisk is registered to a PBX, and there are 2 phones connected to my asterisk |
| 11:24.43 | alin` | I have already set `sip set debug' |
| 11:24.53 | tzanger | mornint |
| 11:24.58 | Hymie | so, you see the messages, what's do you want more? |
| 11:25.12 | alin` | I know ngrep. as tcpdump |
| 11:25.30 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 11:25.31 | Hymie | alin`: after sip debug, that's the only other way I know to get more info on the sip packets |
| 11:25.52 | Hymie | alin`: I know you can increase the sip debug level |
| 11:25.54 | alin` | Hymie: What of all that messages are NOTIFY, and, more, NOTIFY received as result to SUBSCRIBE |
| 11:26.12 | alin` | How can I increase it? |
| 11:26.22 | Hymie | alin`: not sure.. there is some way though |
| 11:26.32 | Hymie | alin`: I did it a few times, a while aback for serious debug |
| 11:27.04 | alin` | maybe asterisk -cvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv ? |
| 11:27.04 | jmls | hey guys |
| 11:27.29 | jmls | does anyone know what the "config" authorization is in the manager.conf ? |
| 11:27.38 | Hymie | alin`: I see you can also set debug for only one peer, might reduce the mesasges for you |
| 11:28.10 | Hymie | jmls: ? |
| 11:28.33 | alin` | yes, sip set debug peer <IP> |
| 11:28.57 | jmls | Hymie: in manager.conf there is "system,call,log,verbose" etc as classes that a connected manager can read / write |
| 11:29.05 | Hymie | jmls: I have no config statement anywhere in my manager.conf |
| 11:29.09 | jmls | I know all of 'em except config |
| 11:29.14 | Hymie | hmm |
| 11:29.15 | jmls | hmm. 1.4 |
| 11:29.16 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 11:29.20 | alin` | yes, sip set debug IP <IP> |
| 11:29.33 | Hymie | jmls: at one time, asterisk could write its own config file.. maybe it still can, and this is for that? |
| 11:29.37 | Hymie | dunno |
| 11:29.54 | tzafrir | Hymie, see the start of the sample extensions.cnf |
| 11:29.57 | Hymie | alin`: I know no more on this topic :( maybe soemone else does |
| 11:30.05 | tzafrir | this anly applies to extensions.conf |
| 11:30.25 | tzafrir | There are new manager commands in 1.4 to write config files from the manager |
| 11:30.28 | alin` | SUBSCRIBE in fact is the same as REGISTER? |
| 11:30.39 | jmls | tzafrir: ahhhhh |
| 11:30.50 | jmls | tzanger: what are the commands ? Any clues ? ;) |
| 11:30.56 | Hymie | hmm, does anyone have the polycom working with the microbrowser? I have an url in the appropriate place, but hitting 'services' does nothing. Is there something else to enable? |
| 11:31.23 | tzafrir | Hymie, what do you need to write? |
| 11:31.30 | tzanger | jmls: eh? |
| 11:31.41 | tzafrir | Maybe the asterisk database (astdb) would be more handy for that? |
| 11:31.50 | tzafrir | jmls? |
| 11:31.52 | jmls | oh dammnt. |
| 11:31.53 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 11:32.04 | tzanger | tzafrir: yeah I am actually *really* impressed with how the manager interface, and http manager specifically can read in a config ifle, alter it, and write it out WITHOUT losing comments or spacing |
| 11:32.08 | *** join/#asterisk the_5th_wheel (n=edd@kalfu.slipgate.za.net) |
| 11:32.20 | tzanger | tzafrir: I've been buried in the code for a while and I'm utterly impressed with that |
| 11:32.24 | jmls | tzafrir: what are the commands ? Any clues ? ;) |
| 11:32.40 | the_5th_wheel | is there an easy way to use a ftcto send smsses? |
| 11:32.40 | tzanger | although the weirdass "language" for passing fragments of the ifle back and forth baffle me, heh |
| 11:32.45 | tzanger | jmls: you'd have to look in the code |
| 11:32.53 | Hymie | jmls: the command is "help hymie get the polycom microbrowser working" ;P |
| 11:32.59 | tzanger | hahaha |
| 11:33.26 | *** join/#asterisk gardo (n=gardo@125.212.14.85) |
| 11:33.33 | tzafrir | tzanger, what about stuff like templates? or anything related to a '()' after the section name |
| 11:33.49 | tzanger | tzafrir: I am not sure I understand what you mean now |
| 11:34.01 | jmls | ooooh oooooh oooooh |
| 11:34.07 | jmls | show manager command UpdateConfig |
| 11:34.14 | tzafrir | 'show manager commands' . I don't have 1.4 handy now. GetConfig and PutConfig? |
| 11:34.17 | tzafrir | right |
| 11:35.38 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 11:37.43 | the_5th_wheel | is there an easy way to use a fixed cellular terminal/premicell to send smsses? |
| 11:37.54 | *** join/#asterisk Corydon76-home (i=gray@pdpc/supporter/sustaining/Corydon76-home) |
| 11:37.54 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 11:38.37 | Dovid | is it possible to install a proxy on a public IP b4 asterisk so that the RTP stream does not have to go through asterisk ? |
| 11:38.43 | Dovid | Proxy -> Asterisk |
| 11:39.23 | Dovid | (I know that there are VOIP switches that dont need to get in the way of the RTP stream even when one of the phones is behind NAT) |
| 11:48.17 | *** join/#asterisk tparcina (n=tparcina@cisco16.fesb.hr) |
| 11:49.32 | Hymie | is anyone using polycom phones here? |
| 11:49.58 | Hymie | I'm also curious if anyone has configured the 'directories' button to just go to the directory, intead of prompting for call lists |
| 11:52.35 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-79.ph.ph.cox.net) |
| 11:57.55 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 12:00.57 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 12:01.29 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 12:09.54 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:11.09 | *** join/#asterisk theglass (n=router@81-208-74-184.ip.fastwebnet.it) |
| 12:11.18 | theglass | hi |
| 12:11.30 | tzanger | ./nick theplate |
| 12:13.30 | theglass | "lol" |
| 12:14.34 | theglass | sorry if I might be off topic but do you know any pcap'ss support channel? |
| 12:14.41 | theglass | -z |
| 12:14.43 | theglass | -s* |
| 12:15.50 | HarryR | libpcap? I dont think there's one |
| 12:16.04 | HarryR | There is a mailing list iirc |
| 12:16.23 | theglass | uhm |
| 12:16.49 | *** join/#asterisk guillote_GNU (n=guillote@host176.201-252-205.telecom.net.ar) |
| 12:16.55 | *** join/#asterisk Tili (n=tili@203.170.74.236) |
| 12:21.02 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 12:21.16 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:21.40 | alin` | what is a `trunk' ? |
| 12:21.54 | Nuitari | ~trunk |
| 12:21.56 | jbot | trunk is, like, my trunk my trunk; my lovely asterisk trunk (check it out) |
| 12:23.11 | [TK]D-Fender | Oh God.... |
| 12:23.22 | [TK]D-Fender | alin`: Forget you ever heard that word. |
| 12:24.45 | alin` | [TK]D-Fender: Why ? |
| 12:25.15 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:25.19 | [TK]D-Fender | alin`: Its jsut a horribly misused word. If you've got a REAL question feel free to move on to it. |
| 12:30.32 | jacq | hey... any suggestion for a good conf system base don asterisk that takes acare of CNG and VAD? |
| 12:35.12 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 12:36.09 | tzanger | jacq: there is none |
| 12:36.16 | tzanger | asterisk does not support CNG and VAD at this time |
| 12:38.26 | *** join/#asterisk frk2 (n=fkhan@202.5.145.13) |
| 12:38.35 | frk2 | whatsup people |
| 12:38.42 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 12:38.43 | zeeesh | hi |
| 12:38.49 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 12:39.08 | *** join/#asterisk gardo (n=gardo@121.97.178.194) |
| 12:39.17 | frk2 | has anybody have their asterisk crash on them by pressing the 'submit' red bar in trixbox? |
| 12:39.17 | zeeesh | Can't locate LWP.pm in @INC (@INC contains: /usr/lib/perl5/site_perl/5.8.8/i386-linux-thread-multi? |
| 12:39.50 | frk2 | started happening with asterisk 1.2.18 |
| 12:41.17 | tzafrir | that can happen if you, e.g., add a zap extension with an invalid zap channel |
| 12:41.51 | tzafrir | http://bugs.digium.com/view.php?id=7290 |
| 12:42.20 | *** join/#asterisk Ciber311 (n=Ciber311@user-12ld42j.cable.mindspring.com) |
| 12:42.57 | [TK]D-Fender | frk2: .... |
| 12:43.00 | [TK]D-Fender | ~trixbox |
| 12:43.01 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 12:46.24 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 12:47.48 | jmls | if I have a manager session connected to asterisk, and want to read the "calls" events, is there anyway of filtering out the "newexten" events but keep all the others ? |
| 12:48.22 | jmls | I don't want to have to filter them out at each client session - it would be easier not to send them in the first place |
| 12:48.28 | alin` | what is the difference between registrations and subscriptions |
| 12:48.30 | alin` | ? |
| 12:49.27 | [TK]D-Fender | jmls: No. |
| 12:49.33 | jmls | bugger |
| 12:49.48 | creativx | jmls: take a look at astmanproxy |
| 12:49.58 | creativx | it can filter to some exten |
| 12:49.58 | creativx | t |
| 12:50.00 | jmls | yeah, astmanproxy often craps out on us |
| 12:50.08 | [TK]D-Fender | alin`: http://www.ietf.org/rfc/rfc3261.txt |
| 12:50.08 | jmls | (it just "freezes") |
| 12:50.11 | creativx | hmm |
| 12:50.23 | creativx | its stable here, but that might be because it only has 1 connection |
| 12:50.23 | creativx | :> |
| 12:50.35 | jmls | that's why I was wanting to limit the data being sent in the first place |
| 12:50.42 | jmls | we have 75+ connections ;) |
| 12:51.06 | creativx | still, its written in perl.. it should handle fine :P |
| 12:51.17 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:51.31 | creativx | but yes I understand your concern |
| 12:51.55 | creativx | are these 75+ connections coming from all over the place? |
| 12:51.58 | creativx | or only the local network |
| 12:53.08 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:53.49 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:59.45 | penguinFunk | hi, anyone know if digium have got a version of the TE120P with hardware echo cancellation? I can only find the 2 port cards with echo cancellation (TE207P or TE212P) |
| 13:00.18 | [TK]D-Fender | penguinFunk: www.digium.com |
| 13:00.25 | penguinFunk | seems a bit unfair you have to go to 2 ports when we will only ever use one here, just for echo cancellation :( |
| 13:00.41 | [TK]D-Fender | penguinFunk: But aside from the "could have found out faster by looking than asking", Sangoma makes one. |
| 13:00.42 | tzanger | penguinFunk: so go the ebay tellabs solution then, it's cheaper :-) |
| 13:01.14 | penguinFunk | [TK]D-Fender: i already looked before asking here. |
| 13:01.16 | denke | penguinFunk: I do not know about 1 port dsp cards... but compare the prices too |
| 13:01.31 | penguinFunk | okay guys will check out other people now |
| 13:01.33 | [TK]D-Fender | penguinFunk: Then you must really like the look of your own type-face ;) |
| 13:01.44 | penguinFunk | ive used digium cards before and know they work well |
| 13:01.59 | penguinFunk | [TK]D-Fender: yeah it's kind of sexy |
| 13:02.37 | denke | lol |
| 13:04.31 | *** join/#asterisk Curus (n=Curus@hd5b9080a.c45-01-12.sta.perspektivbredband.net) |
| 13:07.09 | penguinFunk | are sangoma / tellabs just as good as digium? |
| 13:08.37 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:08.37 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:08.41 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 13:09.51 | penguinFunk | or am i going to be sacrificing quality for cost? |
| 13:11.21 | [TK]D-Fender | penguinFunk: *pstt* |
| 13:15.23 | *** join/#asterisk gazonk (i=HMCFyYrQ@trillian.ASK.FH-Furtwangen.DE) |
| 13:17.08 | HarryR | penguinFunk: I'd say sangoma are better than the digium cards |
| 13:20.14 | Hymie | hmm, does anyone have the polycom working with the microbrowser? I have an url in the appropriate place, but hitting 'services' does nothing. Is there something else to enable? |
| 13:20.46 | *** part/#asterisk tparcina (n=tparcina@cisco16.fesb.hr) |
| 13:22.41 | [TK]D-Fender | Hymie: What model / firmware? |
| 13:22.43 | frk2 | [TK]D-Fender: sorry was distracted |
| 13:23.01 | *** join/#asterisk exazoid (n=chatzill@0x3e42eeca.adsl.cybercity.dk) |
| 13:23.12 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:23.24 | Hymie | [TK]D-Fender: 501... hmm, sec on the firmware (although, there is an option in the web console for the microbrowser stuff...) |
| 13:23.46 | exazoid | hi. Has anyone a recommandation for a brandname (HP, IBM) server for a TE120P in the 800-1000$ range ? |
| 13:23.58 | [TK]D-Fender | Hymie: You need 2.1.0 + for it. |
| 13:24.03 | exazoid | I am looking at a HP ML115 opteron based system |
| 13:24.08 | Hymie | it says sip is 2.0.3 |
| 13:24.13 | Hymie | so I guess that's why |
| 13:24.16 | [TK]D-Fender | Hymie: yup |
| 13:24.21 | Hymie | thanks |
| 13:24.22 | Hymie | btw |
| 13:24.30 | Hymie | did you ever get the 'direcotires' button to go directly to the directory |
| 13:24.36 | Hymie | instead of the directories sub-menu |
| 13:24.53 | Hymie | there's a call lists button below it, it's drives me mad that there's an optino to entre the call lists menu from the directories button |
| 13:24.55 | [TK]D-Fender | Hymie: Nope, never tried to. |
| 13:25.07 | Hymie | thanks |
| 13:25.20 | [TK]D-Fender | Hymie: Don't believe I saw anything that hinted that this would be possible. |
| 13:25.29 | [TK]D-Fender | Hymie: Who are you putting in that list? |
| 13:25.46 | Hymie | me either, yes ;( hmm, do you know if you can downgrade the sip app if you upgrade and are unhappy for other reasons? |
| 13:25.52 | Hymie | [TK]D-Fender: everyone in the company |
| 13:25.53 | frk2 | guys I hate trixbox myself :P clients love it though :) |
| 13:26.59 | mosty | frk2, i've never tried it, what's it like? what do you hate about it? |
| 13:27.29 | exoxe | I'm baaaaaaaaaack |
| 13:27.34 | frk2 | [TK]D-Fender: will try the trixbox channel. It crashes asterisk 1.2.18 when trying to do a bulk update through AMI for some reason |
| 13:27.42 | frk2 | mosty- its a LOT of bloat |
| 13:28.08 | mosty | frk2, does it require a powerful box? |
| 13:28.43 | frk2 | its just a php app |
| 13:28.48 | [TK]D-Fender | Hymie: You can always up/down the SIP app, its the BR you can't go too far back on. |
| 13:28.56 | frk2 | do you know about freepbx? |
| 13:28.58 | mosty | frk2, then what do you mean by bloat? |
| 13:30.05 | alin` | [TK]D-Fender: thanks |
| 13:30.07 | Hymie | [TK]D-Fender: cool |
| 13:30.09 | *** part/#asterisk achu (n=achu@61.17.217.185) |
| 13:31.03 | [TK]D-Fender | Hymie: And not sure if I told you, but browsing through a phone directory for employees is BS on any phone. a plain sheet of paper = 100x better & faster. |
| 13:31.44 | [TK]D-Fender | Hymie: Or a click-to-call web script. |
| 13:31.53 | frk2 | [TK]D-Fender: even worse is a WWW browser on those ciscos! |
| 13:31.58 | frk2 | in blazing full color |
| 13:32.14 | [TK]D-Fender | frk2: I'm sure ti could be very useful. |
| 13:32.31 | frk2 | what? like how? |
| 13:32.43 | Hymie | [TK]D-Fender: yes, I know.. but, management says "why aren't we using this" ;Þ |
| 13:32.57 | Hymie | [TK]D-Fender: likely the paper will be used, but the directory will be there just in case |
| 13:33.04 | Polis_ttt | what command can i use in asterisk-cli to kick a sip-user that's not supposed to be there? |
| 13:33.25 | [TK]D-Fender | frk2: ethernet video camera at reception so you can see who's at the door before letting them in. full-colour operator panel view of PBX activity, etc. |
| 13:33.30 | *** join/#asterisk _DAW (n=chatzill@adsl-156-72-8.msy.bellsouth.net) |
| 13:33.36 | [TK]D-Fender | frk2: Would *I* pay for it howevre... no. |
| 13:33.57 | frk2 | you have a PC to do that, better resolution AND control |
| 13:34.11 | [TK]D-Fender | Polis_ttt: whree is "there", and in what state |
| 13:34.21 | [TK]D-Fender | frk2: Entirely true. |
| 13:34.47 | [TK]D-Fender | frk2: Depends on certain conveniences. I could hardly justify it personally. |
| 13:34.58 | frk2 | nah nobody can |
| 13:35.01 | Polis_ttt | [TK]D-Fender: they are in sweden, on my server, they got an account, but shall not call on the server at this moment, so i just want to get them to disconnect |
| 13:35.23 | *** join/#asterisk lenne_dk (n=leif@cpe.atm2-0-74391.0x535cc77e.hknxx4.customer.tele.dk) |
| 13:35.29 | [TK]D-Fender | Polis_ttt: go change their password on them. |
| 13:35.57 | Polis_ttt | [TK]D-Fender: can i do that when they are logged in, and they get disconnected directly? |
| 13:36.47 | [TK]D-Fender | Polis_ttt: calls will reject, and then when their reg timeout takes place they'll see that they're toast |
| 13:36.57 | Polis_ttt | [TK]D-Fender: thanks a lot |
| 13:37.03 | lenne_dk | Hi. GXW4104 (Grandstream FXO) not accepting calls after (automatic) fw-upgrade to 1.0.0.55 |
| 13:37.09 | [TK]D-Fender | ~gs |
| 13:37.10 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 13:37.11 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 13:37.22 | [TK]D-Fender | </bile> |
| 13:37.55 | lenne_dk | I have 1.0.0.48, and a tftp server, but should I really put the files in /tftpboot? |
| 13:37.56 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 13:42.32 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 13:47.51 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:48.35 | *** join/#asterisk brussel_ (n=brussel@adsl-71-154-207-24.dsl.sndg02.sbcglobal.net) |
| 13:48.50 | lenne_dk | Hello... tftpboot grandstream help... |
| 13:49.40 | mosty | lenne_dk, the short answer is probably "we don't know, we avoid grandstream phones at all cost" |
| 13:50.01 | mosty | try downgrading the firmware. it's not like you can make it worse |
| 13:50.19 | *** join/#asterisk bluedemon (n=belamark@merlintechs.kvinet.com) |
| 13:50.48 | bluedemon | Has anyone here had experiance with the snom 360? |
| 13:51.04 | mosty | bluedemon, a little. i mostly use 320's |
| 13:52.48 | bluedemon | Is there anyone way to have the display recoginise a newline char when sending an snomipphonetext xml object? |
| 13:55.02 | penguinFunk | rofl mosty |
| 13:55.37 | Dovid | anyone know the URL to edgemark ? |
| 13:56.14 | mosty | bluedemon, not sure, i've never played with that |
| 13:57.05 | alin` | can somebody tell me how can I set asterisk in order that it send SUBSCRIBE sip messages? |
| 13:57.18 | alin` | I need to see the reponse from subscribe. |
| 13:57.44 | bluedemon | mostly, thanks anyways. The documentation on these things are lacking. |
| 13:57.47 | alin` | I put breakpoints in the functions that receive/send notify for subscribe, but the br. are not reached |
| 13:57.47 | [TK]D-Fender | alin`: * doesn't SEND them. You'd have to invent a lot to do this. |
| 13:58.40 | alin` | ok, in order that asterisk subscribe somewhere |
| 14:00.23 | lenne_dk | I rtfl (read the fine logs) and saw where tftpd expected files. Put them there, rebooted, GXW grabbed the files, and now it accepts calls again. |
| 14:00.42 | [TK]D-Fender | alin`: Again, * does not do this. You have a lot of coding to do. |
| 14:01.28 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:01.59 | lenne_dk | doesn't * SUBSCRIBE when it "signs in" to another * to receive calls? |
| 14:03.00 | file | SUBSCRIBE is when you want to get the status of another device, or the status of a mailbox |
| 14:03.25 | lenne_dk | Sorry... |
| 14:08.13 | alin` | [TK]D-Fender: YOU MEAN THAT ASTERISK DOES NOT SEND SUBSCRIBE IN NO SITUATION? ARE YOU SURE !? I lost a day searching for this today. |
| 14:08.38 | alin` | [TK]D-Fender: ASTERISK DOES NOT SUPPORT SUBSCRIBE !? |
| 14:09.05 | file | it supports receiving SUBSCRIBE requests, but it does not send them |
| 14:09.21 | [TK]D-Fender | alin`: it ANSWERS requests. It does not SEND them. |
| 14:09.49 | [TK]D-Fender | alin`: Sip devices can subscribe for Presence, or VM. thats it. * does not go looking for info from anything. |
| 14:10.57 | alin` | [TK]D-Fender: And how * answers when it receives a SUBSCRIBE? |
| 14:11.29 | [TK]D-Fender | alin`: Go set up a sip phone and try for yourself. |
| 14:12.06 | alin` | [TK]D-Fender: I have 2 SIP phones set up with my * |
| 14:12.48 | [TK]D-Fender | alin`: Fine. Go set up a VM box for them and have them subscribe to a dialplan hint for presence and jsut watcht he SIP debug roll by |
| 14:13.10 | alin` | VM = voice mail? |
| 14:13.22 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@www.airlinksystems.com) |
| 14:13.25 | [TK]D-Fender | alin`: Yes |
| 14:13.35 | mosty | alin: run tethereal/tshark and watch what it does |
| 14:14.00 | alin` | [TK]D-Fender: thanks |
| 14:14.13 | *** join/#asterisk ManxPower (n=manxpowe@98.sub-75-200-21.myvzw.com) |
| 14:14.22 | [TK]D-Fender | mosty: * SIP debug is free and 1/2 second away pre-decoded.... |
| 14:14.37 | [TK]D-Fender | mosty: but Wireshark is better graphically. |
| 14:15.51 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 14:15.56 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:15.59 | Nuitari | Hi [TK]D-Fender |
| 14:19.02 | mosty | [TK]D-Fender, i like tshark for a quick summary |
| 14:19.29 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:23.32 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:24.33 | alin` | [TK]D-Fender: I called 555 from such a Phone (snom 360) |
| 14:24.34 | alin` | , and I see no SUBSCRIBE in its log. Look extensions.conf: |
| 14:24.34 | alin` | exten => 555,1,Voicemailmain(s0@default) |
| 14:24.51 | *** join/#asterisk seele_ (n=seele@200.30.85.186) |
| 14:25.05 | alin` | What shell I do to see a subscribe/notify :( ? |
| 14:26.57 | seele_ | I have this error http://www.pastebin.ca/566558 in the user.log how can I solve this?? |
| 14:27.58 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 14:29.02 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:29.50 | codefreeze | seele_: fix the agi script; that's where your error is coming from, CALLTRACE_HUNT, something is wrong there. |
| 14:29.52 | [TK]D-Fender | alin`>exten => 555,1,Voicemailmain(s0@default) <- this means NOTHING. |
| 14:30.18 | [TK]D-Fender | alin`: * dialplan has nothing to do with SUBSCRIBE. You need to set the phone's SIP entry to MONITOR the mailbox. |
| 14:31.02 | Nuitari | a lot of people have problem seeing the line between devices and dialplan |
| 14:31.48 | alin` | [TK]D-Fender: Thanks |
| 14:31.59 | [TK]D-Fender | Nuitari: Yeah.... like every guy calling a SIP device an extensions. Ticks me off. |
| 14:32.35 | [TK]D-Fender | alin`: And set up a hint to watch another device hand have your phone subscribe to it. |
| 14:34.12 | alin` | [TK]D-Fender: #-o |
| 14:35.53 | *** join/#asterisk nasls_lsa (n=chatzill@dsl-88-218-29-27.customers.vivodi.gr) |
| 14:36.04 | *** part/#asterisk nasls_lsa (n=chatzill@dsl-88-218-29-27.customers.vivodi.gr) |
| 14:39.07 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:42.35 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-97cb1510c3a0f74c) |
| 14:42.35 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:45.15 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 14:45.48 | [[blah]asfd | If anyone is interested, I am selling a TE405P and an Adit 600 channel bank. If you are interested PM me. |
| 14:45.49 | *** join/#asterisk MihiNomenEst (i=LBVH@cerebus.clandestineresearch.com) |
| 14:47.09 | *** join/#asterisk drewr (n=drew@pdpc/supporter/active/drewr) |
| 14:47.26 | frk2 | [TK]D-Fender - think ive nailed the problem down, dont seem to be freepbx/trixbox related (atleast initially) |
| 14:47.49 | frk2 | asterisk crashes with this as the last debug line: Jun 14 19:44:34 DEBUG[20064] app_macro.c: Executed application: (null) |
| 14:48.09 | frk2 | happens right after a RELOAD has been triggered |
| 14:48.12 | frk2 | any ideas? |
| 14:48.20 | frk2 | happens randomly |
| 14:49.23 | frk2 | i dont know HOW the application can be 'null' first of all |
| 14:51.16 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 14:52.00 | *** join/#asterisk roselineX (n=x@MTLXPQAK-1177996026.sdsl.bell.ca) |
| 14:52.44 | roselineX | Hi. I setup a trunk in asterisk using the SIP information from my provider. That provider gives me 2 virtual numbers on the same account. Can asterisk recognize which number was dialed on an incoming call? |
| 14:53.30 | volker__ | if a number is called, i want that a dtmf tone is transmittet after the called person picks up. how do i handle it? if i just use foo,1,dial() and foo,2,SendDTMF() the dtmf sending wont work till the call isnt finished |
| 14:53.40 | mosty | roselineX, depends how the server is sending the calls to you |
| 14:54.05 | alin` | [[blah]asfd: I go. Thanks for help. Tomorrow maybe I suceed to set up the phone. |
| 14:54.20 | alin` | in order to see SUBSCRIBE messages... |
| 14:54.22 | [[blah]asfd | what did i help with |
| 14:54.45 | penguinFunk | chica baw waw! |
| 14:54.52 | roselineX | mosty: thanks, how can i check for that? |
| 14:55.24 | mosty | roselineX, are your calls coming in to the s extension, or a number? |
| 14:55.39 | roselineX | s |
| 14:55.56 | mosty | then no, the sip server is not sending you the dialled number information |
| 14:55.57 | roselineX | Then I believe it's not possible |
| 14:56.07 | roselineX | Yeah I should've thought twice before asking =) |
| 14:56.10 | roselineX | Thank you :) |
| 14:56.17 | mosty | no problem |
| 14:58.05 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:58.56 | [TK]D-Fender | volker__: "show application dial" |
| 15:00.24 | exoxe | hah, found typo in TFOT, i win |
| 15:01.12 | exoxe | GotoIf($[{$TEST} = 1]?10:20) |
| 15:02.26 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 15:04.39 | volker__ | [TK]D-Fender: thanks. i think u mean M(). had a bad documentation which didnt showed all options :( |
| 15:06.05 | *** join/#asterisk saftsack (n=oliver@p54A7CBE2.dip.t-dialin.net) |
| 15:09.36 | exoxe | get a job sparkling wiggles |
| 15:10.01 | [TK]D-Fender | volker__: * seem to have it BUILT IN. You should read more. |
| 15:10.36 | [TK]D-Fender | exoxe: Enjoy your victory. the next edition is due in about a month :) |
| 15:11.00 | exoxe | damn :( |
| 15:11.40 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 15:12.05 | volker__ | [TK]D-Fender: rightl. D() |
| 15:12.39 | seele_ | please help I have a big problem, when I call to other estension or to zap channel the other person can listen me very well but I listen cuts in the conversation |
| 15:12.40 | exoxe | I forget.. what does u and/or b in front of an extension mean? such as VoiceMail(b101) |
| 15:12.44 | seele_ | how can I solve this?? |
| 15:12.55 | exoxe | I'm having a hard time looking it up since searching for 'u' or 'b' well.. |
| 15:13.29 | seele_ | I'm trying different codecs but the problem continues |
| 15:14.09 | exoxe | oh nevermind I understand |
| 15:14.19 | exoxe | u=navailable, b=usy I gander |
| 15:14.56 | [TK]D-Fender | exoxe: .... "show application voicemail" :) |
| 15:15.58 | *** join/#asterisk btsteve (n=btsteve@204.10.20.30) |
| 15:17.16 | mosty | seele_, that means your download (or the other ends upload) is crap |
| 15:17.32 | drewr | I'm trying to diagnose random CHANUNAVAIL ("Everyone is busy/congested at this time") problems while dialing out. |
| 15:17.45 | seele_ | mosty, how can I test this , any tool for this |
| 15:17.54 | drewr | Usually one call can be in progress, but the second fails. And it doesn't happen all the time. |
| 15:17.58 | mosty | drewr, do you have qualify=yes in sip.conf ? what is the peer's status? |
| 15:18.00 | seele_ | only the download .... in my internal netwotk |
| 15:18.03 | drewr | Does this suggest a hardware issue? |
| 15:18.08 | volker__ | [TK]D-Fender: but it seems that it cant handle ${EXTEND} in it |
| 15:18.25 | [TK]D-Fender | volker__: SHOW ME. |
| 15:18.29 | [TK]D-Fender | ~pb |
| 15:18.29 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 15:18.30 | mosty | seele_, is it a busy network? |
| 15:18.32 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 15:18.40 | seele_ | how can i test the latency or the broadcast level in my switch |
| 15:18.50 | drewr | mosty: No, I don't have that. |
| 15:18.57 | [TK]D-Fender | drewr: that error can mean anything. pastebin the full CLI output of the failed call at verbose 10 |
| 15:19.12 | mosty | drewr, well add that, then do "sip show peer <name>" |
| 15:19.18 | seele_ | mosty, is a new network with 180 phones aprox in a different vlan |
| 15:19.29 | volker__ | [TK]D-Fender: nevermind. i wrote it wrong in the channel and wrong in the config. its ${EXTEN} without D %-> |
| 15:19.42 | mosty | seele_, do your routers support QoS? |
| 15:19.55 | [TK]D-Fender | volker__: GUILTY! |
| 15:20.12 | seele_ | mosty, I don't use any router |
| 15:20.22 | seele_ | mosty, only switch |
| 15:20.34 | _DAW | Any recommendations on a good 24 port FXS gateway with Asterisk? |
| 15:20.48 | frk2 | quintum |
| 15:20.53 | [TK]D-Fender | _DAW: Mediatrix 1124 |
| 15:20.53 | frk2 | audiocodecs |
| 15:21.10 | _DAW | Thanks |
| 15:21.28 | [TK]D-Fender | _DAW: AudioCodes os good, but rather complex. |
| 15:21.32 | Qwell[] | [TK]D-Fender: Is it just me, or does mediatrix sound like...something...wrong |
| 15:21.48 | seele_ | mosty, other suggest ... switches change??? |
| 15:21.59 | [TK]D-Fender | Qwell[]: they had their name before its fuity-ness was ursurped ;) |
| 15:22.06 | Qwell[] | no, something else |
| 15:22.12 | Qwell[] | word also ends in trix :p |
| 15:22.48 | mosty | seele_, it appears that you have packet loss in one direction. is this a busy network? |
| 15:23.31 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 15:23.32 | seele_ | mosty, define busy network |
| 15:23.39 | mosty | seele_, lots of traffic |
| 15:23.41 | seele_ | mosty, only phones and asterisk |
| 15:23.48 | *** join/#asterisk sethtrain (n=seth@nsc69.38.115-233.newsouth.net) |
| 15:24.15 | mosty | seele_, even in other vlans that use the same switches? |
| 15:24.23 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:25.05 | drewr | [TK]D-Fender: Is verbosity 10 supposed to show more about the call than 3? :-) |
| 15:25.09 | seele_ | mosty, no only the phones .... but the switch are interconnected |
| 15:25.14 | drewr | Gave me the same amount. |
| 15:25.32 | Mercestes | drewr: try 99 |
| 15:26.08 | *** join/#asterisk saftsack (n=oliver@p54A7FB43.dip.t-dialin.net) |
| 15:26.33 | mosty | do you have two phones on the same switch as asterisk? |
| 15:26.48 | seele_ | mosty, yes |
| 15:27.04 | mosty | what's the call sound like between those two phones? |
| 15:27.20 | seele_ | mosty, no wait ... let me test |
| 15:27.20 | drewr | Mercestes: Nope, same amount of information. |
| 15:27.43 | Mercestes | drewr: try set verbose 3.628^1024 |
| 15:28.20 | drewr | Mercestes: Kernel panic. |
| 15:28.24 | [TK]D-Fender | drewr: So for all this talk about how much it gave you... why have you still shown me NOTHING? :) |
| 15:28.27 | Mercestes | Yea, that's the max then. |
| 15:28.27 | drewr | Mercestes: :-) |
| 15:28.27 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 15:28.47 | drewr | [TK]D-Fender: Good point. |
| 15:29.25 | frk2 | mediatrix is expensive |
| 15:29.31 | frk2 | for apparently no reason |
| 15:30.26 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 15:30.27 | frk2 | [TK]D-Fender: figured my crash out. FYI 1.2.18 has a serious problem: |
| 15:30.35 | frk2 | http://bugs.digium.com/view.php?id=9602&nbn=15 |
| 15:30.46 | drewr | [TK]D-Fender: http://paste.lisp.org/display/42767 |
| 15:32.21 | [TK]D-Fender | drewr: - Executing Dial("SIP/208-55a3", "Zap/27/8374509") in new stack |
| 15:32.27 | [TK]D-Fender | -- Zap/27-1 answered SIP/208-55a3 |
| 15:32.44 | [TK]D-Fender | -- Executing Dial("Zap/10-1", "Zap/27/5017325") in new stack |
| 15:32.46 | [TK]D-Fender | Jun 14 10:10:29 NOTICE[12881]: app_dial.c:749 dial_exec: Unable to create channel of type 'Zap' |
| 15:32.47 | [TK]D-Fender | <PROTECTED> |
| 15:33.02 | [TK]D-Fender | drewr: SIP/208 is on that channel and was TALKING ON IT |
| 15:33.13 | drewr | Yikes. |
| 15:33.19 | Qwell[] | Nub question of the day! |
| 15:33.25 | [TK]D-Fender | SCHMUCK :) |
| 15:33.34 | Qwell[] | What is a DSU/CSU, exactly? Don't expand the acronyms, it won't help. :P |
| 15:33.40 | mosty | drewr, are you dialing with channel groups or explicit channel numbers? |
| 15:33.57 | [TK]D-Fender | mosty: Explicit and clearly busy |
| 15:34.10 | [TK]D-Fender | mosty: Problem fully diagnosed, you may disengage :) |
| 15:34.25 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:34.52 | mosty | [TK]D-Fender, i agree. but perhaps they're trying to dial using explicit channels when they want to be using channel groups |
| 15:35.09 | *** join/#asterisk ivanfm (n=ivanfm@c934f322.virtua.com.br) |
| 15:35.53 | [TK]D-Fender | mosty: Nobody puts Zap/27 and hopes its a GROUP :) Show them the problem and let them consider if it really meets their needs. I try not to GUESS if they wanted that channel or not. |
| 15:35.57 | _VoiceMeUp_COM | hmm i see it lol |
| 15:36.06 | _VoiceMeUp_COM | Executing Dial("Zap/10-1", "Zap/27/5017325") in new stack |
| 15:36.09 | _VoiceMeUp_COM | theres a zap too many |
| 15:36.13 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 15:36.33 | frk2 | so im guessing NOBODY is running 1.2.18 |
| 15:36.34 | [TK]D-Fender | mosty: Otherwise I could suddenly start suggesting wild shit like dumping TDM cards and going pure VoIP via smoke-signals! ;) |
| 15:36.39 | frk2 | since nobody cares about the blaring bug |
| 15:36.50 | _VoiceMeUp_COM | blaring ? |
| 15:36.54 | frk2 | Your servers will be DOOMED |
| 15:36.54 | _VoiceMeUp_COM | wth is that |
| 15:36.56 | frk2 | :) |
| 15:36.57 | _VoiceMeUp_COM | lol |
| 15:37.01 | [TK]D-Fender | frk2: We're just not unloading zaptel all the time. |
| 15:37.02 | _VoiceMeUp_COM | mem los been reported |
| 15:37.10 | [TK]D-Fender | frk2: like fruiy-asses Trixbox does :) |
| 15:37.11 | Nuitari | frk2: I'm running 1.4.4 and trunk |
| 15:37.13 | seele_ | any way to measure the broadcast level or the congestion level?? |
| 15:37.16 | mosty | [TK]D-Fender, semaphore is clearly superior to smoke signals |
| 15:37.30 | sethtrain | [TK]D-Fender: I work with drewr, how wold you recommend fixing this issue? |
| 15:37.38 | frk2 | hey man |
| 15:37.48 | [TK]D-Fender | mosty: Telegraph > both, and competes with teletype in the hands of a pro :) |
| 15:38.01 | seele_ | differently of look the switch LEDs |
| 15:38.04 | frk2 | im serious |
| 15:38.09 | frk2 | is nobody running 1.2.18? |
| 15:38.16 | [TK]D-Fender | sethtrain: I don't see your full dialplan nor do I know what you're attempting to accomplish. |
| 15:38.33 | jer | frk2, yes |
| 15:38.34 | _VoiceMeUp_COM | wahts the prob frk |
| 15:38.54 | mosty | seele_, depends how much you paid for the switch, i guess |
| 15:38.54 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 15:38.57 | [TK]D-Fender | sethtrain: And again it has not been clearly labeled as an "issue". It may be a CONSEQUENCE of a particular mode of operation that your DECIDED upon. |
| 15:38.58 | frk2 | nothing. i hate applying patches |
| 15:39.08 | sethtrain | [TK]D-Fender: yeah. |
| 15:39.09 | [TK]D-Fender | frk2: Go cold-turkey! |
| 15:39.30 | seele_ | nortel 2526T |
| 15:39.31 | frk2 | http://bugs.digium.com/view.php?id=9602&nbn=15 |
| 15:39.33 | frk2 | that |
| 15:39.37 | frk2 | its a bug in app macro |
| 15:39.41 | [TK]D-Fender | sethtrain: He aksed WHY, and I answered. I did not say this was wrong... thats for you to tell me :) |
| 15:39.46 | frk2 | causes asterisk to randomly seg fault |
| 15:40.09 | NOT_guru | Question: which version of chan_sccp are people running? latest one I see is April 8th 2006 |
| 15:40.38 | drewr | [TK]D-Fender: :-) We inherited this system, and it usually works OK. We're just trying to figure out why this periodically happens. |
| 15:40.56 | drewr | If Asterisk isn't allocated channels properly, it seems like it would work a lot less than it does. |
| 15:40.58 | frk2 | chan_sccp is evil. newer CCMs are sip based anyways. |
| 15:41.12 | NOT_guru | Fender: FYI you were helping me with my TDM4XX card and the FXS module, well yes the FXS module was indeed dead, I got 2 FXO modules in the mail since then and they worked perfectly... thanks for your assistance |
| 15:41.23 | *** join/#asterisk sunsmasher (n=Beamer@66.251.47.154) |
| 15:41.24 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:41.26 | galeras | frk2 that bug was corrected on latest svn |
| 15:41.34 | [TK]D-Fender | drewr: You have something in your dialplan that specifically chooses that single Zap channel, and 1 person tried it while it was already in use. The outcome is entirely predictable. |
| 15:41.45 | [TK]D-Fender | NOT_guru: np |
| 15:42.01 | frk2 | should i go to asterisk 1.4? |
| 15:42.08 | galeras | it is |
| 15:42.08 | [TK]D-Fender | the only good CCM ...... is HOCKEY SKATES. |
| 15:42.09 | NOT_guru | frk2: yes I run my 7940's and 7960's in SIP mode now, but someone here was raving about the chan_sccp and I thought I would give it a try |
| 15:42.12 | Qwell[] | frk2: chan_skinny is FAR FAR better in 1.4 |
| 15:42.21 | Qwell[] | chan_sccp is dead - don't use it |
| 15:42.40 | NOT_guru | thanks Qwell I appreciate the reply |
| 15:42.55 | Qwell[] | there has been quite a lot of work on chan_skinny in the last few weeks to, thanks to DEA, pj, mvanbaak, and a few others |
| 15:42.58 | jer | [TK]D-Fender, pfft; bauer makes a better hockey skate for many types of players |
| 15:43.08 | Qwell[] | too* |
| 15:43.09 | NOT_guru | so if I wanted to try a more native cisco channel what would be the advised path? |
| 15:43.10 | jer | ccms are good for kids who grow quickly |
| 15:43.18 | exoxe | sometimes I hear SayNumber().. sometimes I don't... |
| 15:43.22 | frk2 | galeras thinking of doing that. i hate running SVN systems in production though |
| 15:43.39 | frk2 | trunk i mean |
| 15:43.56 | exoxe | I hope you've got a big trunk... cuz I'm gonna stick my bike in it |
| 15:47.08 | galeras | frk2: i have no choice, i had a lot * crashes |
| 15:47.19 | _VoiceMeUp_COM | can we have vm mail groups for directory |
| 15:47.41 | frk2 | im thinking of going BACK to 1.2.16 |
| 15:47.42 | galeras | i was running my production with latest svn for 2 days fine |
| 15:48.06 | jer | 2 days? not exactly latest svn then eh? |
| 15:48.15 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 15:48.33 | galeras | right, 2_days_old svn |
| 15:50.09 | *** join/#asterisk CunningPike (n=CunningP@204.239.12.183) |
| 15:50.32 | frk2 | i think i'll need to do the same |
| 15:50.45 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:51.28 | NOT_guru | so is 1.2.16 considered the most stable revision for production systems? |
| 15:51.35 | sethtrain | [TK]D-Fender: http://pastie.caboo.se/70461 - here is how are channels are set up and with the little bit of asterisk knowledge I think the first fail safe tries the 1,Dial(${INTTRUNK}/${LOCALAC}${NUMBER:1}) which is using our Zap/G1 which looks like should be channesl 25-32 |
| 15:52.10 | *** join/#asterisk guillote_GNU (n=guillote@190.7.30.134) |
| 15:52.26 | ManxPower | NOT_guru: I use 1.2.15 |
| 15:52.28 | _VoiceMeUp_COM | doh |
| 15:52.34 | _VoiceMeUp_COM | just saw context mappings in vm |
| 15:52.35 | _VoiceMeUp_COM | ok |
| 15:52.40 | _VoiceMeUp_COM | yeah BTW |
| 15:52.45 | _VoiceMeUp_COM | had a REALLY GOOD question |
| 15:52.51 | frk2 | im thinking of going back to 1.2.15/16 |
| 15:53.01 | frk2 | this is gay- asterisk dying for no reason |
| 15:53.04 | _VoiceMeUp_COM | WTF is asterisk doing a CORE .xxxx on EVERY CALL but doesnt crash dump or anything ? |
| 15:53.38 | _VoiceMeUp_COM | co z of -g flag ? |
| 15:54.39 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 15:55.39 | ManxPower | frk2: and that is EXACTLY why we use 1.2.15 |
| 15:56.01 | NOT_guru | well bummer I just rolled my new system and went straight to 1.2.18 |
| 15:56.26 | NOT_guru | then again its a new system so I could always rebuild and push my test box back in place |
| 15:56.27 | NOT_guru | its a VM |
| 15:56.28 | ManxPower | NOT_guru: if you don't use macros, I don't think you'll trigger the bug |
| 15:56.42 | frk2 | yeah |
| 15:56.43 | NOT_guru | I am not that far along into asterisk |
| 15:56.44 | frk2 | if you do |
| 15:56.50 | NOT_guru | so I am not using macros ;) |
| 15:56.53 | frk2 | you are in for a royal ass kicking |
| 15:56.59 | frk2 | asterisk will fall |
| 15:57.08 | NOT_guru | well not YET at least |
| 15:57.08 | frk2 | like i did today |
| 15:57.16 | frk2 | 3 separate installations |
| 15:57.17 | frk2 | crashed |
| 15:57.22 | frk2 | all of them |
| 15:57.25 | frk2 | so stupid |
| 15:57.30 | _VoiceMeUp_COM | weklk |
| 15:57.41 | _VoiceMeUp_COM | well we have some issues sometime.. but 17 seems ok |
| 15:57.43 | ManxPower | frk2: Welcome to my world. |
| 15:57.49 | _VoiceMeUp_COM | except over 2c/s |
| 15:57.49 | ManxPower | But I only upgrade 1 PBX as a time |
| 15:57.59 | _VoiceMeUp_COM | and except over 140 channels |
| 15:57.59 | _VoiceMeUp_COM | ;) |
| 15:58.26 | _VoiceMeUp_COM | that why ser is the key to our problems.. actually not sure if soemone told everyone htat asterisk is a PBX.. |
| 15:58.32 | _VoiceMeUp_COM | that can do routing to a point |
| 15:58.50 | ManxPower | Asterisk isn't REALLY a PBX. |
| 15:58.53 | _VoiceMeUp_COM | and ser is a router that can to some pbx function lol |
| 15:58.54 | _VoiceMeUp_COM | well |
| 15:58.57 | _VoiceMeUp_COM | what you call it ? |
| 15:59.02 | ManxPower | It is a PBX TOOLKIT that lets you BUILD a PBX. |
| 15:59.03 | _VoiceMeUp_COM | an swiss army knofe |
| 15:59.05 | _VoiceMeUp_COM | knofe |
| 15:59.07 | _VoiceMeUp_COM | yeah |
| 15:59.09 | _VoiceMeUp_COM | TRUE |
| 15:59.15 | _VoiceMeUp_COM | trixbox is a POSPBX |
| 15:59.23 | _VoiceMeUp_COM | and this aint no point of sales talk |
| 15:59.25 | file | and if you don't isolate the potential issues... test.. do things properly, it can fail hard |
| 15:59.42 | _VoiceMeUp_COM | thing is we are pushing asterisk to the limits.. |
| 15:59.47 | _VoiceMeUp_COM | i think we use EVERY app |
| 15:59.54 | _VoiceMeUp_COM | or will soon |
| 16:00.20 | _VoiceMeUp_COM | we ony staying away from other channels and local and h323 for now |
| 16:01.02 | _VoiceMeUp_COM | also noticed theres a BIG interop prob between sip astetrisk and Iax openpbx lol , but whos to blame ;) missing eggs with rocks |
| 16:01.07 | seele_ | ok I check the file and is fine ... how can I solve this error http://www.pastebin.ca/566558 |
| 16:01.10 | _VoiceMeUp_COM | s/missing/mixing/ |
| 16:03.54 | mocker | Does anyone know if Asterisk using ODBC storage still suppports MWI/ |
| 16:03.55 | mocker | ? |
| 16:03.58 | seele_ | how can i disable MySQL RealTime?? |
| 16:04.30 | [TK]D-Fender | seele_: Dialparties... YAY... do we even have to say it?! |
| 16:04.31 | _VoiceMeUp_COM | hmmm not use it ? |
| 16:04.32 | [TK]D-Fender | ~freepbx |
| 16:04.33 | jbot | from memory, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:05.16 | seele_ | ok ... sorry |
| 16:05.21 | _VoiceMeUp_COM | wow that freepbx asnwer from bot is the perfect answer to tell.. good job |
| 16:05.35 | [[blah]asfd | is there an easier way to do hints for 100 phones than to do hint,SIP/100&SIP/101&SIP/102... etc since macros dont work? |
| 16:05.40 | frk2 | but freepbx helps |
| 16:05.44 | _VoiceMeUp_COM | macros dont work ? |
| 16:05.58 | _VoiceMeUp_COM | you could use AEL i guess |
| 16:05.59 | *** part/#asterisk sethtrain (n=seth@nsc69.38.115-233.newsouth.net) |
| 16:06.01 | _VoiceMeUp_COM | and loop |
| 16:06.44 | [TK]D-Fender | [[blah]asfd: hints have nothing to do with macros |
| 16:06.51 | ManxPower | _VoiceMeUp_COM: in .18 and .17 at least, there is a bug where macros can crash the system at random times. (system==asterisk) |
| 16:06.58 | [TK]D-Fender | _VoiceMeUp_COM: And what do you think AEL does?! |
| 16:07.14 | [TK]D-Fender | _VoiceMeUp_COM: All AEL does is parse back to standard logic anyways. |
| 16:07.32 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 16:08.11 | _VoiceMeUp_COM | yeah |
| 16:08.44 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 16:08.51 | _VoiceMeUp_COM | cant you do for (x=0; ${x} < 3; x=${x} + 1) { hint,SIP/10$x } ? |
| 16:09.09 | _VoiceMeUp_COM | tought you could sorry then |
| 16:09.10 | [[blah]asfd | [TK]D-Fender: right... that is why i have the issue... I want to do hints on 100+ phones. I was hoping I could just add a hint priority in my macro, but that doesnt work. So I see that one option is to add a hint priority and include every extension in that line. |
| 16:09.17 | [[blah]asfd | is there a more simple way to do that? |
| 16:09.42 | _VoiceMeUp_COM | <PROTECTED> |
| 16:09.43 | _VoiceMeUp_COM | i mean |
| 16:10.16 | _VoiceMeUp_COM | ah he wants all of them ? hmm aint that hint bad anyway |
| 16:10.32 | [[blah]asfd | hints bad? |
| 16:10.36 | _VoiceMeUp_COM | tough hint was exten => 100,hint,sip/100 |
| 16:10.41 | codefreeze | _VoiceMeUp_COM: nope. Can't define priorities like that |
| 16:10.44 | _VoiceMeUp_COM | i mean the way you supplied it |
| 16:11.01 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 16:11.10 | exoxe | <PROTECTED> |
| 16:11.13 | exoxe | yeah, I didn't hear that :x |
| 16:11.36 | _VoiceMeUp_COM | <PROTECTED> |
| 16:11.38 | _VoiceMeUp_COM | this then ? |
| 16:11.42 | _VoiceMeUp_COM | oups |
| 16:11.55 | [[blah]asfd | <PROTECTED> |
| 16:11.57 | _VoiceMeUp_COM | for (x=0; ${x} < 3; x=${x} + 1) { 10${x},hint,SIP/10${x} } |
| 16:11.59 | _VoiceMeUp_COM | like this |
| 16:12.03 | codefreeze | _VoiceMeUp_COM: nope. hints are part of the extension declaration |
| 16:12.17 | _VoiceMeUp_COM | hmm ok |
| 16:12.38 | _VoiceMeUp_COM | then you could used sed / awk and a shell |
| 16:12.49 | _VoiceMeUp_COM | to populate the line and echo >> dialplan and move to right spot |
| 16:16.51 | HarryR | _VoiceMeUp_COM: nice website |
| 16:16.55 | *** join/#asterisk n00dle (n=ccraft@officewall.springsips.com) |
| 16:17.35 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 16:20.10 | n00dle | Has anyone experienced the "ghost-call" problem with CLI message "zt_handle_event:Ring/Off-hook in strange state 6 on channel" ? |
| 16:20.48 | *** join/#asterisk imapfool (n=edhorton@adsl-66-137-204-217.dsl.stlsmo.swbell.net) |
| 16:21.03 | NOT_guru | how stable is the 1.4 branch of asterisk? I have been hesitant to move to that for my production box's ( small office 6 people 4 phones, and home 6 phones and me =P ) |
| 16:21.25 | NOT_guru | oops cisco phones |
| 16:21.45 | *** part/#asterisk btsteve (n=btsteve@204.10.20.30) |
| 16:22.18 | n00dle | NOT_guru, mine's just fine as long as I don't put any bad patches in. |
| 16:22.37 | NOT_guru | LOL |
| 16:22.44 | NOT_guru | well thats promising |
| 16:22.54 | NOT_guru | maybe I will stick with what is working fine |
| 16:22.58 | NOT_guru | whats the ole addage |
| 16:23.07 | NOT_guru | don't fix what ain't broken |
| 16:23.53 | *** join/#asterisk angryuser (n=Miranda@df01t2-212-195-106-226.d4.club-internet.fr) |
| 16:23.53 | imapfool | Mine has been very stable. I even use ODBC voicemail storage, which finally works OK. Forget IMAP storage for the moment with the 1.4.4 release. |
| 16:24.45 | frk2 | so what frontend do you people use for PBX systems - or just directly editing conf files? |
| 16:24.56 | _VoiceMeUp_COM | vi and nano |
| 16:25.07 | frk2 | yeah thats what i used to use |
| 16:25.19 | *** join/#asterisk saftsack (n=saftsack@pD9E044CC.dip.t-dialin.net) |
| 16:25.22 | frk2 | but it gets difficult to manage with consistency |
| 16:25.24 | n00dle | I tried trixbox... it shat itself when I tried to upgrade it. |
| 16:25.30 | _VoiceMeUp_COM | depedns on your tolerance for risk, and problems |
| 16:25.48 | _VoiceMeUp_COM | if you like wasting time and all then use any of the gui's |
| 16:26.07 | _VoiceMeUp_COM | heck i almost wish the dialplan was in c |
| 16:26.23 | [TK]D-Fender | frk2: its as consistent as YOU are... |
| 16:26.25 | imapfool | I use ODBC realtime for most config files. Easy to keep track of what is going on. |
| 16:26.40 | _VoiceMeUp_COM | yeah , but no caching right ? |
| 16:26.43 | frk2 | yeah but i myself dont work on the 25 different installations that are there :P |
| 16:26.46 | _VoiceMeUp_COM | and caching on where you dont want |
| 16:26.52 | Trevor_b | bah, make contexts, and seperate out your PBX into logical configs, its easy to manage then. |
| 16:26.54 | frk2 | thats the problem |
| 16:27.16 | _VoiceMeUp_COM | like.. realtime heck.. change callerid of a sip ara user.. needs a sip prune realtime and a sip show peer x load |
| 16:27.18 | frk2 | realized that GUIs give you some sort of consistency over the deployment, otherwise dudes make mistakes |
| 16:27.44 | Trevor_b | dudes make mistakes in gui's and gui's break too ;) |
| 16:27.48 | Teccy | or script it yourself |
| 16:27.49 | _VoiceMeUp_COM | hey i remember the days where a double comma would crash it on extensions reload |
| 16:28.10 | frk2 | you need to be really stupid to make a mistake in a gui :) |
| 16:28.21 | _VoiceMeUp_COM | but yeah.. for example.. trixbox loads sip.conf..sip_Additional.conf then loads sip_custom |
| 16:28.44 | _VoiceMeUp_COM | BUT.. your register = is in the custom , and soemthing in addition would def break it |
| 16:28.47 | _VoiceMeUp_COM | so it never reg's |
| 16:28.53 | _VoiceMeUp_COM | something along those lines anyway |
| 16:29.01 | _VoiceMeUp_COM | i get about 3-4 calls a day for tat prob |
| 16:29.12 | frk2 | heh |
| 16:29.14 | frk2 | yeah |
| 16:29.19 | frk2 | cant touch the 'addition' ones |
| 16:29.21 | _VoiceMeUp_COM | once you know it its easy to fix.. but that a perfect example of WHY trix looses REG's |
| 16:29.24 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 16:29.32 | _VoiceMeUp_COM | frk .. its all online conf'ed |
| 16:29.41 | _VoiceMeUp_COM | its just loading the file sin wrong order |
| 16:29.52 | mvanbaak | 18:26 < _VoiceMeUp_COM> heck i almost wish the dialplan was in c |
| 16:29.57 | mvanbaak | did you try AEL ? |
| 16:30.01 | *** join/#asterisk barretj (n=barretj@unaffiliated/barretj) |
| 16:30.07 | Defraz | I keep having Asterisk stoping every so often. It just stops. I go in and it starts up just fine. |
| 16:30.09 | _VoiceMeUp_COM | so .. hop we go vi it .. then chmod 444 sip*.conf |
| 16:30.35 | _VoiceMeUp_COM | defrax |
| 16:30.49 | frk2 | are you using 1.2.17/18? |
| 16:30.49 | _VoiceMeUp_COM | enable debug and verbose.. make sure full is enabled in logger.conf |
| 16:30.50 | frk2 | :) |
| 16:31.01 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 16:31.03 | Defraz | okay VoiceMeUp |
| 16:31.05 | angryuser | _VoiceMeUp_COM: a cheap way to advertise yourself |
| 16:31.07 | _VoiceMeUp_COM | then parse last 1500 lines of log to see what it doesnt like |
| 16:31.16 | _VoiceMeUp_COM | yeah i know.. ask voicepulse about it |
| 16:31.56 | _VoiceMeUp_COM | i guess its easier for our clients to find us when they need us , then sending crap to lists saying we never around to help |
| 16:32.40 | *** join/#asterisk logyati (n=suporte@201.29.73.49) |
| 16:32.48 | logyati | hi ppl :D |
| 16:33.33 | angryuser | _VoiceMeUp_COM: i dont mind helping people, but it is * channel |
| 16:34.51 | logyati | i know its asterisk channel, but, where can i chat about integration openser+asterisk? |
| 16:34.57 | *** join/#asterisk MrSnivvel (n=MrSnivve@66.239.96.66.ptr.us.xo.net) |
| 16:35.26 | HarryR | grr, if anybody else complains about the asterisk dialplan i'll see if I can hack a javascript interpreter into it to replace the extensions module ;) |
| 16:35.49 | logyati | i agree |
| 16:35.51 | logyati | ^^ |
| 16:36.19 | HarryR | logyati: what sort of integration do you need to do? this channel's as good as any for it |
| 16:36.23 | [TK]D-Fender | HarryR: .... EW |
| 16:36.39 | HarryR | [TK]D-Fender: anything but asterisk's retarded extension language |
| 16:37.51 | [TK]D-Fender | HarryR: Its not the parser thats the problem. Its *'s complete lack of data types. All the APPS work the way they do, as do variables, etc. the ENTIRE foundation, and ALL app would need a complete rewrite |
| 16:38.16 | HarryR | :D |
| 16:38.25 | logyati | well, first a wanna ask what asterisk can do with sip? asterisk isnt a sip proxy, right? |
| 16:38.32 | [TK]D-Fender | HarryR: Of course... I didn't say this would be a BAD thing ;) |
| 16:38.41 | frk2 | whats the point of strict prototyping anyways? |
| 16:38.47 | [TK]D-Fender | logyati: Correct, it most certainly NOT. |
| 16:38.53 | Corydon76-home | logyati: correct. It can be a gateway, not a proxy |
| 16:39.02 | frk2 | hate Java for it - only gets in the way of things |
| 16:39.03 | [TK]D-Fender | ~b2bua |
| 16:39.03 | jbot | i guess b2bua is a back 2 back user agent |
| 16:39.05 | [TK]D-Fender | ^^^^^^ |
| 16:39.14 | [TK]D-Fender | logyati: Go read... THE BOOK |
| 16:39.16 | [TK]D-Fender | ~book |
| 16:39.17 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:39.17 | HarryR | frk2: Typing in Java is quite nice, and in C# it's even nice |
| 16:39.27 | Corydon76-home | Hey, jbot's not slow today |
| 16:39.34 | frk2 | ived used java |
| 16:39.54 | HarryR | and I use it maybe 6 hours a day |
| 16:40.13 | logyati | yes, you got my point, how it works? i mean, whats the role of asterisk when it acts as a sip gateway? how and when (the moment) asterisk communicates with openser? |
| 16:40.20 | frk2 | i used it for 18 hours a day 3 years ago for 1 year straight |
| 16:40.29 | HarryR | 18 hours? |
| 16:40.45 | logyati | im reading, but the book doesnt explin about integration with sip proxies |
| 16:40.47 | frk2 | then i decided it was completely gay as the exact same thing can be done in half the time using c, perl or php |
| 16:41.02 | [TK]D-Fender | logyati: * acts like an ENDPOINT in all respects. |
| 16:41.07 | Corydon76-home | frk2: uh, hey, we don't talk like that in here |
| 16:41.31 | frk2 | sorry :) |
| 16:41.38 | frk2 | didnt know there were rules :) |
| 16:41.46 | Corydon76-home | There are far too many gay developers |
| 16:41.55 | _VoiceMeUp_COM | lol |
| 16:41.55 | frk2 | :P |
| 16:41.57 | frk2 | haha |
| 16:42.13 | frk2 | who said i meant gay in a negative way? |
| 16:42.14 | [TK]D-Fender | RulesStrict-- |
| 16:42.15 | frk2 | gay = happy |
| 16:42.21 | logyati | [tk]d-fender please can you be more specific? i mean, explain more, cos i didnt find any google results about this integration, i just found "how to"s |
| 16:42.27 | HarryR | frk2: it's 2007, not 1945 |
| 16:42.39 | frk2 | :D |
| 16:42.57 | [TK]D-Fender | frk2: Yeah.... like when that "eccentric" Uncle of yours gets dragged away in an oversized white jacket for being "too happy" <- |
| 16:43.20 | [TK]D-Fender | logyati: * acts like a Phone would on EACH end of a call and just BRIDGES them internally. |
| 16:43.22 | HarryR | why would he get dragged away in a white jacket for being gay? |
| 16:43.36 | [TK]D-Fender | logyati: So look how any SIP phone would react, and that's * |
| 16:43.53 | frk2 | yeah that made no sense [TK]D-Fender |
| 16:44.10 | [TK]D-Fender | frk2: crazy. |
| 16:44.20 | barretj | anyone know what is involved in turning on land-line phone service for someone who just has dry loop dsl? |
| 16:44.27 | logyati | hmmm now i understand... * act like as a bridge |
| 16:44.34 | logyati | ops, act as a bridge |
| 16:44.37 | [TK]D-Fender | barretj: Call the telco. |
| 16:44.37 | mocker | ? |
| 16:44.38 | barretj | i'm just wondering, because i called my phone company and they said it would take 7 days |
| 16:45.04 | [TK]D-Fender | barretj: because they can't send a guy over to your switch to bridge the wire. |
| 16:45.07 | barretj | which is weird because all along i've had a dial tone, so you would think they'd just have to flip a switch |
| 16:45.23 | barretj | [TK]D-Fender: what is providing the dial tone then? |
| 16:45.32 | [TK]D-Fender | barretj: You shouldn't have any |
| 16:45.38 | barretj | [TK]D-Fender: i do though |
| 16:45.45 | logyati | [tk]d-fender so, when i make a sip call in openser, it checks if the user exists, if not, openser direct the call to asterisk, i think |
| 16:45.48 | coppice | DSL without dialtone is bad news |
| 16:45.51 | barretj | [TK]D-Fender: when i try to make a call, i get a busy signal every time |
| 16:45.59 | barretj | coppice: why is it bad news? |
| 16:46.10 | [TK]D-Fender | barretj: Could be they have you hooked up to an empty tone generator. |
| 16:46.14 | *** join/#asterisk cullenincrease (n=cp@c-75-64-44-200.hsd1.tn.comcast.net) |
| 16:46.16 | cullenincrease | hey |
| 16:46.27 | coppice | because the monkeys keep ripping out and reallocating the pair when a butt phone reveals no dialtone |
| 16:46.33 | barretj | coppice: i only went back to regular phone service because my voip service sucked |
| 16:46.43 | *** join/#asterisk KnckrBckr (n=KnckrBck@200.32.224.186) |
| 16:46.44 | barretj | coppice: haha |
| 16:47.29 | Cresl1n | how is that? |
| 16:47.31 | barretj | [TK]D-Fender: when i try to make a long distance call without the area code, it tells me to dial the area code first, doesnt sound like an empty tone generator to me |
| 16:47.40 | coppice | they really shouldn't make is dry. wetting current would improve reliability no end |
| 16:47.48 | barretj | lol |
| 16:48.01 | barretj | actually, whenever there's a heavy rain, my dsl service drops out |
| 16:48.04 | barretj | completely |
| 16:48.14 | barretj | no dsl service at all when its raining heavily |
| 16:48.19 | logyati | [tk]d-fender but how asterisk convert the call that came using SIP from openser, to be sent through fxo to pstn? |
| 16:48.26 | coppice | a lot of people get that |
| 16:48.46 | barretj | coppice: alot of people get the rain problem? |
| 16:48.58 | cullenincrease | we've set up an office in manila to act as a call center for our customers and are looking towards using asterisk to handle voip calls. i've got a few questions i was hoping someone could answer a few questions for me or could possibly hire someone as a consultant to help us get going? |
| 16:49.00 | HarryR | logyati: it acts like a phone, answers the call from the phone that made the call, and then bridges it internally |
| 16:49.06 | coppice | we used to, and had to complain at the total PITA level for ages before we got a new pair |
| 16:49.16 | barretj | haha |
| 16:49.18 | coppice | barretj: yeah. |
| 16:49.22 | HarryR | cullenincrease: how many people? |
| 16:49.35 | barretj | its like the wires get soggy or something |
| 16:49.36 | HarryR | uh, how many agents are you looking at having in the call centre? |
| 16:49.37 | cullenincrease | we have 10-15 customer service agents ready over there |
| 16:49.45 | barretj | why manila? |
| 16:49.52 | HarryR | Because labor is cheap? |
| 16:49.54 | cullenincrease | thats another story :) |
| 16:49.55 | logyati | harryr where do i configure asterisk to answer this sip call? extensions.conf? do i need sip.conf? |
| 16:49.57 | barretj | lower taxes? |
| 16:50.03 | [TK]D-Fender | logyati: * acts like a phone and it answers as such. it then does whatever your dialplan tells it to based on the INVITE parameters and BRIDGES |
| 16:50.05 | HarryR | logyati: yup, extensions.conf |
| 16:50.12 | [TK]D-Fender | logyati: no differnt than anything else. |
| 16:50.16 | coppice | gee, I hate working in manila |
| 16:50.16 | HarryR | cullenincrease: take a look at call queuing and agentlogon |
| 16:50.17 | logyati | hmmm |
| 16:50.21 | cullenincrease | hmm |
| 16:50.25 | cullenincrease | ok |
| 16:50.45 | barretj | do they use manila envelopes in manila? or just regular ones? |
| 16:50.46 | [TK]D-Fender | logyati: Without extensions.conf your system does NOTHING. Devices do NOT magically "talk" to each other. |
| 16:50.49 | barretj | :) |
| 16:50.52 | HarryR | there are companies that can do this sort of thing for you, take a look at Telappliant ( http://www.telappliant.com ) - the VoIPoffice product |
| 16:50.55 | logyati | ok, and where sip.conf is usefull? |
| 16:50.59 | [TK]D-Fender | logyati: * is a B2BUa for ALL tech, not just SIP. |
| 16:51.00 | cullenincrease | one of my biggest questions is where to find an extremely reliable hosting company for asterisk that can handle a large volume of calls |
| 16:51.24 | coppice | is a .ph bankrupcy a manila folder? |
| 16:51.44 | [TK]D-Fender | logyati: sip.conf tells * the parameters and accounts it should ALLOW to taalk, and using what codecs, ip's, etc. HOW it processes a call that actually ARRIVES is the responsibility of extensions.conf |
| 16:52.24 | [TK]D-Fender | coppice: a .ph bankrupcy is where someone ran out of vinger and reaches for baking-soda ;) |
| 16:53.23 | barretj | anyone use the grandstream GXP-2000 ? |
| 16:53.32 | [TK]D-Fender | ~gs |
| 16:53.32 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 16:53.35 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 16:53.41 | *** part/#asterisk Nuitari (n=nuitari@melchior.nuitari.net) |
| 16:53.42 | barretj | mine seems to drop calls randomly |
| 16:53.45 | HarryR | ahah |
| 16:53.51 | n00dle | Aw, c'mon... I'm using them just fine. |
| 16:54.07 | HarryR | something like 80% of the product returns we have are grandstreams |
| 16:54.18 | barretj | haha |
| 16:54.45 | barretj | i dont mind it too much when i'm on the phone with a friend, but when i'm on the phone with my boss, it sucks |
| 16:54.56 | barretj | so i just said screw it, i'm switching back to POTS |
| 16:55.14 | [TK]D-Fender | barretj: Sad... could have simply bought a DECENT phone instead... |
| 16:55.33 | barretj | [TK]D-Fender: how much would a decent phone cost? |
| 16:55.43 | barretj | [TK]D-Fender: i already spend $80 on the grandstream |
| 16:55.44 | [TK]D-Fender | barretj: $93 USd |
| 16:56.29 | [TK]D-Fender | barretj: http://www.digiumcards.com/polycom_soundpoint_ip_320.html |
| 16:56.47 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 16:56.57 | barretj | whats the point when i can get nearly 100% reliability with a $10 radioshack phone and POTS? |
| 16:57.14 | barretj | i dont care about features, only reliability |
| 16:57.31 | KnckrBckr | somebody else setup my system here... i am very new to asterisk. all of the sudden, if on outgoing calls, the other party picks up in the first 2 rings they get dead air and i keep hearing the line ring. especially annoying with 800#s where they always pick up immediately. any ideas where to start? I have a feeling it has to with a flaky PRI from our CLEC |
| 16:57.36 | HarryR | because pots phones are extremely simple |
| 16:58.03 | barretj | greater simplicity == greater reliability |
| 16:58.04 | [TK]D-Fender | barretj: Reduced recurring line costs, speakerphone & headset options, multi-line conferencing, interactive services, lack of telco reconnect fees... I could go on you know... |
| 16:58.17 | HarryR | which is why everybody still uses cups & string |
| 16:58.55 | [TK]D-Fender | KnckrBckr: We'd need CLI output of the failure at verbose 10 in a * PASTEBIN * |
| 16:58.57 | [TK]D-Fender | ~pb |
| 16:58.58 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 16:59.14 | KnckrBckr | ahh, ok |
| 17:00.30 | logyati | guys, and what about integration of gnugk with asterisk? why ppl do that? |
| 17:01.42 | [TK]D-Fender | logyati: Because H.323 in * SUCKS. its unstable and the B2BUA nature of it requires workarounds in several cases. |
| 17:01.44 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-96-179.ph.ph.cox.net) |
| 17:02.55 | logyati | [tk]d-fender but i dont understand why need h323, if asterisk answer sip calls and bridge to pstn |
| 17:03.30 | [TK]D-Fender | logyati: YOU brough up the subject. * also speak in SCCP and a few other protocals like IAX. Why aren't you complaining about THEM TOO? |
| 17:17.45 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 17:17.45 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.18, 1.4.3 (June 8, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 17:18.14 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 17:35.08 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 17:35.08 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.18, 1.4.3 (June 8, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 17:35.32 | [TK]D-Fender | Taadow: What ver of * are you using, and Zaptel if applicable? |
| 17:35.55 | Taadow | D-Fender: 1.2.18 for both. |
| 17:36.11 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 17:36.17 | Taadow | Have a tdm2432e in her. |
| 17:37.14 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 17:37.28 | Taadow | Which mog helped me setup, once upon a time. Mighty kind of him. |
| 17:37.32 | [TK]D-Fender | Taadow: Well newer zaptel's are supposed to help, but the Intel E1000 used to be on the supreme-Digium-no-no list |
| 17:38.08 | Taadow | Really. The interesting thing is this occurance was happening before with two seperate nic's. The change was recent and only because of this issue, and appears to have had no effect. |
| 17:38.10 | [TK]D-Fender | Taadow: You might want to check out your CPI / IRQ load |
| 17:38.22 | Taadow | ie, issue still persists. System load is next to nothing. |
| 17:38.29 | [TK]D-Fender | and PCI / IRQ? |
| 17:38.55 | *** join/#asterisk Trevor_b (n=tbenson@69.12.220.201) |
| 17:38.57 | Taadow | I've never learned how to check PCI/IRQ load.. forgive my newbness in that respect. An easy method? |
| 17:39.13 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 17:39.27 | syzygyBSD | is it bad if I get a bunch of "chan_sip.c: BAD! BAD! BAD!" in my message log? |
| 17:39.35 | Taadow | heheh |
| 17:39.40 | [TK]D-Fender | Taadow: Not from me. I'm still a Linux n00b, but have passed on as much relevent hints as I can. |
| 17:40.00 | [TK]D-Fender | syzygyBSD: "Doctor, it hurts when I raise my arm like this!" |
| 17:40.05 | Taadow | Must appreciated. :D |
| 17:40.10 | syzygyBSD | [TK]D-Fender: :) |
| 17:40.11 | Qwell[] | [TK]D-Fender: DON'T DO THAT THEN |
| 17:43.19 | *** join/#asterisk guillote_GNU (n=guillote@190.7.30.134) |
| 17:43.46 | Qwell[] | [TK]D-Fender: newer 501 firmware implemented micro browser, right? |
| 17:43.54 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:44.51 | [TK]D-Fender | Qwell[]: Yup, older purchases keep getting better... |
| 17:45.54 | *** join/#asterisk xkev (i=kevin@orbit.xmission.com) |
| 17:46.10 | xkev | any users of polycom 550? need to know if the idle screen microbrowser exists like 600/601 has |
| 17:46.22 | Qwell[] | heh |
| 17:46.47 | xkev | I display all my phone mode, queue stats/membership, voicemail count etc on it |
| 17:47.12 | [TK]D-Fender | xkev: All that have the MB have an idel. |
| 17:47.20 | Qwell[] | xkev: [TK]D-Fender just told me that the 501's do, so yeah, I would imagine that the 550 would too |
| 17:47.20 | [TK]D-Fender | xkev: IP 550 = WASTE |
| 17:47.26 | xkev | 501s don't |
| 17:47.34 | [TK]D-Fender | xkev: Yes, they DO. |
| 17:47.34 | xkev | unless they added it in newer firmware |
| 17:48.04 | [TK]D-Fender | "They don't!!!!!!!!!!!!!!!!! (unless they do)" |
| 17:48.06 | xkev | I only have 500s, which don't have enough resource for >1.3.x :) |
| 17:48.08 | [TK]D-Fender | feh! |
| 17:48.44 | [TK]D-Fender | xkev: incorrect. the IP 500 is supported by every SIP release through current |
| 17:48.59 | xkev | well, 1.4.x caused it to fail to flash |
| 17:49.07 | [TK]D-Fender | xkev: if your third statement is correct you'd have a good batting average! |
| 17:49.08 | Taadow | On a positive note, I found some juicy info last night that leads me to believe it is possible to get my Shortel 530 working w/ *. Yay! |
| 17:49.08 | xkev | never attempted to raise above 1.3.4 |
| 17:49.20 | [TK]D-Fender | xkev: sad... |
| 17:49.45 | xkev | fender, explain "550 = WASTE" (I have 601s everywhere, want 550 for "601 with backlight") |
| 17:49.55 | Qwell[] | xkev: 650 |
| 17:50.34 | xkev | 550 -> 650 = $50 more, multiply by many phones |
| 17:50.35 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 17:50.40 | xkev | call center |
| 17:51.07 | [TK]D-Fender | xkev: 650 is 601 + a backlight. the 550 costs only a little less than the 650 only it lacks 2 line keys and has no expansion possibility. |
| 17:51.20 | xkev | ok, then it's fine for me |
| 17:51.21 | xkev | :) |
| 17:51.38 | [TK]D-Fender | xkev: Need a backlight that bad for a call center? |
| 17:51.47 | xkev | it's not a typical call center |
| 17:51.59 | xkev | old school ISP with no overhead lighting |
| 17:52.05 | xkev | dark pit of techs with desk lamps |
| 17:52.19 | xkev | good for the geeks |
| 17:52.20 | [TK]D-Fender | xkev: Get them a simple phone like the 320 and spend that extra $ on a NICE lamp & a cheap painting and watch their productivity SOAR |
| 17:53.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:53.21 | Qwell[] | or a cheap lamp and nice painting |
| 17:53.25 | Qwell[] | light is light is light |
| 17:53.50 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 17:53.59 | Qwell[] | though I guess the difference in price between a nice lamp and a cheap lamp is far less than a nice painting vs a cheap painting... |
| 17:54.16 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 17:58.14 | *** part/#asterisk cr4z3d (n=cr4z3d@ip70-162-96-179.ph.ph.cox.net) |
| 17:59.07 | *** join/#asterisk EvilGreen_ (n=Miranda@ppp85-141-153-93.pppoe.mtu-net.ru) |
| 18:01.00 | *** join/#asterisk zaggynl (n=zaggynl@dsl-083-247-110-137.solcon.nl) |
| 18:03.59 | *** join/#asterisk mitcheloc (n=mitchelo@h460767a6.area7.spcsdns.net) |
| 18:07.42 | *** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl) |
| 18:09.10 | *** join/#asterisk Dovid (n=Dovid@bzq-88-154-62-187.red.bezeqint.net) |
| 18:11.11 | *** join/#asterisk Netgeeks (n=root@pbx5.netgeeks.net) |
| 18:11.37 | Netgeeks | Hey, anyone familiar with [featuremap] in features.conf here? |
| 18:16.46 | xkev | lamps glare badly off the angle of the phone, and we've tried usb LED lamp things, they age quickly |
| 18:17.01 | xkev | backlights make a geek happy |
| 18:17.01 | xkev | :) |
| 18:17.39 | xkev | it's really a non-issue, they've managed for 2 years w/o backlight |
| 18:20.16 | [TK]D-Fender | xkev: At which point the IP 320 is massively more cost effective and offers you dual headset options. |
| 18:21.19 | xkev | but does it have an idle mb that will show 4 lines of data updating every 15 sec |
| 18:21.44 | *** join/#asterisk Nuitari (n=nuitari@142.46.207.230) |
| 18:21.55 | Nuitari | Hi |
| 18:22.33 | Nuitari | Since I've started using trunk, I'm getting a lot of Internal RTCP NTP clock skew detected: lsr=83641711, now=83741984, dlsr=131000 (1:998ms), diff=30727 messages while on a call. Is it an issue and / or can it be fixed ? |
| 18:22.45 | [TK]D-Fender | xkev: Depending on how much actual datat, yes |
| 18:23.18 | *** join/#asterisk NirS_ (i=Nir@87.68.79.182.cable.012.net.il) |
| 18:24.24 | xkev | looks like an asstastic 301 screen |
| 18:24.43 | *** part/#asterisk Netgeeks (n=root@pbx5.netgeeks.net) |
| 18:28.07 | [TK]D-Fender | xkev: Not as big, but its pixel based |
| 18:29.53 | [TK]D-Fender | xkev: And truth be told I'd rather have an IP 301 at my desk than the Aastra 57i CT I have now. |
| 18:30.18 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 18:30.35 | skirmisha | guys any idea how can i auth incoming calls based on ip |
| 18:31.14 | xkev | I have a friend who rolled * with the 57i's, they seemed pretty badass |
| 18:31.47 | xkev | ..mainly for customization stuff |
| 18:32.11 | cullenincrease | anyone have experience stabalizing asterisk and VOIP connections? specifically, we want to see about hosting our asterisk server here in the U.S. and having it route calls to softphones at our office in the philippines. It is my understanding that a method such as this will dramatically increase quality and stability and we really need some consulting on the issue. |
| 18:32.35 | xkev | all the things that polycom plagues me on, like remapping keys and context-based data, centralized phonebook (polycom's config-based thing is crap) |
| 18:32.37 | [TK]D-Fender | xkev: Audio quality is lacking, the handset has NO wieght, screen though pixel based is running of a CHAR MATRIX friggen firmware. No "join" option, ANNOYING RUBBER BUTTONS, and the wireless handset can't operate independent of the base (eg it rings here all the time) |
| 18:32.57 | [TK]D-Fender | xkev: Call handling is second rate |
| 18:33.05 | xkev | I submit all phones are shit in one way or another |
| 18:33.21 | [TK]D-Fender | xkev: Only strong side is the kick-ass attendant module (which I didn't get), and the soft-keys. |
| 18:33.47 | [TK]D-Fender | xkev: comparing Polycom & Aastra (or heck ANYONE), tends to have a LOT more wins on the Polycom side. |
| 18:34.34 | [TK]D-Fender | xkev: Only phone that does corporate directories right is Cisco... and there's a boatload of greif right there :) |
| 18:34.56 | xkev | yeah |
| 18:34.57 | [TK]D-Fender | xkev: As I've said to many others. ALL phone directories suck by comparison to a PAPER LIST. |
| 18:35.24 | xkev | if polycom would disclose exactly what the browser can do, and perhaps add a custom tag for <dialthis> ? |
| 18:35.25 | mocker | sigh.. |
| 18:35.26 | [TK]D-Fender | IP 301 + Paper list and this 57i can kiss my ass! |
| 18:35.48 | irule | hi, I want to set languages per extension, is this the way to go? Set(ext500=es) Set(ext502=en) |
| 18:36.12 | [TK]D-Fender | xkev: <a href="dial://1234567"> ........ and you're welcome. |
| 18:36.19 | xkev | orly |
| 18:36.22 | [TK]D-Fender | rly |
| 18:36.29 | [TK]D-Fender | kthxbye |
| 18:36.37 | xkev | is there a doc covering the tags they /do/ support? |
| 18:36.48 | xkev | they just expanded it a bit with 2.x |
| 18:36.50 | [TK]D-Fender | irule: extensions don't HAVE a language. |
| 18:36.51 | mcab | [TK]D-Fender: s/dial/tel/ |
| 18:37.00 | [TK]D-Fender | oops. |
| 18:37.02 | [TK]D-Fender | yeah, that |
| 18:37.05 | mcab | :-) |
| 18:37.12 | [TK]D-Fender | xkev: <a href="tel://1234567"> ........ and you're welcome. |
| 18:37.20 | [TK]D-Fender | mcab: brain-fart :) |
| 18:37.28 | [TK]D-Fender | mcab: only used it once. |
| 18:37.30 | mcab | :-D |
| 18:37.49 | *** join/#asterisk ghento__ (n=ghento@CPE001124d2c50e-CM0011e6c416f1.cpe.net.cable.rogers.com) |
| 18:38.54 | irule | [TK]D-Fender I know, but I have a case where a caller should choose among english and spanish |
| 18:39.38 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 18:44.06 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 18:44.09 | skirmisha | guys is it possible to tell asterisk to do nat on calls that are just forwarded |
| 18:44.41 | Dr-Linux | well, i've installed asterisk-gui with 1.4.4 and configured it, now how can i access it? :S |
| 18:45.49 | [TK]D-Fender | irule: Yuo can change the CHANNEL's language via its appropriately named FUNCTION. |
| 18:46.12 | [TK]D-Fender | skirmisha: elaborate please |
| 18:46.26 | [TK]D-Fender | Dr-Linux: Go ask in #asteriskgui like the topic says |
| 18:47.00 | *** join/#asterisk dlynes_ (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 18:47.05 | Dr-Linux | [TK]D-Fender: there is no channel #asteriskgui |
| 18:47.20 | gerphimum | #asterisk-gui.. like the topic says |
| 18:47.23 | Dr-Linux | ohh got it |
| 18:48.17 | Trevor_b | If I was looking to have asterisk make 100 simultainous sip calls, how much CPU and RAM would you suggest? |
| 18:48.30 | [TK]D-Fender | So I failed to hyphenate.... SUE ME. I'll tie you up in litigation just like I did last tuesday on BDSM night! |
| 18:48.43 | [TK]D-Fender | Trevor_b: Depnds on transcoding. |
| 18:49.01 | gerphimum | also if any of those will be conference calls |
| 18:49.32 | Trevor_b | You can assume all 100 are meetme conferences. |
| 18:50.10 | Trevor_b | well actually |
| 18:50.13 | [TK]D-Fender | Trevor_b: If you have 100 SIP calls and 100 meetme conferences.... you could jsut do "MusicOnHold()" for each and be done with it ;) |
| 18:50.29 | [TK]D-Fender | Trevor_b: Since they'll all be... ALONE! |
| 18:50.40 | Trevor_b | Heheh yeah, they will be alone ;) |
| 18:51.10 | Trevor_b | actually i think the meetme was only with an agent waiting anyway, so it would be a background or whatnot until a transfer. |
| 18:51.50 | [TK]D-Fender | Trevor_b: Again the real load is in transcoding, and the words "agent" and "meetme" do not belong in the same place together. |
| 18:51.55 | Trevor_b | MusicOnHold allow for extension like background? |
| 18:52.00 | Trevor_b | hehe |
| 18:52.02 | Trevor_b | sorry |
| 18:52.13 | [TK]D-Fender | Trevor_b: Time to start ALL OVER. |
| 18:52.17 | Trevor_b | most likely ulaw. |
| 18:53.44 | Trevor_b | 100 sip outbound calls, play to background waiting for a digit, and transfer if hit, so no meetme. transcoding likely to be ulaw or gsm, but will use whatever works best. |
| 18:54.19 | [TK]D-Fender | Trevor_b: your stance on transcoding is completely mixed up. |
| 18:54.42 | [TK]D-Fender | Trevor_b: And thats a LOT of bandwidth. Whats terminating these calls? |
| 18:54.50 | [TK]D-Fender | Trevor_b: Connected to how? |
| 18:56.00 | Trevor_b | customer is already doing 100 calls per second on the transport, just wants to replace existing system with asterisk. |
| 18:56.52 | irule | why does this not work? exten => 1,1,Noop(the language is $CHANNEL(language)) |
| 18:57.01 | Qwell[] | ${CHANNEL(language)} |
| 18:57.05 | irule | thanks |
| 18:57.48 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 18:57.59 | irule | works great, thanks! |
| 18:58.52 | Trevor_b | Ill test it in the lab. |
| 19:03.30 | Trevor_b | ~ITSP |
| 19:03.30 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others." (tm) (c) 2007 ManxPower |
| 19:04.05 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 19:05.07 | Dr-Linux | /gone |
| 19:05.09 | irule | exten => 1,n,set(CHANNEL(language)=en) works like a charm |
| 19:05.18 | irule | exten => 1,n,set(CHANNEL(language)=es) |
| 19:07.06 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 19:07.22 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 19:11.19 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 19:18.58 | *** join/#asterisk rene- (n=rene@200.34.66.137) |
| 19:22.16 | *** join/#asterisk suma (n=suma@63.83.99.163) |
| 19:22.41 | suma | can someone please who is the best and cheap provider of voip service in california ? |
| 19:22.51 | suma | i'm looking only for me for outgoing calls |
| 19:23.19 | Trevor_b | best and cheap dont go hand in hand, but teliax offers a good service and has a REALLY low latency CA beta server i use. |
| 19:23.36 | suma | beta ? |
| 19:23.52 | suma | cheap of the best ;) |
| 19:24.08 | Trevor_b | Available for use, but not considered as stable as production (been using it flawless for over 30 days) |
| 19:24.35 | Trevor_b | [TK]D-Fender: if i wasnt transcoding and it was coming in and going out the same, how much CPU and RAM would you suggest? |
| 19:25.47 | [TK]D-Fender | I'd go dual CPU 2gig ram minimum. |
| 19:26.00 | [TK]D-Fender | don't go psycho on the CPU however. |
| 19:28.34 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:30.12 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-47.usadatanet.com) |
| 19:32.33 | exoxe | are invalid entries (i) per context? e.g. if you include another context that has invalid extension defined, should that be inherited? I'm guessing not since it doesn't seem to be the case |
| 19:32.55 | n00dle | anyone know how to get a text message to a GXP2000 without a call being in progress? |
| 19:36.07 | [TK]D-Fender | exoxe: yes, it is inherited. |
| 19:39.44 | exoxe | then I must be doing something wrong |
| 19:39.50 | exoxe | sounds about right |
| 19:41.00 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 19:41.50 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 19:45.08 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 19:46.03 | MihiNomenEst | I'm getting an error when I send a call to the parking lot, it says, "Jun 14 10:03:01 NOTICE[16923]: channel.c:2496 __ast_request_and_dial: Unable to request channel Agent/209 |
| 19:46.03 | MihiNomenEst | Jun 14 10:03:01 WARNING[16923]: app_parkandannounce.c:202 parkandannounce_exec: PARK: Unable to allocate announce channel." where can I get more info? |
| 19:49.19 | KnaveMan | I am having a problem with connecting a softphone from outside my network. If i use a vpn i can connect fine locally, but when i change the domain of xlite to the external ip it fails. Ports for sip, iax, rtp, iax2, have all been opened. Error shown in ngrep is 401 Unauthorized. Any suggestions? |
| 19:49.30 | KnaveMan | btw, nat=yes and all that configuration has been done :) |
| 19:51.20 | MihiNomenEst | host=dynamic? |
| 19:51.25 | KnaveMan | Yes. |
| 19:51.29 | KnaveMan | Even though my office is static ip |
| 19:51.31 | KnaveMan | (we have 13 statics). |
| 19:52.06 | KnaveMan | Its always failing during registration... so its probably something just being overlooked. |
| 19:52.18 | KnaveMan | but i can be safe the configuraiton is correct as it works over vpn with a local ip. |
| 19:54.10 | MihiNomenEst | I'd watch the sip debug and see what the difference is between the public and the tunnel. |
| 19:54.23 | KnaveMan | How would I do that? |
| 19:54.29 | KnaveMan | (im still learning.... fast crash course) |
| 19:55.05 | MihiNomenEst | "sip debug" |
| 19:55.13 | MihiNomenEst | "sip no debug" turns it off. |
| 19:55.20 | KnaveMan | gotcha. |
| 19:55.27 | KnaveMan | it says the same as the ngrep that i was running. |
| 19:55.53 | KnaveMan | Trying.... then 401 Unauthorized. |
| 19:56.35 | *** join/#asterisk DrCron (n=rszasz@c-67-174-231-152.hsd1.ca.comcast.net) |
| 19:56.35 | MihiNomenEst | so what's the difference between the vpn and the public address? |
| 19:56.45 | KnaveMan | vpn gave me a local 192.168 ip. |
| 19:57.04 | KnaveMan | so i was using the local trixbox ip when i connected to it. |
| 19:57.18 | KnaveMan | Whats interesing is this line though.... |
| 19:57.26 | KnaveMan | Sending to 192.168.1.250 : 28967 (NAT) |
| 19:57.42 | KnaveMan | We are using the subnet 192.168.69. for our trixbox |
| 19:57.48 | KnaveMan | I wonder where that 1.250 is hardcoded. |
| 19:57.54 | [TK]D-Fender | ~trixbox |
| 19:57.55 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 19:58.43 | KnaveMan | D-Fender, this isnt a trixbox related issue i dont feel. Its all from the asterisks command line. |
| 19:59.45 | [TK]D-Fender | KnaveMan: Trixbox = Freepbx = soul has been sold to the lowest bidder. |
| 19:59.50 | KnaveMan | hah |
| 20:00.14 | *** join/#asterisk guillote_GNU (n=guillote@host176.201-252-205.telecom.net.ar) |
| 20:00.15 | [TK]D-Fender | KnaveMan: pastebin your sip.conf and everything linked to it masking only passwords. |
| 20:00.26 | KnaveMan | okay, thanks. give me a sec. |
| 20:02.53 | KnaveMan | http://pastebin.com/929152 |
| 20:03.55 | *** join/#asterisk apogee7 (n=steven_h@74.92.229.209) |
| 20:03.56 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:07.17 | [TK]D-Fender | KnaveMan: sounds like you need to add the 192.168.1.0/24 subnet to your local. you'll have an improper return path. |
| 20:08.00 | [TK]D-Fender | KnaveMan: And ensure you have a static route to that subnet. |
| 20:08.09 | KnaveMan | i dont know where 1.0 was assigned... we didnt assign it |
| 20:08.14 | KnaveMan | we dont even use 1.* |
| 20:08.50 | KnaveMan | but that would explain a lot... atleast, the reason why the registration isnt happening... its waiting for a response and never receives it because its going to never-ever-land. |
| 20:09.39 | exoxe | so.. what's the trick to keep MOH playing instead of pausing everytime it's not in use |
| 20:10.14 | [TK]D-Fender | exoxe: use another MoH source. Native works its way, others have their own. |
| 20:10.15 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 20:12.36 | *** join/#asterisk Capps- (n=andrew@67-67-242-2.ded.swbell.net) |
| 20:18.32 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-166-137.columbus.res.rr.com) |
| 20:20.27 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 20:22.58 | *** join/#asterisk brea (n=brea@c-71-195-248-169.hsd1.ma.comcast.net) |
| 20:23.22 | brea | Is there a commercial replacement for spandsp? |
| 20:24.04 | tzafrir_laptop | source compatible? or functionally-equivalent? |
| 20:24.49 | brea | functionally equivalent |
| 20:25.03 | tzafrir_laptop | There are probably. Never bothered looking |
| 20:25.22 | brea | I've looked quite a bit and haven't found anything yet |
| 20:25.35 | brea | Wonder if Digium has anything in the works |
| 20:27.38 | mvanbaak | latero |
| 20:30.45 | *** join/#asterisk suma (n=suma@63.83.99.163) |
| 20:36.10 | irule | where can I find info on users.conf? |
| 20:38.45 | *** join/#asterisk Corydon76-work (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 20:38.45 | *** mode/#asterisk [+o Corydon76-work] by ChanServ |
| 20:39.06 | javar | irule, how configure? |
| 20:40.15 | irule | yes |
| 20:40.53 | irule | I have a doubt, should the user extension match my sip.conf configuration? |
| 20:41.08 | irule | same axtension numbers? |
| 20:43.20 | irule | I see zapchan = , should I change it to sipchan or something? |
| 20:43.48 | seele_ | in what file I add "featuredigittimeout" |
| 20:43.51 | seele_ | ??? |
| 20:44.36 | De_Mon | irule I don't think its required, but it would make sense |
| 20:44.58 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 20:45.19 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 20:46.23 | *** join/#asterisk Marshall-Laptop (n=eman0n@cpe-76-181-166-137.columbus.res.rr.com) |
| 20:48.19 | javar | irule, try: http://www.asterisk.org/doxygen/trunk/Config_sip.html |
| 20:49.18 | seele_ | where can I configure the call pickup *8 |
| 20:49.20 | seele_ | ?? |
| 20:49.41 | De_Mon | seele_ features.conf? |
| 20:49.51 | cullenincrease | anyone have experience stabalizing asterisk and VOIP connections? specifically, we want to see about hosting our asterisk server here in the U.S. and having it route calls to softphones at our office in the philippines. It is my understanding that a method such as this will dramatically increase quality and stability and we really need some consulting on the issue. |
| 20:51.06 | Qwell[] | dramatically increase? no |
| 20:51.31 | Qwell[] | You'll be going over the public internet. There is no guarantee of QoS. |
| 20:51.45 | blitzrage | there is no guarentee of anything |
| 20:51.53 | blitzrage | in fact, you're going to be better off having the server closer to your softphones |
| 20:51.56 | Qwell[] | except trolls |
| 20:52.00 | Qwell[] | trolls are guaranteed |
| 20:52.05 | blitzrage | Qwell[]: and taxes, and death |
| 20:52.10 | Qwell[] | I mean on the internet |
| 20:52.21 | cullenincrease | hmm |
| 20:52.31 | cullenincrease | thats not what they're telling me |
| 20:52.35 | Qwell[] | who is "they"? |
| 20:52.42 | cullenincrease | voip provider |
| 20:52.47 | Qwell[] | good, get it in writing |
| 20:52.55 | Qwell[] | when you drop a packet, or have any jitter, sue |
| 20:53.00 | cullenincrease | lol! |
| 20:53.11 | cullenincrease | so you dont think that would work at all? |
| 20:53.15 | Qwell[] | Sure it'll work |
| 20:53.31 | blitzrage | whether the server is in the US, or the phillippines, the softphones are in the phillippines, so either way, the audio has to cross the pacific |
| 20:53.31 | cullenincrease | well do you think there would even be minor improvement? |
| 20:53.38 | cullenincrease | yeah |
| 20:53.42 | Qwell[] | voip is great, but it most definitely will not "dramatically increase" quality/stability - ESPECIALLY an international connection |
| 20:53.51 | blitzrage | it'll work -- you just won't see any different between the server being in the US or in PI |
| 20:54.17 | cullenincrease | and theres not any sort of additional compression techniques or anything we could do with a setup like that? |
| 20:54.36 | *** join/#asterisk A-Data (n=asd@196.218.74.249) |
| 20:54.56 | A-Data | hello all how to change music on hold? |
| 20:54.56 | Qwell[] | sure there are, something like g729 |
| 20:55.10 | cullenincrease | hmm? |
| 20:55.18 | Qwell[] | codec_g729 |
| 20:55.33 | Qwell[] | it's a highly compressed codec...the quality isn't the best though |
| 20:55.41 | cullenincrease | oh ok |
| 20:55.50 | cullenincrease | but it still wouldnt make a difference where the * server is right? |
| 20:56.06 | Qwell[] | no, because it has to go over the pacific at some point |
| 20:56.10 | cullenincrease | yep |
| 20:56.12 | cullenincrease | this sucks |
| 20:56.16 | Qwell[] | personally, I would put the server in PI, but... |
| 20:56.19 | cullenincrease | we're getting dropped calls |
| 20:56.22 | cullenincrease | galore |
| 20:56.33 | cullenincrease | bad voice quality |
| 20:56.37 | *** part/#asterisk EvilGreen_ (n=Miranda@ppp85-141-153-93.pppoe.mtu-net.ru) |
| 20:56.38 | cpm | don't put it on mars, if it goes down, you'll never get it back up |
| 20:56.49 | Qwell[] | cpm: That's why you have a hot spare. |
| 20:56.57 | Qwell[] | get it...hot?...nevermind |
| 20:57.04 | A-Data | hello all how to change music on hold? i need it as mp3 not stream server any solution? |
| 20:57.41 | cullenincrease | then again |
| 20:57.45 | cullenincrease | if you think about it |
| 20:58.01 | cullenincrease | (tell me if im being stupid here but...) |
| 20:59.07 | cullenincrease | say you've got your * server in PI, then your VOIP provider in USA has to send 1 big fat datastream to route the calls but if your server is here the VOIP provider could be only a couple hundred miles away and you'd be sending lots of tiny data straems to ip phones on PI right? |
| 20:59.15 | cullenincrease | so technically that would decrease chance of dropped calls etc |
| 20:59.23 | cullenincrease | just a theory |
| 20:59.48 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 20:59.52 | cullenincrease | does it add up or no? |
| 21:00.11 | Qwell[] | if the server is in PI, you could do either |
| 21:00.29 | cullenincrease | hm |
| 21:00.47 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:00.49 | cullenincrease | so technically you can send the voip signal straight to the individual softphones? |
| 21:02.06 | *** join/#asterisk krp (n=krp@mar92-10-82-239-65-214.fbx.proxad.net) |
| 21:02.16 | cullenincrease | because sending lots of individual signals instead of just 1 seems more robust to me |
| 21:02.44 | krp | is it the right place to ask about callerid problems with asterisk and tdm400 card ? |
| 21:03.18 | jkiff | krp: Shoot. |
| 21:03.51 | krp | i have a tdm400 card, 2 fxo 2 fxs |
| 21:03.58 | A-Data | wb [TK]D-Fender |
| 21:04.04 | krp | asterisk is running fine |
| 21:04.09 | [TK]D-Fender | krp, pastebin your zaptel.conf and zapata.conf |
| 21:04.09 | [TK]D-Fender | ~pb |
| 21:04.10 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 21:04.10 | krp | i can do what i want |
| 21:04.12 | [TK]D-Fender | ^^^^^^^^^^ |
| 21:04.18 | krp | but the callerid is 'unknown' |
| 21:04.24 | krp | [TK]D-Fender: ok |
| 21:04.57 | A-Data | [TK]D-Fender how can i put a user in que and while waiting operator or agent to answer him he hear Music? |
| 21:05.02 | De_Mon | A-Data voip-info.org |
| 21:05.38 | krp | My zaptel.conf: http://pastebin.com/929203 |
| 21:06.35 | [TK]D-Fender | A-Data, go read up on Queues. "show application queue" and look at the queues.conf and agents.conf sample files. |
| 21:06.55 | *** join/#asterisk pulu (i=pulu@wsip-68-98-213-162.ph.ph.cox.net) |
| 21:07.32 | krp | My zapata.conf: http://pastebin.com/929205 |
| 21:07.33 | pulu | wow, this channel is alot bigger than it used to be... umm, there used to be someone from asterlink that lurked here, they still around? |
| 21:07.50 | *** join/#asterisk astronut (n=astronut@sfnc-162-39-87-189.sandhills.us) |
| 21:07.56 | De_Mon | pulu sure! |
| 21:08.20 | Qwell[] | pulu: if it's an asterlink issue, I'd try #asterlink |
| 21:08.24 | astronut | how hard would it be to set up a basic SIP -> POTS gateway so that I could make calls remotely using my home phone? is an old modem sufficient hardware? |
| 21:08.29 | krp | [TK]D-Fender: i see there's a "cidsignalling" parameter in zapata.conf |
| 21:08.34 | pulu | oh, that's nice. alot happens in a year, thanks guys |
| 21:08.43 | krp | there's the 'bell', 'v23', and 'dtmf' options |
| 21:08.50 | krp | but none are for my country (fr) |
| 21:08.58 | [TK]D-Fender | krp, heres a problem : # |
| 21:08.58 | [TK]D-Fender | channel=3 is the THIRD line of your zapata.conf. your definition of things you want for that channel are occuring AFTER it. put all your settings first , THEN channel=3 or it won't apply to that channel |
| 21:09.32 | krp | ah |
| 21:10.07 | krp | trying |
| 21:10.41 | [TK]D-Fender | krp, don't forget you need to reload chan_zap.so or restart * completely |
| 21:10.43 | De_Mon | oh, asterlink.. read it too fast ;) |
| 21:10.49 | krp | same problem |
| 21:10.57 | krp | (i restarted the system fully) |
| 21:11.16 | krp | but if, as you said, the channel=3 line has to appear at the end, it's at least one problem put away |
| 21:11.51 | krp | (i'm using some FastAGI btw) |
| 21:11.56 | krp | and it says in the debug |
| 21:11.57 | [TK]D-Fender | krp, also change the zone data in your zaptel.con |
| 21:11.59 | krp | INFO:FastAGI:agi_callerid = 'unknown' |
| 21:12.21 | [TK]D-Fender | krp, Forget AGI, NoOp it in the dialplan immediately to be sure |
| 21:12.28 | krp | [TK]D-Fender: it's no good ? i have defaultzone=fr |
| 21:12.31 | pulu | unbelievable how many people are in this channel now, kudos to all you helpful asterisk people. See you all in another year. |
| 21:12.56 | krp | wait, don't i need a channel=3 line AFTER everything in zaptel.conf also ? |
| 21:12.58 | [TK]D-Fender | krp, You also have a US in there. |
| 21:13.03 | krp | ok removing the us line |
| 21:13.13 | [TK]D-Fender | krp, no, zaptel doesn't work that way |
| 21:13.17 | krp | ok |
| 21:13.33 | krp | you sure cause read this : |
| 21:13.37 | krp | # # We are all done with our channel parameters, so now we specify what |
| 21:13.37 | krp | # # channels they apply to |
| 21:13.37 | krp | # channels=1-4 |
| 21:13.54 | [TK]D-Fender | krp, yes, I'm sure |
| 21:14.24 | krp | i added it just to see |
| 21:14.29 | krp | and running ztcfg says |
| 21:14.36 | krp | Notice: Configuration file is /etc/zaptel.conf |
| 21:14.36 | krp | line 231: Cannot get number of tones for channel 3 |
| 21:14.36 | krp | line 231: Cannot init tones for channel 3 |
| 21:14.36 | krp | line 231: Cannot set rxtone on channel 3 |
| 21:14.38 | krp | line 231: Cannot set rxtone on channel 3 |
| 21:14.42 | astronut | how hard would it be to set up a basic SIP -> POTS gateway so that I could make calls remotely using my home phone? is an old modem (creative modemblaster) sufficient hardware to get on the POTS network? |
| 21:14.43 | krp | and it continues ... |
| 21:15.20 | [TK]D-Fender | astronut, No your old craptastic modem is worthless for * |
| 21:15.55 | astronut | [TK]D-Fender: it's only for a short time |
| 21:16.06 | astronut | [TK]D-Fender: is it doable with poor quality or impossible? |
| 21:16.27 | [TK]D-Fender | astronut, as for the difficulty in setting up * for this, I'd say not that hard, but you've got a learning curve ahead of you. |
| 21:16.43 | [TK]D-Fender | astronut, No, that modem is UNUSABLE. |
| 21:16.46 | De_Mon | its like using a round peg for a square hole |
| 21:17.16 | astronut | ok, thanks |
| 21:17.25 | astronut | is there something else i can use that for? |
| 21:17.28 | [TK]D-Fender | De_Mon, I wouldn't use that analogy.... I always proved my teachers wrong when they said I couldn't with those ;) |
| 21:17.29 | astronut | err, use it with? |
| 21:17.41 | Capps- | [TK]D-Fender: what would be a better alternative for astronut? |
| 21:17.46 | [TK]D-Fender | astronut, Yes, go to the WIKI and check out the ahrdware compatability list. |
| 21:17.48 | [TK]D-Fender | ~wikis |
| 21:17.48 | jbot | rumour has it, wikis is http://www.voip-info.org |
| 21:18.16 | [TK]D-Fender | Capps-, how about something * can at least USE. ANYTHING would be better than the NOTHING he has now. |
| 21:18.40 | De_Mon | [TK]D-Fender it may go it but it woln't "fit" ! |
| 21:18.45 | Trevor_b | astronut: x100p.com is inexpensive but quality (or at least from my testing it is) |
| 21:19.20 | astronut | bascially - i'm leaving the coutnry tommorow, and earlier had the idea "wouldn't it be great if i could use my modem as a gateway when i'm gone" |
| 21:19.22 | A-Data | [TK]D-Fender patton smartnode 4110 can be used to convert FXO/FXS to ip and then link it to * or i miss understand the situtation? |
| 21:19.45 | krp | [TK]D-Fender: isn't it channel => 3 (and not channel=3) that i need to put ? |
| 21:19.46 | astronut | so that i coul dmake calls from away |
| 21:20.01 | astronut | is there some software that could utilize my modem that way? |
| 21:20.23 | De_Mon | astronut your modem is not supported in asterisk try again |
| 21:20.35 | [TK]D-Fender | krp, I believe either will technically work. |
| 21:20.52 | krp | ok |
| 21:20.52 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 21:20.59 | A-Data | patton smartnode 4110 can be used to convert FXO/FXS to ip and then link it to * or i miss understand the situtation? |
| 21:21.01 | krp | still having an 'unknown' callerid |
| 21:21.53 | astronut | is there some software that could utilize my modem in that way? |
| 21:21.53 | [TK]D-Fender | A-Data, Sure, though there are other models I'd sooner suggest dependiong on your needs |
| 21:21.55 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 21:22.20 | [TK]D-Fender | astronut, No. It is 100% useless. PERIOD. |
| 21:22.39 | [TK]D-Fender | astronut, Get over the thought that you're not going to have to buy suported hardware. |
| 21:22.55 | A-Data | [TK]D-Fender the only proplem for modles is that not all models or manfuctures have reseller in my home country (Egypt) |
| 21:23.00 | [TK]D-Fender | s/not// |
| 21:23.39 | [TK]D-Fender | A-Data, if you say so. But it should work I'll figure. |
| 21:23.58 | astronut | ok, thanks |
| 21:24.04 | astronut | it seems vgetty supports it as an answering machine |
| 21:24.22 | irule | I cant find much documentation for queues! |
| 21:24.22 | astronut | i'll look around and see if there's somethign vgetty based that can do it |
| 21:24.25 | irule | whats up? |
| 21:24.29 | astronut | it's just a thought |
| 21:24.32 | A-Data | [TK]D-Fender can i give u the product link and if u don`t mind tell me wither it will work or not? |
| 21:24.37 | [TK]D-Fender | astronut, Zaptel does not support it. End of story. Forget any other illusions you may have on it usability. |
| 21:24.40 | astronut | all: thanks for your help |
| 21:24.43 | irule | can I get some clues on that to check for? |
| 21:24.55 | astronut | ~zaptel |
| 21:24.56 | jbot | it has been said that zaptel is zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access. |
| 21:24.58 | De_Mon | astronut good luck |
| 21:25.02 | astronut | De_Mon: thanks |
| 21:25.05 | astronut | it's just a thought |
| 21:25.17 | astronut | cignular hasnt' sent me my unlock code so i can't get a prepaid SIM for travel |
| 21:25.33 | A-Data | this is specs http://www.patton.com/products/pe_products.asp?category=328&tab=sp&MiDAS_SessionID=f120972672a7459e8bf777d234945b5a can you find for me it will work or not |
| 21:25.34 | [TK]D-Fender | irule, Wiki + sample configs + book. |
| 21:25.57 | De_Mon | <PROTECTED> |
| 21:26.04 | irule | TK]D-Fender wiki is empty |
| 21:26.08 | De_Mon | oh I see it |
| 21:26.23 | *** part/#asterisk astronut (n=astronut@sfnc-162-39-87-189.sandhills.us) |
| 21:26.47 | [TK]D-Fender | irule, empty.... NO, it isn't |
| 21:26.47 | *** part/#asterisk apogee7 (n=steven_h@74.92.229.209) |
| 21:27.17 | [TK]D-Fender | A-Data, Looks fine. |
| 21:27.35 | *** join/#asterisk _DAW (n=chatzill@adsl-241-93-3.msy.bellsouth.net) |
| 21:27.43 | A-Data | [TK]D-Fender thanks alot |
| 21:27.47 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 21:28.05 | irule | as if my understanding, users.conf is part of the whole queue enchilada, http://www.voip-info.org/wiki-Asterisk+config+users.conf tsk tsk |
| 21:28.44 | irule | [K]D-Fender or am I mistaken? |
| 21:29.01 | krp | [TK]D-Fender: getting |
| 21:29.02 | krp | Jun 14 22:28:46 NOTICE[12961] chan_zap.c: Got event 18 (Ring Begin)... |
| 21:29.03 | krp | Jun 14 22:28:47 ERROR[12961] callerid.c: fsk_serie made mylen < 0 (-17) |
| 21:29.03 | krp | Jun 14 22:28:47 WARNING[12961] chan_zap.c: CallerID feed failed: Success |
| 21:29.03 | krp | Jun 14 22:28:47 WARNING[12961] chan_zap.c: CallerID returned with error on channel 'Zap/3-1' |
| 21:29.06 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:29.27 | [TK]D-Fender | ikrp : check that zaptel error concerning the FSK delay for UK CID |
| 21:29.36 | [TK]D-Fender | krp, : may apply to your area |
| 21:29.49 | [TK]D-Fender | irule, users.conf = flaming pile 'o' shit. |
| 21:30.03 | krp | [TK]D-Fender: thank you |
| 21:30.04 | [TK]D-Fender | irule, and unrelated to your previous request. |
| 21:33.52 | tzafrir_laptop | is UK caller ID used in France as well? |
| 21:34.05 | tzafrir_laptop | (v23?) |
| 21:34.31 | krp | it seems so |
| 21:34.49 | krp | but maybe it comes from the cellphone i'm using |
| 21:34.59 | krp | i'll try from another phone |
| 21:35.39 | kvidell | yes? |
| 21:35.48 | tzafrir_laptop | Where is it connected to? The caller ID comes from the FXS interface you're connected to (e.g: the telco) |
| 21:36.07 | kvidell | oh, wrong room, hah |
| 21:36.16 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 21:36.25 | krp | tzafrir_laptop: it's connected directly to the telco |
| 21:36.40 | *** join/#asterisk Simon- (i=simon@proxima.lp0.eu) |
| 21:36.41 | ReDNeQ | is it possible to link 2 asterisk servers using PSTN channels.. |
| 21:36.54 | ReDNeQ | and would it be a perm link? or dial on demand? |
| 21:37.05 | tzafrir_laptop | is there a pasebin of your current zapata.conf ? |
| 21:37.06 | krp | tzafrir_laptop: coming |
| 21:37.28 | krp | tzafrir_laptop: http://pastebin.com/929220 |
| 21:37.47 | *** part/#asterisk Nuitari (n=nuitari@142.46.207.230) |
| 21:38.14 | Simon- | for some reason asterisk is completely ignoring my phone after asking it to authenticate: <- 101 INVITE, -> 407, <- 101 ACK, <- 102 INVITE... the second invite(s) don't even appear with sip debug enabled |
| 21:39.04 | tzafrir_laptop | what happens if you rem-out 'cidsignalling=v23 cidstart=ring' and reload? |
| 21:39.34 | krp | trying |
| 21:39.45 | krp | i keep the cadence parameter ? |
| 21:39.47 | [TK]D-Fender | ReDNeQ, link how? to do what? |
| 21:40.55 | krp | tzafrir_laptop: same |
| 21:41.08 | krp | tzafrir_laptop: but how could i really see what's the callerid seen by asterisk ? |
| 21:41.30 | ReDNeQ | TK to handle calls between locations |
| 21:41.32 | tzafrir_laptop | duh. You have immediate=yes |
| 21:41.37 | krp | yeah |
| 21:41.38 | tzafrir_laptop | why do you need it? |
| 21:41.42 | krp | i can remove it |
| 21:41.52 | krp | it's quicker to answer this way ;) |
| 21:42.15 | [TK]D-Fender | ReDNeQ, that is a grossly generic term. No IP betweent hem? |
| 21:42.24 | krp | trying without, but should be the same |
| 21:42.26 | tzafrir_laptop | yes, the callerid is the thing causing delays... |
| 21:42.44 | tzafrir_laptop | alternatively, Wait() before answering? |
| 21:42.59 | krp | i do have a Wait,1 |
| 21:43.10 | krp | [ligne1] |
| 21:43.10 | krp | exten => s,1,Wait,1 |
| 21:43.10 | krp | exten => s,n,Answer |
| 21:43.10 | krp | exten => s,n,AGI(agi://127.0.0.1:4573) |
| 21:43.13 | tzafrir_laptop | try without the immediate |
| 21:43.17 | krp | i tried |
| 21:43.17 | krp | same |
| 21:43.30 | krp | but i really have no errors in asterisk's messages log |
| 21:43.37 | ReDNeQ | TK: Of course there is IP between them but the g729 codec and ISP fluctuating is not cutting it |
| 21:43.42 | krp | my agi script works and all |
| 21:43.47 | krp | but caller id is unknown |
| 21:43.53 | krp | at least from the agi script's perspective |
| 21:44.15 | tzafrir_laptop | To see the callerid: NoOp(callerid=${CALLERID(all)}) |
| 21:44.23 | tzafrir_laptop | I believe |
| 21:44.24 | [TK]D-Fender | ReDNeQ, you should be able to afford a seperate DSL connection in north-america for less than the lines would cost you..... |
| 21:44.38 | tzafrir_laptop | add that in a dialplan line somewhere: |
| 21:45.00 | tzafrir_laptop | exten => s,n,NoOp(callerid=${CALLERID(all)}) |
| 21:45.05 | [TK]D-Fender | ReDNeQ, But no, it would not be continuous. If youw be a boring call like any other, with no more intelligence about how to handle that call than you can do normally. |
| 21:45.07 | krp | added, trying |
| 21:45.08 | Simon- | why would asterisk completely ignore some sip messages? |
| 21:46.04 | krp | tzafrir_laptop: will this line print it or not ? |
| 21:46.18 | ReDNeQ | TK maybe i should state it this way.. Location 1 has 6 lines they have the main number. Thye have added a second locaiton across town.. It has 6 new lines but none of the numbers are known. When we transfer calls |
| 21:46.41 | ReDNeQ | between buildings its using VPN/SIP and ofcourse its not happening so we want ot add another * box |
| 21:46.44 | krp | <PROTECTED> |
| 21:46.47 | ReDNeQ | and offset the ISP traffic |
| 21:46.49 | *** join/#asterisk sharp (n=sharp@pool-72-94-209-98.phlapa.east.verizon.net) |
| 21:47.15 | tzafrir_laptop | so it got no callerid |
| 21:47.15 | ReDNeQ | thinking I could use the phone lines to try and offset this by dedicating maybe 1 of the ports on both sides to hanndle |
| 21:47.41 | [TK]D-Fender | ReDNeQ, just think about how * can TREAT that line. its as dumb as it is for your clients calling in. |
| 21:48.12 | tzafrir_laptop | Try a Wait(5) there. More than needed, but to eliminate any doubt |
| 21:48.21 | krp | tzafrir_laptop: alright |
| 21:49.22 | krp | same |
| 21:50.15 | ReDNeQ | TK: right so maybe im not clear.. they best would be to set diaplan that if ext X use PSTN 6 and on other side if call from X use port 6 incoming? |
| 21:52.13 | A-Data | <PROTECTED> |
| 21:52.23 | krp | sucks :( |
| 21:52.54 | [TK]D-Fender | ReDNeQ, Still not terribly clear.. |
| 21:52.57 | ReDNeQ | hehehe |
| 21:53.06 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 21:53.19 | ReDNeQ | ok is there a way to dedicate 1 line to call into a specific line on another * |
| 21:54.11 | A-Data | <PROTECTED> |
| 21:54.27 | krp | tzafrir_laptop: http://www.voip-info.org/wiki/view/CID+Issues+with+some+Siemens+DECT+phones+in+France |
| 21:55.43 | tzafrir_laptop | that is sending CID to a phone, right? |
| 21:55.49 | krp | yeah |
| 21:56.15 | tzafrir_laptop | you don't need to ring any phone. You need to read a caller ID |
| 21:56.40 | krp | no it's talking about the different ring cadence |
| 21:56.49 | krp | i have the line he's talking about |
| 21:57.18 | tzafrir_laptop | this page advises the telco what to do. You don't generate a ring: you listen to it |
| 21:59.49 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 22:01.13 | krp | well, maybe the line is not allowed to transmit the cid signal |
| 22:01.21 | krp | i'm not sure |
| 22:02.44 | *** join/#asterisk perf3kt (n=perf3kt@adsl-68-73-150-167.dsl.ipltin.ameritech.net) |
| 22:03.55 | irule | how can I save one variable per sip device that will be there persistently for use by its channel? |
| 22:05.52 | lowlevel | hmm, i need some sip phones in a couple months.. wheres the best place to order form in canada/toronto? |
| 22:06.01 | lowlevel | say for polycom/cisco |
| 22:06.28 | Hymie | I have a dude in Ottawa, but you'd probably find an ok place in Toronto somewhere |
| 22:06.44 | Hymie | I just setup about 14 polycom 501s, quite pleased |
| 22:06.45 | lowlevel | I know a guy, but he's never around |
| 22:06.52 | lowlevel | 501 looks gay |
| 22:06.52 | lowlevel | :/ |
| 22:07.06 | lowlevel | I like the soundpoint ip330's I guess |
| 22:07.10 | Hymie | asterisk 1.2.x blows chunks for echo cancellation, if you're juset using analog |
| 22:07.11 | lowlevel | wayyy cheaper tho |
| 22:07.19 | [TK]D-Fender | lowlevel, far from. I could pick plenty far worse, and many of the pics don't do them justice |
| 22:07.22 | Hymie | 1.4.x destroyed the echo fine, here |
| 22:07.29 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.13) |
| 22:07.34 | lowlevel | dfender : so they DO look beter ein real life eh |
| 22:07.35 | lowlevel | hehe |
| 22:07.46 | Hymie | lowlevel: they look fine here.. you know, do what I did |
| 22:07.56 | Hymie | lowlevel: order one, overnight... test / config it, make sure you like it |
| 22:07.59 | Hymie | then do the big order |
| 22:08.05 | lowlevel | yeah, thats a good plan |
| 22:08.25 | lowlevel | I only need 4 or 5 tho.. |
| 22:08.33 | *** join/#asterisk angom (n=angom@red-corp-201.143.81.252.telnor.net) |
| 22:08.34 | Hymie | whatever you do, don't order those uniden anchors |
| 22:08.35 | Hymie | heh |
| 22:08.39 | [TK]D-Fender | lowlevel, I have not seen a cost-effective Polycom reseller in Canada. better off going throught he US. |
| 22:08.40 | lowlevel | ok, noted. |
| 22:08.41 | Hymie | worst phones in the entire universe ;) |
| 22:08.51 | [TK]D-Fender | Hymie, agreed |
| 22:08.58 | Hymie | [TK]D-Fender: I just got some for $219 per |
| 22:09.05 | lowlevel | dfender: yeah just dont wanna get corn holed by UPS |
| 22:09.07 | [TK]D-Fender | Hymie, What model? |
| 22:09.11 | Hymie | 501s, to my door |
| 22:09.18 | lowlevel | 219 per eh |
| 22:09.20 | lowlevel | thats good |
| 22:09.21 | [TK]D-Fender | Hymie, .... thats not impressive :/ |
| 22:09.32 | lowlevel | prices I'm seeing ar like $448 cdn from CDW |
| 22:09.39 | Hymie | [TK]D-Fender: we only ordered 12, there was no delivery charge, and the guy's local for warrenty... |
| 22:09.48 | [TK]D-Fender | its +/- 170$ USD ea, and thanks to GWB, thats about PAR |
| 22:09.53 | Hymie | lowlevel: ?! for the 501?? |
| 22:09.59 | lowlevel | hymie: yeah! |
| 22:10.02 | lowlevel | and 601 |
| 22:10.03 | [TK]D-Fender | lowlevel, CDW = BS |
| 22:10.05 | lowlevel | is like $700 |
| 22:10.07 | lowlevel | heh |
| 22:10.10 | lowlevel | CDW = ASS |
| 22:10.16 | lowlevel | I dont' know why I deal with them |
| 22:10.50 | Hymie | [TK]D-Fender: yes, but after I pay shipping, and dealing with any warrantey issues... this is more of a "local store versus online web ordering" price thing |
| 22:10.51 | Hymie | I mean |
| 22:10.56 | Hymie | the margin is so sad |
| 22:10.57 | [TK]D-Fender | ok, off to martial arts, back in many hours time... |
| 22:11.00 | lowlevel | *shrug*, order a spare? |
| 22:11.01 | lowlevel | heh |
| 22:11.01 | lowlevel | ;) |
| 22:11.06 | lowlevel | later d-f |
| 22:11.24 | [TK]D-Fender | lowlevel, http://www.ccpin.com/ <--- call them up and ask for a better price |
| 22:11.24 | Hymie | well, I just want to see my local reseller make $20 per phone, I mean... that's not really a lot |
| 22:11.52 | [TK]D-Fender | lowlevel, CCP is good (and my reseller for them). |
| 22:12.05 | [TK]D-Fender | lowlevel, and they will send you a demo |
| 22:12.11 | thekidrio | anyone know a mp32gsm converter? (command line preferred) |
| 22:12.15 | lowlevel | thanks d-f, I'll check them out |
| 22:12.38 | Hymie | hmm |
| 22:12.49 | Hymie | what are you guys using for headsets, for the polycoms |
| 22:13.00 | Hymie | I mean, I see absurd pricing of $150 for them on some sites :D |
| 22:13.06 | lowlevel | I got a polycom 1000vtx a while back |
| 22:13.14 | [TK]D-Fender | Hymie, Yup.. they're woth it :0 |
| 22:13.45 | [TK]D-Fender | Hymie, IP 600 + Plantronics M22 Amp + H261 Binaural Polaris quick-connect headset. |
| 22:13.55 | [TK]D-Fender | ok, I'm off |
| 22:15.00 | lowlevel | gave notice at my apartment today |
| 22:15.05 | lowlevel | kind of stressful |
| 22:15.06 | lowlevel | heh |
| 22:15.29 | lowlevel | back later |
| 22:17.11 | Hymie | why the amp, /me wonders |
| 22:17.13 | irule | how can I save one variable per sip device that will be there persistently for use by its channel? |
| 22:17.40 | Hymie | $GLOBAL_VAR_SIP1 $GLOBAL_VAR_SIP2 etc? |
| 22:18.47 | Hymie | thekidrio: sox? |
| 22:19.07 | thekidrio | Hymie, yeah hehe silly I did not think of that |
| 22:19.20 | thekidrio | i was using gstreamer, but the quality was horrid |
| 22:19.30 | Hymie | funny |
| 22:19.33 | thekidrio | #ubuntustudio helped me out :) |
| 22:19.34 | Hymie | we have the "GST" here |
| 22:19.36 | Hymie | and everyone hates it |
| 22:19.41 | Hymie | and, you have the GSTreamer ;P |
| 22:19.45 | thekidrio | hahaha |
| 22:22.00 | thekidrio | anyone know if asterisk caches the sounds directory? |
| 22:22.16 | krp | tzafrir_laptop: thanks for your help anyway |
| 22:22.21 | krp | much appreciated |
| 22:26.23 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 22:26.24 | thekidrio | fyi asterisk does not appear to cache the sounds dir |
| 22:27.18 | *** join/#asterisk lwh (n=lwh192@66.212.165.24.tor.pathcom.com) |
| 22:31.41 | tzafrir_laptop | thekidrio, asterisk supposed to cache sound files? |
| 22:31.43 | tzafrir_laptop | thekidrio, sounds generated from a text-to-speech engine? |
| 22:31.54 | thekidrio | i did not if it was supposed to |
| 22:31.57 | thekidrio | I was just unsure |
| 22:32.05 | thekidrio | and no not generated |
| 22:32.15 | thekidrio | its an mp3/gsm file |
| 22:32.57 | *** part/#asterisk Simon- (i=simon@proxima.lp0.eu) |
| 22:33.40 | tzafrir_laptop | thekidrio, again: sounds generated from a text-to-speech engine? |
| 22:33.40 | tzafrir_laptop | ah, ok |
| 22:33.40 | tzafrir_laptop | it's not asterisk's job to cache sound files |
| 22:33.42 | tzafrir_laptop | it's the OS/kernel's job |
| 22:33.46 | tzafrir_laptop | to test for caching, try cat file >/dev/null |
| 22:33.48 | tzafrir_laptop | /dev/null doesn't care |
| 22:34.25 | tzafrir_laptop | make that: time cat path/to/sound.mp3 >/dev/null |
| 22:34.32 | tzafrir_laptop | run it twice |
| 22:34.34 | thekidrio | why would it matter if it is generated by anything? |
| 22:34.53 | tzafrir_laptop | If the second time is much faster, it wasn't cached and it is cached now |
| 22:35.25 | thekidrio | and more clearly, I was wondering if it cached the sounds directory itself was cached in a config or sooooooomething so that if i added a new directory I was wondering if i needed to reload something in asterisk |
| 22:35.33 | thekidrio | please forgive repeats, on very very old keyboard atm |
| 22:36.06 | tzafrir_laptop | because that would be a different kind of caching (saving the work of the text-to-speech engine) |
| 22:36.29 | tzafrir_laptop | no |
| 22:37.14 | tzafrir_laptop | asterisk will check the "disk" on every time you play. But then again, the tirectory itself would probably be cached in memory |
| 22:37.56 | thekidrio | tzafrir, yeah that is what I was wondering. I was not sure if I needed to explicitly tell asterisk that thiiiiiiis new directory existed |
| 22:38.37 | thekidrio | however I was able to just create the directory with proper permissions and slap in any old mp3 (right now using linus's old soundblaster test) and voila |
| 22:39.30 | tzafrir_laptop | in short: nothing to worry about and nothing to do |
| 22:40.21 | blitzrage | anyone here figure out how to make MeetMe() not play MoH when a phone inside the conference goes on hold? |
| 22:41.30 | Mercestes | blitzrage, Did you try a rem -dvfr *.mp3? >.> |
| 22:41.46 | blitzrage | yep, I did |
| 22:41.48 | russellb | blitzrage: honestly, if you have a phone directly connected to the box running meetme and it does that, it's a bug |
| 22:41.51 | Mercestes | Didn't work huh? |
| 22:41.54 | russellb | i really don't know why it would happen |
| 22:42.00 | thekidrio | tzafrir, heh hrmm yes but putting it that way makes me feel lame haha |
| 22:42.13 | blitzrage | russellb: ok... I'll post a bug, just wanted to see if someone has figured out a work around :) |
| 22:42.32 | Mercestes | blitzrage, Fire the person putting the conference on hold |
| 22:42.48 | russellb | blitzrage: cool |
| 22:43.42 | Trevor_b | anyone done any testing with oslec yet? |
| 22:45.07 | *** join/#asterisk plut0 (n=plut0@cpe-24-25-137-173.nycap.res.rr.com) |
| 22:45.07 | `Sean | russellb |
| 22:45.10 | `Sean | who owns Jbot? |
| 22:45.29 | Mercestes | no one owns jbot. He was proven to be a sentient being in a court of law. |
| 22:45.30 | russellb | jbot: who owns you? |
| 22:45.31 | jbot | TimRiker does |
| 22:45.44 | russellb | there ya go :) |
| 22:46.11 | `Sean | hrmp now when does he come around to getting on irc |
| 22:46.23 | `Sean | ah hes online atm |
| 22:46.23 | `Sean | :) |
| 22:46.33 | `Sean | just not in asterisk tho |
| 22:47.02 | Mercestes | Has Jbot been bad again? |
| 22:49.07 | blitzrage | anyone know how to make the Linksys SPA-942 not react to star codes and just pass them through instead of acting on them? |
| 22:50.19 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 22:54.41 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:58.48 | *** join/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
| 22:58.55 | Simon-- | so asterisk 1.4 parses ;- as not a comment |
| 22:58.56 | Simon-- | nice ;) |
| 22:59.28 | Qwell[] | really? |
| 22:59.32 | Mercestes | blitzrage, Remove them from that silly *-map thing they have in there. Or dsiable the features. |
| 22:59.40 | blitzrage | Mercestes: did that -- still matches on them |
| 22:59.45 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 22:59.54 | Mercestes | blitzrage, really? hrm. did you add them to the digitmap? |
| 23:00.02 | blitzrage | where is there? |
| 23:00.19 | Mercestes | the digitmap? Dont' rememberly exactly where.... |
| 23:02.32 | *** join/#asterisk plla (n=nekomimi@corporacionlely.com.pe) |
| 23:02.54 | plla | Hello |
| 23:03.20 | plla | Is this the right place to ask for help when google has failed me? |
| 23:03.35 | blitzrage | aha... found it |
| 23:04.55 | Qwell[] | plla: sure |
| 23:05.11 | plla | I am using the SVN version of Asterisk in a testing environment. |
| 23:05.35 | plla | I have configured realtime with postgresql back end. |
| 23:06.34 | plla | I found that if I turn off the cache of the realtime |
| 23:06.46 | xkev | simon--, and this is why I still run cvs from 2005 :) |
| 23:06.53 | *** part/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 23:07.03 | xkev | random dumbshit things change and break stuff |
| 23:07.04 | plla | postgres goes up to 75% ram and asterisk 25% when someone registers. |
| 23:07.29 | plla | when normally it's in nothing. |
| 23:08.25 | plla | I turned on the debug and the log is filled with asterisk updating the user state infinite times. |
| 23:08.37 | Qwell[] | Simon--: what exactly do you have on that line? |
| 23:08.49 | Qwell[] | is it perhaps ;-- ? |
| 23:09.09 | Simon-- | Qwell[]: yah, like ;-- wooooooooooot this section kicks ass ------------------ |
| 23:09.15 | Qwell[] | yeah, that's a block comment |
| 23:09.22 | Qwell[] | close it with --; |
| 23:09.22 | Simon-- | it says "unterminated comment on line 23419419412" now |
| 23:09.25 | Simon-- | yeah |
| 23:09.29 | Simon-- | sqlism :) |
| 23:09.53 | Simon-- | ok, "don't do that then" accepted |
| 23:10.15 | Qwell[] | line 23419419412, seriously? |
| 23:10.18 | rene- | hey |
| 23:10.20 | Simon-- | yah, I think that's buggy |
| 23:10.22 | Qwell[] | heh |
| 23:10.43 | rene- | what is the dialstatus of a Dial like Dial(Sip/1&Sip/2&...) |
| 23:10.49 | Simon-- | [Jun 14 15:55:04] WARNING[16630] config.c: Unterminated comment detected beginning on line 1100702820 |
| 23:10.50 | rene- | say i get connected to sip/1 |
| 23:11.11 | Qwell[] | looks like a pointer address |
| 23:11.14 | rene- | what happens to DIALSTATUS ? |
| 23:11.33 | plla | the peer keeps getting destroyed and being recreated in a matter of seconds. |
| 23:11.44 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:12.01 | plla | I have Asterisk 1.4.1 with the same configuration and this doesn't happen. |
| 23:12.40 | plla | I am trying to find what have changed on the source but I would appreciate some pointers on where to find it. |
| 23:15.26 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 23:15.37 | Mercestes | plla: Check sip debug. Is there any NAT involved or any funny qualify times?? |
| 23:15.52 | Mercestes | multiple databases maybe? |
| 23:15.57 | plla | nope, not nat. only one database. |
| 23:16.06 | plla | it's a local database. |
| 23:16.18 | blitzrage | rene-: DIALSTATUS == ANSWER |
| 23:17.28 | rene- | blitzrage: what about all the other dial attempts? |
| 23:18.42 | rene- | i mean say sip/2 was busy, and sip/3 was no answer, or worse, sip/1 unavail, sip2/busy and sip/3 no answer |
| 23:19.04 | Mercestes | plla, Check sip debug then, see what the phone is doing. |
| 23:19.47 | thekidrio | Jun 14 03:41:46 NOTICE[4817] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) is this a NAT issue? |
| 23:19.56 | rene- | no |
| 23:20.01 | rene- | it might be |
| 23:20.12 | rene- | it just means asterisk doesnt know about your sip device |
| 23:20.15 | blitzrage | rene-: Verbose(1|${DIALSTATUS}) in the 'h' extension -- it's easy to find out what it says |
| 23:20.22 | rene- | a.k.a it is not registered |
| 23:20.31 | Mercestes | thekidrio, It means there is no route to the destination |
| 23:20.42 | thekidrio | wow you are so smart Mercestes |
| 23:20.45 | rene- | blitzrage i didnt knew about a that |
| 23:20.52 | rene- | thats really cool |
| 23:20.55 | blitzrage | I'm going grocery shopping, lates |
| 23:20.58 | Mercestes | thekidrio, Could be nat, could be phone offline, could be an unroutable IP, could be a monkey with a pair of scissors |
| 23:21.01 | rene- | thx |
| 23:21.11 | thekidrio | damn it, how did you know |
| 23:21.23 | Mercestes | Does your phone show as online?? |
| 23:21.23 | Qwell[] | Simon--: fixed :P |
| 23:21.25 | thekidrio | well the phone is offline so that makes sense |
| 23:21.31 | Mercestes | There you go. |
| 23:21.37 | Mercestes | That *could* be nat or 100 other things. |
| 23:21.41 | Mercestes | But, if it's natted, I'd check that first. |
| 23:21.46 | thekidrio | I am trying to figure why it is not dumping to vm |
| 23:21.57 | Mercestes | Did you tell it to? |
| 23:22.02 | thekidrio | yup |
| 23:22.07 | Mercestes | beat it. |
| 23:22.12 | Mercestes | then tell it again. |
| 23:22.23 | Mercestes | pastebin your dialplan and lemme take a look |
| 23:22.46 | *** join/#asterisk riddlebox (n=victoria@75-132-215-110.dhcp.stls.mo.charter.com) |
| 23:22.56 | thekidrio | exten => 1234,1,Dial(SIP/foo 30) exten => 1234,2,VoiceMail(2222@blah) |
| 23:23.10 | Mercestes | Sorry, that's not a valid pastebin link. |
| 23:23.26 | thekidrio | it works if the phone is registered and no one picks up, but not if the phone is offline |
| 23:23.41 | thekidrio | Mercestes, sorry seemed silly to pastebin 2 lines |
| 23:23.51 | Mercestes | Try 1234,101,Voicemail(2222@blah) or a nice goto(1234-${dialstatus} and then a 1234-UNAVAILABLE,1,Voicemail(2222@blah) |
| 23:24.06 | riddlebox | hrmm I called another asterisk server which had an auto attendant, and I was unable to press any digits, I could hear the the dtmf, but the other end never responds? |
| 23:24.34 | Mercestes | I don't think "no route to host" results in a priority+1, it is either +101 or UNAVAILABLE. |
| 23:24.45 | Mercestes | depending on if you enabled priority jumping or not. |
| 23:24.56 | thekidrio | Mercestes, ahhh, that was my dumistake |
| 23:25.19 | Mercestes | your welcome |
| 23:25.31 | thekidrio | err mistake, ${dialstatus} worked thank you so mmuch!! |
| 23:25.44 | Mercestes | riddlebox, Answer before you press buttons, canreinvite=yes, dtmfmode = auto |
| 23:25.45 | thekidrio | still strange that it worked for the other extension though |
| 23:25.57 | Mercestes | np. :) |
| 23:27.14 | riddlebox | Mercestes, but I can call any other IVR system and am able to navigate through their menus, but this other asterisk server I cannot do? |
| 23:27.36 | Mercestes | riddlebox, and what does that tell you? |
| 23:28.07 | Mercestes | riddlebox, and I bet I can find atleast one other IVR you can't dial through. |
| 23:28.14 | riddlebox | Mercestes, I can call the other asterisk server, with my cell phone and it will work? |
| 23:28.27 | Mercestes | riddlebox, ok. |
| 23:29.23 | Mercestes | riddlebox, The options you need on your end are "canreinvite=yes" and "dtmfmode=auto" If you insist on not using auto then I suggest rfc-2833 but you probably already have that and they want inband because your probably reaching them via sip, right? |
| 23:29.39 | Mercestes | and on the remote end you have to call Answer() before you start hearing DTMF tones. |
| 23:29.57 | riddlebox | Mercestes, I am calling them using a broadvoice account, and they also have a broadvoice account |
| 23:30.09 | Mercestes | Regardless of the symptoms those are the answers to the DTMF issue. Unless those are already set, then you have a much bigger issue. |
| 23:30.35 | Mercestes | riddlebox, ok. that would definately fit my suspicion |
| 23:31.18 | riddlebox | Mercestes, those options would be set in sip.conf right? |
| 23:31.41 | *** join/#asterisk pfn_cIc (n=pfnguyen@64.235.249.50) |
| 23:34.28 | riddlebox | Mercestes, dtmf=auto did the trick |
| 23:34.44 | Mercestes | :) |
| 23:34.56 | plla | ok, I checked most of the stuff. |
| 23:35.06 | riddlebox | but now that extension cannot do anything in the voicemail |
| 23:35.10 | plla | The phone just sends a register and a deadlock begins. |
| 23:35.36 | plla | starts with this: |
| 23:35.38 | plla | [Jun 14 13:27:17] DEBUG[3252] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER |
| 23:35.54 | plla | at then infinite queries to the database |
| 23:37.04 | Mercestes | riddlebox, weird. |
| 23:37.12 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 23:37.30 | Mercestes | plla: What is the query? |
| 23:37.44 | riddlebox | Mercestes, this is on a sipura 2100 with an analog phone connected to it |
| 23:38.24 | Mercestes | oh...hrm. Try inband v/s rfc-2833 and see which one your broadvoice friend is looking for. |
| 23:38.53 | riddlebox | Mercestes, I did it with rfc-2833 but could not do anything in my vm as well |
| 23:39.07 | Mercestes | waht did you have dtmf set to before? |
| 23:40.01 | riddlebox | Mercestes, inband |
| 23:40.01 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 23:40.45 | Mercestes | Shouldn't be necessary. |
| 23:40.52 | Mercestes | rfc-2833 and auto should work for VM> |
| 23:41.01 | Mercestes | Is there a DTMF settign in the sipura? |
| 23:42.10 | Mercestes | riddlebox, Should not require inband to call VM, that's weird. |
| 23:42.15 | Mercestes | riddlebox, however, here is a work around. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPdtmfmode |
| 23:42.28 | Mercestes | jsut change your DTMFmode when you call broadvoice. |
| 23:45.06 | *** join/#asterisk punkgode (n=Punkgode@rev-200-40-119-222.netgate.com.uy) |
| 23:46.01 | Mercestes | Goodnight |
| 23:47.47 | punkgode | hi, does anyone tried to setup the internal asterisk database into Mysql. My primary concern is performance, is the connection overhead for each query a big deal compared to Berkeley? |
| 23:48.55 | plla | Mercestes: http://pllamosas.googlepages.com/messages_cut.txt this is part of the log. |
| 23:51.25 | punkgode | my load is quite low, mainly queue activity, and phone number-channel mappings. I'm using dynamic agents implemented with Local channels. I wonder if it's a good thing to do, moving to mysql to do such things |
| 23:52.27 | plla | if I de-register the loop stops. |
| 23:53.05 | *** join/#asterisk bjohnson (n=bjohnson@i209-195-85-119.cia.com) |
| 23:54.09 | plla | I tried playing with the values of rtcachefriends and rtupdate. It works right if I disable both. |
| 23:55.20 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.us.warpdriveonline.com) |
| 23:55.38 | plla | rtcachefriends=on rtupdate=off (that works) |
| 23:55.58 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 23:56.04 | plla | rtcachefriends=off rtupdate=on (falls into infinite loop) |
| 23:56.11 | plla | rtcachefriends=off rtupdate=off (falls into infinite loop) |
| 23:56.45 | plla | rtcachefriends=on rtupdate=on (falls into infinite loop) |
| 23:58.18 | plla | I don't require the register information that badly but still infinite loops = bad. For now I will just use rtcachefriends=on rtupdate=off |