00:01.39 | *** join/#asterisk yashax (~yasha_x@69.15.218.218) |
00:02.02 | yashax | guys, what is the "best" distro to lead asterisk? |
00:02.23 | tessier_ | yashax: Which is better, vi or emacs? |
00:02.45 | tessier_ | yashax: Jew or Muslim? Protestant or Catholic? Ford or Chevy? Budweiser or Miller? |
00:02.47 | yashax | hahaha.. I knew someone would say something like this.... |
00:02.58 | tessier_ | Trojan or Durex? |
00:03.00 | tessier_ | (Trojan) |
00:03.04 | yashax | :) |
00:03.10 | tessier_ | Ultra-thin spermicidal please. |
00:03.16 | yashax | what would YOU recommend? |
00:03.17 | iMediax | use your favorite distro |
00:03.21 | tessier_ | I like to feel it but I don't want no kids. |
00:03.29 | yashax | RH9, FC1? |
00:03.33 | iMediax | sure |
00:04.17 | yashax | What kind of install do I need to do... most likely no X-windows, what else not to install? |
00:04.47 | tessier_ | I've got two boxes of 36 each in my closet. I'm prepared in case a bus load of hot chicks show up at my house and demand that I service each and every one of them. |
00:05.26 | iMediax | lol |
00:05.38 | yashax | I am happy to hear that humor is in the room, but to get back to reality... :) |
00:05.51 | terrapen | i wish i was more mechanically-oriented |
00:05.56 | terrapen | i need a pickup for my guitar |
00:06.05 | terrapen | and this guy wants to charge me $50 to install it |
00:06.11 | terrapen | and i know it will take him 20min tops |
00:06.14 | tzanger | terrapen: I just buy a guitar with pickups |
00:06.15 | tzanger | :-) |
00:06.25 | terrapen | i have a Martin acoustic |
00:06.30 | terrapen | they don't come with pickups |
00:06.30 | tzanger | I've got a Fender Mustang and a Ibanez V70CE |
00:06.32 | terrapen | well, most don't |
00:06.46 | yashax | guys, anyone? |
00:07.04 | tzanger | yashax: try building |
00:07.10 | tessier_ | I'm not sure I ever have. |
00:07.11 | tzanger | if it doesn't work, you took out too much |
00:07.14 | tzanger | tessier_: ?? |
00:07.20 | terrapen | i'm wanting to go to this open mic night tomorrow |
00:07.24 | tzanger | everyone I know with a guitar cna play |
00:07.26 | terrapen | and i need something that can be plugged in |
00:07.30 | tzanger | some more than others |
00:07.38 | tessier_ | tzanger: Some more than others? |
00:07.49 | tessier_ | Anyone can make a bunch of squeeling noise with a guitar it seems. |
00:07.50 | tzanger | tessier_: well there are different levels of playing ability |
00:08.09 | tessier_ | I'd like to meet someone who can play something that sounds nice. |
00:08.28 | tessier_ | I wish I could play classical spanish guitar. |
00:08.35 | tessier_ | I think that would be a real chick getter. |
00:08.38 | tzanger | tessier_: well yeah but I'm talking something melodic |
00:08.46 | tzanger | tessier_: "sounds nice" is also relative |
00:08.53 | *** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk) |
00:09.03 | tzanger | I play some classic rock and some blues and bluegrass but not much else |
00:09.08 | tzanger | want to learn more blues |
00:09.17 | terrapen | i play country, bluegrass, folk, rock |
00:09.25 | tessier_ | I want to play like El Mariachi |
00:09.28 | tessier_ | (Gypsy Kings) |
00:09.37 | terrapen | not Nashville radio-style country though |
00:09.48 | terrapen | the kind of country i play, you've probably never heard anything like it |
00:09.50 | terrapen | unless you live in texas |
00:09.51 | tessier_ | o/~ Soy un hombre muy honrado, que me gusta lo major. A mujeres no me faltan, ni el dinero ni el amor. o/~ |
00:09.54 | terrapen | or oklahoma |
00:10.00 | tessier_ | mejor |
00:10.01 | __Sparks_ | Can some kind soul tell me what ports beside 5060 and 10000-20000 (UDP) I need to forward to my asterisk box to make sipgate calls work properly! (Currently I cant hear the caller!) |
00:10.15 | yashax | Is there a good doc available somewhere on the installation? Please help! |
00:10.31 | tessier_ | __Sparks_: asterisk is behind nat? Yikes. |
00:10.43 | __Sparks_ | yea :S |
00:10.57 | __Sparks_ | surely port forwading can work!? |
00:11.35 | terrapen | tzanger, where are you located |
00:12.17 | terrapen | bwahahah |
00:12.17 | terrapen | In accordance with NETCOM guidance 2004-11 (https://www.us.army.mil/suite/doc/1229431), AKO has begun stripping attachments with the following suffixes: |
00:12.17 | terrapen | .b64,.bat,.bhx,.ceo,.ce0,.cpl,.dbx,.dll,.dot,.eml,.exe,.hqx,.lnk,.mim,.nch,.ocx, |
00:12.17 | terrapen | .pi,.pif,.scr,.sct,.uue,.uu,.vbe,.vbs,.wsc,.wsf,.wsh,.xxe, and .zip. |
00:12.17 | terrapen | Since this is an Army policy, AKO will not be able to grant exceptions - please do not call the AKO help desk on this issue, as they will not be able to help you. |
00:12.19 | terrapen | We regret any inconvenience. |
00:12.21 | terrapen | AKO removed an attachment due to NETCOM 2004-11 restrictions. |
00:12.30 | terrapen | my buddy tried to send me something |
00:12.32 | tzanger | terrapen: midwestern ontario, canada |
00:12.38 | ariel_ | __Sparks_, the other ports will be for iax2 4569 & 5061 for a 2nd sip account. |
00:12.49 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
00:13.14 | tzanger | terrapen: I dunno about that, I know that kind of country |
00:13.22 | tzanger | I'm not a nashville country kind of guy |
00:13.33 | __Sparks_ | ariel_, Thanks, I will try them now :) |
00:13.44 | terrapen | i hate nashvegas country |
00:13.48 | terrapen | it sucks |
00:13.58 | yashax | Guys, anyone? Is there a good doc available somewhere on the installation? Please help! |
00:14.05 | tzanger | yashax: TRY IT |
00:14.06 | tzanger | jesus |
00:14.09 | tzanger | try a basic install |
00:14.22 | ariel_ | yashax, what would like to do from scratch install? |
00:14.22 | tzanger | you OBVIOUSLY have not enough experience to judge what is and isn't needed |
00:14.30 | tzanger | so get yourself a basic install and GET SOME EXPERIENCE |
00:14.36 | eKo1 | yashax: make && make install |
00:14.37 | tzanger | **THEN** optimize the system |
00:14.38 | yashax | yes, from scratch.... |
00:14.47 | tzanger | premature optimization is the mother of all fuckups |
00:14.59 | ariel_ | yashax, if your starting out get yourself the iso from asterisk@home it will get you started. |
00:15.07 | ariel_ | Then read up on the wiki and |
00:15.13 | ariel_ | ~doc |
00:15.21 | tzanger | a 5G HDD will install practically everything you want outside of X, which you already know you don't want |
00:15.25 | yashax | ariel: Thank you SO much.... |
00:15.26 | tzanger | and I dare you to find a brand new 5G HDD |
00:15.30 | ariel_ | I guess the link is not working. |
00:16.07 | ariel_ | http://www.voip-info.org/wiki-Asterisk, |
00:16.31 | ariel_ | http://asteriskathome.sourceforge.net/ |
00:17.15 | yashax | THANK YOU!!!! |
00:17.31 | *** part/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk) |
00:17.33 | *** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk) |
00:17.44 | *** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk) |
00:20.08 | terrapen | yeah, tzanger, the artists here don't get up to .ca very often |
00:20.27 | *** join/#asterisk Legend (~legend@24.244.142.133) |
00:20.29 | tzafrir_home | actually, the installation is not too complicated. And the installation is generally easy to automate. It is the configuration that is complicated |
00:21.30 | tzafrir_home | And with a 5G you can even have both gnome and kde, to burn precious cpu cycles on |
00:23.02 | *** join/#asterisk Mneumonic (~Mnemonic@206.231.230.230) |
00:23.42 | ariel_ | tzafrir_home, I have a test system running asterisk with only a 4 gig wd drive. |
00:23.54 | *** join/#asterisk neuro_[rus] (~neuro_[ru@212.176.51.231) |
00:23.55 | tzanger | :-) |
00:23.56 | Mneumonic | Anyone know how to set up overhead paging thru the sound card in *? |
00:24.01 | ariel_ | it's a p1 233 with 128mg ram. |
00:24.06 | tzanger | my * installs are nder 500M and that's not trying ot scrape much out |
00:24.10 | tzanger | I could probably get it in 32M |
00:24.13 | ariel_ | Mneumonic, no don't do it. |
00:24.26 | greg_work | ariel_: ? |
00:24.29 | Mneumonic | ariel - why not? is there a better way? |
00:25.07 | greg_work | Mneumonic: configure alsa, Dial(console/dsp) |
00:25.14 | tzanger | yeah, is there a better way? |
00:25.28 | ariel_ | overhead paging I do it via an fxo port and a pager from viking. |
00:25.34 | Mneumonic | how do i configure alsa? |
00:25.37 | tzanger | you'd want a relay on the parallel port or serila to engage a paging relay but anyway |
00:25.44 | fafnir | is that you johnny? |
00:25.50 | Mneumonic | nope |
00:25.51 | fafnir | johnny mnuemonic? |
00:25.56 | fafnir | long time no see! |
00:26.01 | Mneumonic | :) |
00:26.05 | fafnir | heh |
00:26.11 | fafnir | that movie is on ondemand |
00:26.17 | Mneumonic | cool |
00:26.21 | fafnir | for the next 24 hours |
00:26.25 | Mneumonic | that movie was way underbudgeted |
00:26.31 | fafnir | it came out aight |
00:26.35 | ariel_ | great movie |
00:26.38 | fafnir | i liked the dolphin |
00:26.53 | fafnir | heh |
00:26.57 | fafnir | go to joohns |
00:27.07 | fafnir | er jones |
00:27.33 | Mneumonic | you gotta hack your own brain.. the damn dolphin cant help u |
00:28.17 | greg_work | i bought a few $15 computer speakers (the $10 ones had an internal transfomer..), put them in the ceiling, and then ran a 2-pair cable back to my electrial room for each.. one pair carries power from a transformer in the server room, the other pair is connected to a 1/8" jack that goes into the soundcard. each set is plugged into a different output, so i can control volumes of each zone in software (ie, reception area is front, hall |
00:28.17 | greg_work | way is rear L, kitchen is rear R, warehouse is center) |
00:28.58 | Mneumonic | and u have that running thru the sound card? |
00:30.25 | riksta | anyone ever implemented a system with a * box that can have like, tele-marketers working from home, that log in and calls are originated to them? |
00:30.49 | tzanger | nothing official riksta |
00:30.54 | tzanger | wouldn't be too tricky though |
00:31.04 | riksta | yeah im sat here planning a system out |
00:31.07 | tzanger | someone at the toronto asterisk meetup was talking about that |
00:31.13 | riksta | was wondering if anyone has any input |
00:34.39 | greg_work | Mneumonic: yes, thats why i said "the other pair is connected to a 1/8" jack that goes into the soundcard." |
00:38.18 | Darwin35 | what happen to res_sqlite ? |
00:38.35 | *** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk) |
00:41.16 | *** join/#asterisk {zombie} (zombie@soulasylum.penguincare.com.au) |
00:42.11 | *** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk) |
00:44.53 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
00:44.55 | puzzled | hello |
00:45.26 | iMediax | tele-marketing is evil |
00:45.35 | CoaxD | I hate telemarketers |
00:45.40 | *** join/#asterisk qwerp (~abc@219.95.105.74) |
00:45.48 | qwerp | harlo |
00:46.40 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
00:48.26 | stonefly | Does anyone have any recomendations on t1 echo cans? |
00:50.15 | ariel_ | t1 echo cans....wow. |
00:50.31 | qwerp | anybody can help me on a TE110p ? |
00:51.06 | *** join/#asterisk Weezey (~Weezey@206.210.109.226) |
00:51.16 | Weezey | how do I make my SPA-3000 reset? |
00:52.04 | puzzled | stonefly: think on the list a while back the products from tellabs were mentioned. at the time some where available on eBay |
00:53.59 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
00:56.23 | stonefly | puzzled, thanx I check the list out... I was just wondering if any had actually used one.. |
00:56.45 | *** join/#asterisk ReVoK (ReVoK@82.224.60.46) |
00:56.47 | ReVoK | hi |
00:57.38 | puzzled | stonefly: np. if you hook up directly to a telco with e.g. a t1 and go straight over to their tdm network then they will already have the echo cans in place. why do you need them? |
00:58.23 | stonefly | puzzled, really? I'm still getting some bad echo on some incomming calls on a t1 from tellamerica |
00:58.49 | stonefly | The T100p isn't sharing irqs either... |
00:59.23 | puzzled | stonefly: hmm, what was the thing again: if *you* hear echo, it is caused by the other side and vice versa? not sure of an echo can on your side would fix it if I had it right |
01:00.24 | stonefly | I hear echo when I talk, so it is fixable.... |
01:00.39 | tzafrir_home | grep , /proc/interrupts |
01:01.06 | puzzled | stonefly: ok, hope you figure it out. voodoo like changing the card around or using another mobo has helped people in the past iirc |
01:01.09 | stonefly | I've got echocancel, and echocancelwhenbridged set to yes and echotraining=800 |
01:01.20 | tzafrir_home | this will show all of those *evil* irq-sharaes |
01:01.37 | stonefly | tzafrir_home, everything is on its own irq.... |
01:02.17 | stonefly | I'll have to try changing slos, but I don't think changing mb's is possible... |
01:02.25 | stonefly | slos=slots... |
01:02.44 | puzzled | stonefly: for voodoo's sake you may want to try sticking the card in another pci slot or sticking it in an entirely different box just to rule out funky stuff |
01:02.46 | stonefly | umm, I can't change slots.. there is only one pci slot... |
01:03.04 | stonefly | its a 1u server... |
01:04.04 | Moc | YES !!! YES !!! |
01:04.38 | zigman | nice |
01:04.42 | puzzled | Moc: on the list he said he was working on it |
01:04.49 | ariel_ | wounder if he knows this yet? |
01:05.02 | puzzled | an extra $1000 would motivate me :) |
01:05.08 | Moc | puzzled ? what ? |
01:05.17 | *** join/#asterisk IsMe (~some@219.95.224.115) |
01:05.32 | Darwin35 | http://www.konceptusa.com/index.php?page=wifi_phone |
01:05.33 | puzzled | Moc: your "YES" was about the T.38 support? |
01:05.38 | Moc | nope hehe |
01:05.42 | puzzled | ah ok |
01:06.03 | qwerp | tzafrir: harlo.. |
01:06.08 | redder86 | can callgroup be set from the dialplan? |
01:06.33 | riksta | anyone know if any VOIP providers in the UK can do TPS screening on outgoing calls routed through SIP ? |
01:07.11 | stonefly | Does SMP cause problems with echo? |
01:07.18 | stonefly | Does=do |
01:07.33 | stonefly | I can't type today.... it's time to go home... |
01:07.47 | puzzled | stonefly: dunno but can't you test it by booting with a UP kernel? |
01:10.32 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
01:10.41 | puzzled | hi ManxPower |
01:11.16 | stonefly | puzzled, yeah... will do... |
01:11.57 | tzanger | evening manx |
01:12.17 | ManxPower | ~docs |
01:12.19 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
01:13.27 | *** join/#asterisk MrEntropy (~entropy@ppp55-252.lns1.adl2.internode.on.net) |
01:13.28 | MrEntropy | yo |
01:13.40 | tzanger | yo |
01:13.45 | NormAst | Hi all. |
01:13.59 | tzanger | werd norm |
01:14.12 | redder86 | is there a way to set up a pseudo-extension? Something that I can Dial() but that will never answer the call? |
01:15.43 | ManxPower | redder86, what are you trying to accomplish? |
01:17.09 | redder86 | ManxPower: I want to use callgroup/pickupgroup, but I don't like having to key callgroup onto a phone in sip.conf. I'd love it if callgroup could be set via the dialplan. |
01:17.31 | redder86 | ManxPower: so I'd like to Dial() an imaginary SIP phone that has callgroup set. |
01:18.01 | ZX81 | hi NormAst |
01:18.06 | ZX81 | hi all |
01:18.47 | ManxPower | redder86, So you can #8 it? |
01:19.42 | ZX81 | how long till 2PM CST Feb17th? |
01:19.53 | ZX81 | ne1? |
01:20.01 | redder86 | ManxPower: yes |
01:20.12 | redder86 | ManxPower: but it's *8 |
01:20.53 | ZX81 | redder86: app_changrap.c |
01:20.54 | ZX81 | :) |
01:21.00 | ZX81 | but I dunno where it is |
01:21.04 | ZX81 | I need it too |
01:21.10 | ZX81 | so I can do call pickup from manager |
01:21.11 | ZX81 | :) |
01:21.31 | redder86 | ManxPower: I have incoming calls to the "main company number" go to a group of phones, but sometimes people are not there to answer (like on a weekend) and someone in one of the nearby offices where the phone isn't ringing would like to pick up the phone, but doesn't want to run into the "floor" area to answer it. |
01:22.06 | ZX81 | Big News: Asterisk Daily News and Asterisk Documentation Project link up through cross-syndication ( http://www.sineapps.com/news.php ) |
01:22.11 | ZX81 | :) |
01:22.25 | redder86 | ManxPower: right now I have to "key" off of one of the "floor" phones, but I don't like that because the "floor" phones all have their own extensions, and I don't want someone intercepting a direct call. |
01:22.48 | shmaltz | ~seen ManxPower |
01:22.50 | jbot | manxpower is currently on #asterisk (12m 18s). Has said a total of 3 messages. Is idling for 4m 3s |
01:23.04 | shmaltz | Hi, ManxPower |
01:23.22 | ZX81 | /ignore lists? |
01:23.23 | *** part/#asterisk IsMe (~some@219.95.224.115) |
01:23.24 | ZX81 | :) |
01:23.53 | Qwell | ZX81: About 18.5 hours |
01:24.00 | ZX81 | ok cool |
01:24.01 | ZX81 | ta |
01:24.10 | ZX81 | enough time to sleep and wakeup |
01:24.11 | ZX81 | :) |
01:24.16 | *** join/#asterisk Guest^DJ (~some@219.95.224.115) |
01:24.32 | Qwell | I can barely do conversion from PST to CST, so I might be off by an hour, heh |
01:24.37 | ZX81 | :) |
01:25.07 | ZX81 | fkn voicepulse won't accept my money |
01:25.17 | ZX81 | says declined on credit card for $10 |
01:25.21 | ManxPower | ZX81, That's good news. We need more consolidation of Asterisk documentation sites. |
01:25.25 | ZX81 | so I went and used it for $4000 |
01:25.26 | Guest^DJ | hey ZX81 |
01:25.27 | ZX81 | ManxPower: yah |
01:25.29 | ZX81 | hi |
01:25.30 | ZX81 | :) |
01:25.32 | Guest^DJ | hi ManxPower |
01:25.36 | shmaltz | ManxPower, can I ask u a question? |
01:25.39 | ZX81 | Guest^DJ: how are you? |
01:26.00 | ManxPower | shmaltz, You can always ask me a question, but I'm not much in the mood for providing answers at the moment. |
01:26.11 | shmaltz | ManxPower, Do you still have the weather script? |
01:27.21 | ManxPower | shmaltz, All that stuff was donated to the Asterisk Documentation Project because I got tired of answering questions about my site. |
01:27.43 | ManxPower | If you smile REALLY nice I might give you a link to a tar.gz of the entire site that I used to have on line. |
01:28.05 | shmaltz | ManxPower, :):):):):):):):):):):):):):):):):):):)::) |
01:28.25 | ManxPower | http://www.fnords.org/~eric/asterisk/wffs.tar.gz I think |
01:28.32 | shmaltz | . -) |
01:28.34 | shmaltz | . -) |
01:30.36 | ZX81 | and there is a street race outside in a few days |
01:30.37 | riksta | is there a howto on making "agents" that can log into asterisk on the wiki |
01:30.42 | ZX81 | sure |
01:30.43 | ZX81 | wiki |
01:30.47 | ZX81 | ~voip-info |
01:30.48 | jbot | it has been said that voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
01:30.48 | riksta | nice |
01:30.57 | riksta | what search terms would i use |
01:31.45 | riksta | i'm looking to see if you can have agents, that can log out briefly for after call work, or for lunch...is that possible too? |
01:31.46 | ZX81 | agents? |
01:31.47 | ZX81 | maybe |
01:31.48 | ZX81 | :) |
01:31.49 | modulus_ | use terms like "stuff" and "thingy" |
01:31.56 | ZX81 | hehe |
01:32.19 | *** join/#asterisk soulz- (~Soulz-@host-137-132-45-204.imcb.nus.edu.sg) |
01:32.23 | soulz- | hi all |
01:32.27 | ZX81 | high soulz- |
01:32.38 | soulz- | sup? |
01:32.41 | ManxPower | ~google site:lists.digium.com stuff thingy |
01:32.41 | ZX81 | nm |
01:32.42 | ZX81 | you |
01:32.45 | ZX81 | lol |
01:33.16 | ZX81 | no kaboodle? |
01:33.39 | ManxPower | I guess I need to post to the -users mailing list a message with "stuff" and "thingy" in it. |
01:33.44 | ZX81 | ~google manxpower |
01:33.45 | ZX81 | lol |
01:33.53 | ZX81 | ta rue |
01:33.58 | ZX81 | :) |
01:34.13 | ZX81 | ~google site:lists.digium.com stuff |
01:34.17 | ManxPower | The geocities link has nothing to do with me. |
01:34.18 | ZX81 | heh |
01:34.43 | soulz- | if i have a TDM04B and TDM40B, what will my /etc/zaptel.conf be? fxsks=1-4? |
01:34.55 | ManxPower | I should sue them for trademanrk infringemnet. |
01:35.14 | ZX81 | lol |
01:35.16 | ZX81 | :) |
01:35.22 | ManxPower | soulz-, it depends on the order in which you load the drivers I think |
01:35.27 | ZX81 | yup |
01:35.45 | soulz- | i did a modprobe wctdm, and it says http://pastebin.ca/5990 |
01:36.12 | soulz- | manxpower: it sees all if i do fxsks=1-4 |
01:36.18 | ManxPower | soulz-, Ah! Yes. Then it would (I THINK) depend on the order the cards are recognized, usually the closest to the power supply will be the first card. |
01:36.19 | ZX81 | soulz-: so what is the problem? |
01:36.36 | ZX81 | he has success |
01:36.38 | ZX81 | :) |
01:36.54 | tzafrir_home | when in doubt, consult /proc/zaptel |
01:36.56 | soulz- | zx81: how do i reference it say whether its a fxs or fxo? |
01:37.06 | shmaltz | has anybody used Verizon Wireless BroadBand Access with an IAXy or any other VOIP? |
01:37.11 | ZX81 | soulz-: groups |
01:37.13 | ZX81 | maybe |
01:37.17 | ZX81 | ~voip-info |
01:37.18 | jbot | somebody said voip-info was the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
01:37.19 | ManxPower | soulz-, FXO ports use FXS signalling |
01:37.35 | soulz- | http://pastebin.ca/5991 |
01:37.37 | ZX81 | shmaltz: not i |
01:38.08 | ZX81 | first four connect to telephone lines |
01:38.13 | soulz- | ok as per digium website the zaptel conf says fxsks=1-4 |
01:38.15 | ZX81 | second four connect to extensions |
01:38.29 | ZX81 | the (FCC mode) gives it away |
01:38.36 | ManxPower | soulz-, then your FXO ports are the first ones. |
01:38.37 | riksta | so in the sip.conf under context, how do i refer to a queue |
01:39.31 | soulz- | manxpower: sorry for sounding lame, but what about fxs? |
01:39.40 | ManxPower | riksta, You don't. That's in extensions.conf and queues.con |
01:39.51 | ManxPower | soulz-, FXS ports use FXO signalling. Yes, it's confuzzleing |
01:40.10 | riksta | so an incoming in extensions.conf is redirected to a queue how? it's a bit unclear on the wiki |
01:40.26 | soulz- | manxpower: so what i did was correct by using fxsks=1-4 |
01:40.29 | soulz- | ? |
01:41.31 | ZX81 | yep |
01:42.33 | ManxPower | fxsks=1-4 and fxoks=5-8 |
01:43.05 | ZX81 | and fkicantbefkd=1 |
01:43.07 | ZX81 | :) |
01:43.16 | soulz- | manxpower: thanks dude |
01:44.47 | riksta | can someone show me an example of putting someone in the queue from extensions.conf please |
01:45.01 | shmaltz | anybody used the local channesl for dialing multiple mutiline sip phones? |
01:45.15 | neuro_[rus] | please advice some linux software phone for use with asterisk? |
01:45.37 | soulz- | gnuphone works |
01:45.42 | Qwell | I like iaxcomm |
01:45.56 | Qwell | never used gnuphone...any good? |
01:46.30 | soulz- | used for testing |
01:46.35 | soulz- | use a xten for myself |
01:48.32 | neuro_[rus] | thanks... |
01:48.33 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
01:50.43 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
01:52.12 | riksta | man i can't get this queue to work, the person calls in, and the agent hears the beep, the music on hols is stopped, then the caller gets cut off immediately, and the agent hears on hold music again |
01:52.58 | ManxPower | riksta, Queues are hard to set up, complicated to use, and have several issues which made them unsuitable for me to use. |
01:53.06 | dsmouse | riksta: can the extention that answers recieve normal calls? is it AgentLogin or agentlogincallback? |
01:53.20 | riksta | AgentLogin |
01:53.28 | riksta | it shows the agent is logged in successfully on the cli |
01:53.43 | dsmouse | but if you call the agent's extention directly does it work? |
01:54.07 | riksta | yeah |
01:54.29 | dsmouse | do you have ackcall defined in agents.conf? |
01:54.36 | riksta | no |
01:54.40 | riksta | my agent hears the beep |
01:54.43 | riksta | but then it cuts off the call |
01:54.53 | dsmouse | if you don't set that to no, the agent has to press # to ack the call |
01:55.04 | *** join/#asterisk kks (~kks@203.115.210.253) |
01:55.13 | riksta | i think that you mean the other way round |
01:55.18 | riksta | if you set it to yea, he has to press # |
01:55.33 | dsmouse | er, that says for agentcallbacklogin... but it says default to yes... |
01:55.44 | riksta | i have it undefined, which means the call should connect immediately, which is what it tries to connect the call |
01:55.46 | riksta | then it loses it |
01:56.07 | riksta | i'm using AgentLogin |
01:56.25 | dsmouse | That was my best guess :/ sorry |
01:56.39 | riksta | ok thanks |
01:57.51 | riksta | oh i tihnk i see whats happening |
01:57.57 | riksta | it goes -- agent_call, call to agent '1000' call on 'SIP/1000-93db' |
01:58.04 | riksta | but im already on a call, with the music on hold? |
01:59.39 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) |
02:00.02 | algorithmn | silence suppression with rtptimeout can cut a (for example) sip call |
02:00.44 | algorithmn | happends when music on hold is on also |
02:01.15 | algorithmn | riksta: is the call cut off after a similar duration? |
02:01.42 | riksta | the caller is cut off directly when the call is tried to be transferred from the Queue to agent 1000 |
02:04.30 | algorithmn | riksta: sip? |
02:04.34 | riksta | yes |
02:05.03 | algorithmn | in sip.conf do you have rtptimeout set (other then that off the top of my head i wouldn't know) |
02:05.23 | riksta | its undefined |
02:05.39 | algorithmn | as in not mentioned? |
02:05.46 | riksta | its commented |
02:05.48 | algorithmn | ok |
02:06.04 | algorithmn | hmm... let me sit here and grab a beer to help the thinking process |
02:06.10 | riksta | thanks dude :) |
02:06.34 | *** join/#asterisk PyroSteve (~steve@wsip-70-183-114-254.no.no.cox.net) |
02:06.38 | PyroSteve | yO !! |
02:06.49 | PyroSteve | <PROTECTED> |
02:06.51 | PyroSteve | <PROTECTED> |
02:06.54 | PyroSteve | it works great |
02:06.59 | PyroSteve | but for the dtmf |
02:07.39 | PyroSteve | when I press button on my analog phone, the other end just hears a very very short blurp of the tone |
02:07.41 | hardwire | my telco provider just told me they have a patented conferencing called MeetMe |
02:07.42 | hardwire | hah |
02:08.01 | PyroSteve | and for the rest of duration of the tone is something like static |
02:08.21 | PyroSteve | any ideas |
02:08.30 | riksta | algorithmn: any chance i can PM you ? |
02:08.36 | algorithmn | sure |
02:08.41 | algorithmn | do it |
02:09.10 | algorithmn | PyroSteve: gsm? |
02:09.32 | PyroSteve | im using .... i think ulaw |
02:09.52 | PyroSteve | yeah.... g711u |
02:10.13 | PyroSteve | does spa-2000 support gsm ? |
02:10.55 | PyroSteve | what codec am i supposed to using |
02:12.01 | algorithmn | gsm has limited dtmf support |
02:12.12 | algorithmn | ulaw should be straight |
02:12.37 | PyroSteve | i noticed the problem when I tried navigating a pbx |
02:12.41 | *** join/#asterisk syslod (~sysglod@65.114.0.198) |
02:12.52 | algorithmn | are you AGI'in anything? |
02:12.52 | PyroSteve | then I called my cellphone and pressed buttons on the analog phone |
02:12.57 | syslod | sup |
02:13.01 | PyroSteve | no |
02:13.12 | hardwire | any providers that can give me a 1-800 incoming w/o a sales rep :) |
02:13.21 | algorithmn | ok.. i know during beta agi, things tend to hang and zap tends to crash |
02:13.28 | kks | Calling party name: [Tenor Call Relay SP Gateway], h323 incoming call failed with callerid with space. According to another side that they can't change the callerid. Is anything i can do on my asterisk ? |
02:14.10 | algorithmn | kks: try a softphone to test caller id... some older hardware boxes don't understand some characters |
02:14.30 | algorithmn | quite intermittent between brands, but not by how old the box is |
02:15.35 | syslod | Anyone have any channel banks with *? |
02:16.20 | kks | i have tried with SJphone, with callerid [Tenor Call Relay SP Gateway], but fail. Callerid [Tenor_Call_Relay_SP_Gateway] will work |
02:16.22 | *** join/#asterisk JamesDotCom (~james@sweep.bur.st) |
02:16.50 | algorithmn | i've gotten spaces in before... |
02:17.33 | *** join/#asterisk tzafrir (foobar@85-65-203-192.barak.net.il) |
02:20.03 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) |
02:20.43 | PyroSteve | can someone call me 662-796-1413 to hear my dtmf tone problem |
02:20.52 | PyroSteve | dial ext 103 |
02:22.07 | algorithmn | where is 662? |
02:22.29 | algorithmn | US48? |
02:22.36 | kks | thanks algorithmn. |
02:22.42 | syslod | PyroSteve. No ans. |
02:23.40 | algorithmn | kks: umm... with i new exactly what the CID prob is... |
02:23.45 | algorithmn | wish |
02:23.46 | algorithmn | knew |
02:25.24 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
02:25.24 | *** topic/#asterisk is Asterisk: The Open Source PBX || Dev Conf 2PM CST FEB 17th -> IAX2/guest@66.250.68.194/996 |
02:28.21 | kks | i have tried many callerid with softphone with and wihout space. and also tried to hardcode the callerid at dialplan, but channel h323 will take the original callerid for session initialization. |
02:29.34 | algorithmn | you know... im not sure.. but h323, maybe, doesn't support spaces?? i really have to idea |
02:29.45 | algorithmn | no idea |
02:29.51 | algorithmn | its possable though |
02:31.52 | tzanger | hahahaha |
02:31.59 | tzanger | Blinken, I'd like to you meet Achoo. |
02:32.01 | tzanger | A Jew? Here? |
02:32.03 | tzanger | hahahaha |
02:32.12 | ZX81 | lol |
02:32.24 | tzanger | this movie is awesome |
02:32.40 | ZX81 | whatcha watching |
02:32.42 | algorithmn | yah |
02:32.47 | tzanger | Robin Hood -- Men In Tights |
02:32.51 | algorithmn | ahhhh |
02:32.59 | mikegrb | TIGHT TIGHTS! |
02:33.01 | tzanger | Mel Brooks |
02:33.10 | algorithmn | Dave Chappel? |
02:33.38 | ZX81 | lol |
02:33.44 | tzanger | yup |
02:33.59 | ZX81 | and henceforth the Asterisk channel dissolves :) |
02:34.03 | tzanger | hahaha |
02:34.13 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
02:34.13 | tzanger | .... I have a MOLE?? |
02:34.29 | algorithmn | mmmm... thanks |
02:34.34 | algorithmn | i needed that |
02:34.52 | algorithmn | i missed the l-train downtown to see jimmy on the corner of 89th and 12th |
02:35.04 | algorithmn | so im a little tweaked |
02:35.32 | JerJer | hop a cab |
02:36.02 | algorithmn | nah.. i got my alternate reserve under my bed... but i can't seem to remember the combo to the lock |
02:36.20 | algorithmn | should've not drank the handle of svedka |
02:36.32 | algorithmn | now its on... |
02:36.39 | algorithmn | be back in 2min fellas |
02:37.18 | tzanger | drank the handle of svedka? |
02:37.55 | algorithmn | mmmmmmmmmmmmm mmmmmm |
02:38.14 | algorithmn | after it was empty.. i did <|;-( |
02:38.45 | PyroSteve | ahhh |
02:38.46 | algorithmn | wasn't born husulus, i was birthin'em |
02:38.48 | PyroSteve | haaaa |
02:38.49 | *** join/#asterisk Chuji (Chuji@pcp09929633pcs.tulipgrove.tn.nash.comcast.net) |
02:38.57 | PyroSteve | whoever just called me from NC |
02:39.00 | PyroSteve | thanks !! |
02:39.19 | PyroSteve | I forced both my spa-2000 and my sip connection to broadvice |
02:39.22 | PyroSteve | to use ulaw |
02:39.27 | algorithmn | JerJer: your buddy missed out on 500 spa2k's? |
02:39.32 | sudhir492 | when I do ztcfg -v I get the error: line 8: Unable to open master device '/dev/zap/ctl' |
02:40.00 | sudhir492 | any suggestions? I did run make install in zaptel |
02:40.03 | Chuji | sudhir492 : what modules are you loading? |
02:40.05 | Nivex | wha, someone else in here from NC? |
02:40.14 | algorithmn | sudhir492: red hat? |
02:40.19 | sudhir492 | yes, rh9 |
02:40.48 | algorithmn | you must make config before asterisk install/reboot |
02:40.50 | sudhir492 | I intend to load wct1xxp, for E1 card |
02:41.00 | Chuji | sudhir492 : Do this.... |
02:41.09 | algorithmn | init.d red hat n * sometimes don't like eachothr |
02:41.10 | sudhir492 | I did run make config in asterisk |
02:41.18 | algorithmn | darn |
02:41.50 | algorithmn | ?? gave it a shot, i'll keep a mid priority process running in the background for you |
02:41.52 | Chuji | sudhir492 : modprobe wct1xxp ; rmmod wct1xxp ; modprobe wct1xxp ; modprobe zaptel ; ztcfg -vv |
02:42.09 | Chuji | see if it still does it |
02:45.17 | sudhir492 | Chuji: Still the same error on ztcfg -vv, line 10: Unable to open master device '/dev/zap/ctl' |
02:46.24 | Chuji | does asterisk start? |
02:46.54 | sudhir492 | since modprobe wct1xxp fails, I did not even try to start asterisk |
02:47.00 | dsmouse | PyroSteve: did that fix it? |
02:48.07 | Chuji | try starting it. It might actually be loaded |
02:48.16 | dsmouse | Nivex: you're in NC? |
02:48.16 | vaewyn | Anyone that missed it earlier: http://www.wwwrogue.com/voip/WIP5000.html my review of the Hitachi Cable WIP-5000... Cool phone! |
02:48.17 | Chuji | also, check dmesg |
02:48.25 | sudhir492 | Chuji, lsmod doesnot show that :-( |
02:48.30 | Weezey | can I dial an extension to connect to the moh? |
02:48.41 | algorithmn | yes |
02:48.51 | Weezey | what's the command? |
02:48.52 | Nivex | dsmouse: I am. |
02:48.59 | *** join/#asterisk WizardWlf (~shawn@wrt54g.djernes.org) |
02:49.07 | dsmouse | Nivex: where about? |
02:49.08 | algorithmn | exten => _X.,1,MusicOnHold(random) |
02:49.09 | algorithmn | ex |
02:49.12 | Nivex | dsmouse: Raleigh |
02:49.19 | dsmouse | ?! |
02:49.20 | Weezey | thanks |
02:49.24 | dsmouse | As am I! |
02:49.46 | algorithmn | no worries |
02:49.52 | Chuji | sudhir492 : I get the same error on one of my boxes with a TE410 in it |
02:50.00 | Nivex | dsmouse: whoa... freaky |
02:50.02 | Chuji | sudhir492 : but I can pass right through it |
02:50.04 | sudhir492 | Yes, mine is TE110 too |
02:50.10 | dsmouse | Nivex: yea, really |
02:50.17 | WizardWlf | anyone know of a script to autogen .call files from a database |
02:50.40 | Chuji | WizardWlf : Wouldn't be hard to right one in perl |
02:50.50 | Chuji | WizardWlf : Should only take a few minutes |
02:51.03 | WizardWlf | but don't know enought perl to do the db stuff |
02:51.16 | Chuji | WizardWlf : What db you storing it in? |
02:51.40 | WizardWlf | prob be mysql but astdb would be ok also |
02:51.53 | sudhir492 | Chuji: I get the errors when I try modprobe wct1xxp |
02:51.54 | Nivex | dsmouse: maybe we should start a VUG :-P |
02:51.55 | sudhir492 | /lib/modules/2.4.20-8smp/misc/wct1xxp.o: init_module: No such device |
02:51.55 | sudhir492 | Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. |
02:51.55 | sudhir492 | <PROTECTED> |
02:51.55 | sudhir492 | /lib/modules/2.4.20-8smp/misc/wct1xxp.o: insmod /lib/modules/2.4.20-8smp/misc/wct1xxp.o failed |
02:51.55 | sudhir492 | /lib/modules/2.4.20-8smp/misc/wct1xxp.o: insmod wct1xxp failed |
02:52.10 | dsmouse | Nivex: *UG? |
02:52.17 | Nivex | http://www.voip-info.org/wiki-VoIP+User+Groups+USA |
02:52.23 | Chuji | sudhir492 : cat /proc/interupts |
02:52.42 | Chuji | sudhir492 : You got any conficts |
02:53.28 | Chuji | WizardWlf : I could probably piece you something together if you don't find anything |
02:53.35 | dsmouse | <PROTECTED> |
02:53.38 | dsmouse | bah |
02:53.43 | dsmouse | wow that's sparce |
02:53.50 | sudhir492 | Chuji, how do I look for conflicts in cat /prco/interrupts? |
02:54.09 | WizardWlf | Chuji: ok will keep looking |
02:54.11 | Chuji | sudhir492 : is your wcxx sharing anything? |
02:54.14 | Nivex | dsmouse: do you make it to any of the LUG meetings around here? |
02:54.37 | sudhir492 | I do not see wcxx in the list at all |
02:54.46 | dsmouse | I made a few... 1/2 the people there know me. |
02:55.02 | Chuji | sudhir492 : Well hell, bios isn't even seeing it then |
02:55.02 | sudhir492 | This is what I see after cat /proc/interrupts |
02:55.13 | sudhir492 | <PROTECTED> |
02:55.13 | sudhir492 | <PROTECTED> |
02:55.13 | sudhir492 | <PROTECTED> |
02:55.13 | sudhir492 | <PROTECTED> |
02:55.13 | sudhir492 | <PROTECTED> |
02:55.14 | sudhir492 | <PROTECTED> |
02:55.16 | sudhir492 | <PROTECTED> |
02:55.16 | Chuji | ~pastebin |
02:55.17 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
02:55.18 | sudhir492 | <PROTECTED> |
02:55.20 | sudhir492 | <PROTECTED> |
02:55.24 | sudhir492 | <PROTECTED> |
02:55.24 | Nivex | grak! |
02:55.26 | sudhir492 | <PROTECTED> |
02:55.28 | sudhir492 | <PROTECTED> |
02:55.30 | sudhir492 | <PROTECTED> |
02:55.30 | dsmouse | bad sudhir492 !!!!!!!!!!!!!!!! |
02:55.32 | sudhir492 | NMI: 0 0 0 0 |
02:55.34 | sudhir492 | locksy: 18137484 18137483 18137483 18137483 |
02:55.36 | sudhir492 | ERR: 0 |
02:55.38 | sudhir492 | mishehu: 0 |
02:55.41 | Chuji | ~pastebin |
02:55.42 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
02:55.42 | Nivex | /ignore sudhir492 |
02:55.48 | sudhir492 | sorry about that. I did not about ~pastebin |
02:55.54 | sudhir492 | Thanks for educating |
02:56.06 | sudhir492 | ~pastebin |
02:56.07 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
02:56.11 | Nivex | dsmouse: #trilug ? |
02:56.21 | Sedorox | Question for those with Ops.. why don't you put pastebin in the topic? |
02:56.27 | dsmouse | only a few times |
02:56.39 | dsmouse | I was more NCSU lug awhile back |
02:56.47 | JerJer | because people should be smarter than what they are working on |
02:56.51 | Chuji | sudhir492 : You sure your mobo is a compatible voltage with the T110? |
02:56.55 | dsmouse | you've met Peter then |
02:57.03 | Qwell | Sedorox: If they put everything in the topic, there wouldn't be room. ;] |
02:57.18 | Chuji | ~rtfw |
02:57.19 | jbot | i guess rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
02:57.25 | Chuji | That should be in the topic |
02:57.27 | Chuji | :) |
02:58.09 | Chuji | Or my personal fav for JerJer! |
02:58.18 | Sedorox | lol |
02:58.21 | Chuji | ~h323 |
02:58.22 | jbot | i heard h323 is evil! Don't ask about h323 here. Ask JerJer if you need to bug someone. He says it works, but others don't. |
02:58.29 | Sedorox | well I was just thinking the pastebin link |
02:59.18 | *** join/#asterisk didz_ (~omg@200.218.193.30) |
02:59.52 | didz_ | Ouch ... error while writing audio data: : Broken pipe |
02:59.52 | didz_ | Warning, flexibel rate not heavily tested! |
02:59.52 | didz_ | Segmentation fault (core dumped) |
02:59.55 | sudhir492 | Chuji: hmm, On the same mobo another box, I have TE410, no problem. Hence I put TE110 on this one even without checking |
03:00.03 | sudhir492 | 5v PCI |
03:00.06 | didz_ | anybody with a good heart could help me ? :) |
03:00.19 | Chuji | sudhir492 : is 5v what it takes? I can't remember |
03:00.47 | didz_ | twisted? |
03:00.52 | Chuji | didz_ : what version of mpg123 you running? |
03:00.54 | Weezey | anyone ever hooked a Norstar ATA into a FXO? |
03:01.18 | didz_ | i've been using the 0.59r, but now i'm using the format_mp3.so from asterisk-addons |
03:01.26 | didz_ | the same thing happens... |
03:01.44 | didz_ | already converted the mp3 to 8000 hz mono with lame |
03:02.04 | Chuji | when you lose the mp3's does that go away? |
03:03.20 | didz_ | haven't tried... because this is happening in a production environment... if i lose de mp3's the "clients" will hangup, since they will not hear anything =[ |
03:03.36 | didz_ | using with queues etc... |
03:03.55 | bjohnson | Beirdo: signed up and it works |
03:03.56 | Chuji | well, get the ones out of the asterisk source |
03:04.03 | Chuji | and see if it goes away |
03:04.05 | bjohnson | greg_work: you there |
03:04.16 | Chuji | They are nice and calming |
03:04.21 | didz_ | i'm using the ones from asterisk source |
03:04.35 | didz_ | fpm-calmriver, fpm-worldmix, fpm-sunshine |
03:04.36 | greg_work | bjohnson: i am |
03:04.42 | didz_ | i'm totally lost |
03:04.42 | bjohnson | pm |
03:04.52 | greg_work | sure |
03:04.57 | tzanger | hehehe |
03:04.58 | sudhir492 | Chuji: I am just checking digium website. |
03:05.00 | tzanger | strikey has loxed again! |
03:05.10 | Beirdo | bjohnson: cool |
03:05.13 | Chuji | didz_ : Yeah, I use those too |
03:05.27 | *** join/#asterisk ayano (~erik_leee@adsl-66-51-208-150.dslextreme.com) |
03:06.09 | *** join/#asterisk lilneon (~tj_r3@cuscon10992.tstt.net.tt) |
03:06.16 | lilneon | hi everyone.. and good night |
03:06.20 | didz_ | could it be a problem of lots of agents logged in ? |
03:06.21 | Chuji | didz_ : Does it crash you? |
03:06.26 | didz_ | yes |
03:06.34 | didz_ | segfault with coredump |
03:06.56 | Chuji | didz_ : I wouldn't think so, but it's possible. |
03:07.01 | syslod | Is there something you have to do different with FXO/FXS vs PRI??? |
03:07.06 | Chuji | How many mpg processes get spawned? |
03:07.15 | Chuji | syslod : Huh? |
03:07.21 | lilneon | hey guys, recently downloaded FC3. would you recommend moving my asterisk box from RH9 to FC3? any one had problems? |
03:07.26 | Chuji | PRI is fxo |
03:07.42 | Chuji | lilneon : Y? |
03:07.51 | didz_ | no one... i'm not using mpg123 anymore, i'm using now native mp3 support from asterisk (format_mp3.so) |
03:07.53 | dsmouse | lilneon: I've heard there's a trick to getting udev to work if you're useing local hardware |
03:08.02 | lilneon | chuji: don't know.. someone recommend i do it cuz it was better ..??? |
03:08.22 | syslod | Chuji: I've got a bunch of PRI working but the other day I decided to get Channel banks working and can't. Now I bought a few FXO cards and they don't seem to wanna work either. |
03:08.58 | Chuji | syslod : The more digium cards you put in one box you are asking for trouble |
03:08.59 | lilneon | so you would recommend i stay with rh9? still haven't gotten festival to wrk on there yet :( |
03:09.03 | Chuji | they are interrupt hogs |
03:09.17 | syslod | I kicked the box across the room a min ago but that didn't help. made me feel a little better though. |
03:09.26 | neuro_[rus] | I have Celeron-900Mhz PC with 256Mb RAM... Would it work well with _only_ two concurrent calls? I'm using software phones. |
03:09.34 | Chuji | lilneon : Well, It's just preference actually |
03:09.42 | Qwell | neuro_[rus]: should |
03:09.50 | Chuji | lilneon: Maybe try capstral too |
03:09.53 | lilneon | Chuji: so there is really no performance gain? |
03:09.56 | syslod | Chuji: I have a box with a new T1/E1 card and another box with 2 digium FXO cards 4 FXO/2FXO |
03:09.56 | dsmouse | lilneon: I have no first hand knoledge about asterisk on either FC3 or RH9, so I'm just repeating rumors. |
03:10.13 | Chuji | neuro_[rus] : yes, it will work, but soft phones suck |
03:10.17 | lilneon | dsmouse: so what do u run it on? debian? |
03:10.20 | Chuji | ~softphone |
03:10.21 | jbot | something that should be drug out into the street and shot |
03:10.27 | dsmouse | FC2 :) |
03:10.36 | syslod | ~hardphone |
03:10.38 | neuro_[rus] | Chuji: why? |
03:10.42 | Nivex | I'm still waiting for a good inexpensive hard phone. |
03:10.54 | Nivex | One of the guys here pre-ordered the SPA-841. He still hasn't seen it |
03:10.55 | Chuji | neuro_[rus] : Because they are only as good as the OS they run on |
03:11.01 | Chuji | what OS are they on? |
03:11.08 | lilneon | dsmouse:ok ..and your experience with asterisk on there was??? A.painful? B.very painful c.all of the above? |
03:11.15 | neuro_[rus] | Chuji: Linux and Windows |
03:11.20 | dsmouse | I like to say I've had no problems getting asterisk to work on fc2 :) |
03:11.26 | Chuji | :) |
03:11.40 | dsmouse | well, a few problems, but they were pure asterisk |
03:12.27 | lilneon | dsmouse: ok but since u had no prior experience with rh9 and asterisk i guess i can't really ask which u would recommend.. :S sigh |
03:12.47 | dsmouse | yea, but I had no prior experience with asterisk at all. |
03:12.58 | dsmouse | in fact, till sunday, I've only read about it :) |
03:13.08 | dsmouse | now I *LOVE* it. |
03:13.30 | Chuji | dsmouse : Hope you aren't busy for the next couple of months |
03:13.31 | Chuji | heh |
03:13.40 | dsmouse | Chuji: why? |
03:13.53 | Chuji | ~aa |
03:13.54 | jbot | test |
03:14.01 | lilneon | dsmouse: dont worry.. they only playing |
03:14.02 | Chuji | Heh |
03:14.29 | lilneon | asterisk is pretty much a tamed beast since the stable 1.0 release |
03:14.30 | dsmouse | Eh, asterisk is just fun for me. if it didn't work or annoied me I could just go back to a landline |
03:14.34 | lilneon | well in my opinion |
03:14.56 | Chuji | jbot AA is Asterisk Anonymous. Something we all should join after a few months of Astriholism |
03:14.57 | jbot | ...but aa is already something else... |
03:15.00 | lilneon | dsmouse: well i don't even use my landline anymore. . |
03:15.16 | dsmouse | lilneon: I don't trust my ISP that much yet. |
03:15.25 | Chuji | jbot erase aa |
03:15.40 | Chuji | jbot remove aa |
03:15.46 | Chuji | blah, stupid bot |
03:15.48 | dsmouse | jbot eat aa |
03:15.50 | jbot | ACTION eats aa and falls over dead |
03:16.04 | dsmouse | Who knew? |
03:16.06 | lilneon | dsmouse: well.. my 'clients' use it to make calls here. yeah and my ISP is my telco so.. they pretty much suck right thru.. but i don't really make that many calls |
03:16.11 | *** join/#asterisk Carp1 (~chatzilla@ip-204-97-151-110.modem.logical.net) |
03:16.38 | dsmouse | lilneon: the diffrence between asterisk at home vs asterisk at work* |
03:16.50 | dsmouse | * no I didn't use "asterisk at home" |
03:17.21 | Chuji | jbot Astriholics are people that spend every waking hour working with Asterisk. They need a life! |
03:17.22 | jbot | Chuji: okay |
03:17.35 | tzanger | we're men... we're men in tights... we roam aroudn the forest looking for fights... |
03:18.14 | lilneon | dsmouse: well i am the opposite.. i use it @ home.. and use my wrk's telephone @ wrk.. pretty much underpaying us anywayz |
03:18.18 | Carp1 | Anyone in here use/have Cell Socket? |
03:18.34 | lilneon | so wee use the 'facilities' they give us ;) |
03:18.40 | *** join/#asterisk tangotool (~sysglod@65.114.0.198) |
03:19.11 | tessier_ | Feb 16 19:15:55 WARNING[14177]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end) |
03:19.16 | tessier_ | Anyone know what causes this? |
03:19.26 | lilneon | hey anyone from teliax in here? |
03:19.28 | tessier_ | I really wish asterisk had better error messages. I would at least like to know what client or ip is causing this. |
03:19.40 | tessier_ | It has been streaming across my console like mad for the last hour. |
03:20.30 | Chuji | tangotool : You work for qwest? |
03:21.01 | sudhir492 | Chuji: You may be right. TE110 may be 3.3v card :-( I do not see any mention of 3.3v for this card at digium site thoug |
03:21.15 | Qwell | lilneon: dca I believe |
03:21.23 | sudhir492 | TE410 is 3.3v, TE405 is 5v |
03:21.35 | tzanger | sudhir492: correct |
03:21.41 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-110.modem.logical.net) |
03:21.42 | dsmouse | lilneon: I do |
03:21.44 | Chuji | sudhir492 : I think the new cards are 3.3 |
03:21.59 | sudhir492 | Darn |
03:22.07 | Carp1 | AstWind work good? |
03:22.20 | sudhir492 | for Quad T1/E1 they have both 5v and 3.3v versions |
03:22.20 | dsmouse | lilneon: right now I'm lukewarm about them |
03:23.03 | Qwell | dsmouse: lukewarm? |
03:23.32 | tangotool | Chuji: Used to years ago. |
03:23.34 | dsmouse | Qwell: well, they're pruduct seems ok, but they screwed up my order... |
03:23.41 | lilneon | Qwell: u know me? |
03:24.45 | dsmouse | I love that I can set my own callerid and they honor it... I can have multiple outgoing streems w/IAX2 support, etc |
03:25.17 | dsmouse | but I didn't get a incoming line like I expected, and he/she/it//they haven't returned my email |
03:25.24 | Chuji | tangotool : I just noticed your ip address when you /joined. I thought that /16 was their reserved backbone addressing |
03:25.39 | dsmouse | If I were a buisness, that would be really annoying |
03:26.10 | tangotool | Yea. I'm at work with a friend. Sitting in a NAP. |
03:26.40 | Chuji | Cool, throw up some warez! |
03:26.42 | Chuji | :) |
03:26.43 | lilneon | dsmouse:well i am a bit worried bout their new rates.. seems the routes i use most are those that got raised |
03:27.28 | lilneon | dsmouse: but i email them and get back a response.. most fo the time.. the following day .. two at most |
03:27.43 | dsmouse | well, it's only been 24 hours now |
03:27.54 | Qwell | lilneon: Do I? |
03:27.56 | lilneon | dsmouse: but they have been down quite a bit recently.. so i started looking for a secondary itsp |
03:28.13 | Qwell | lilneon: Ask dca, he should be able to help you. |
03:28.20 | Qwell | He was trying to pimp teliax to me a few weeks ago, heh |
03:28.22 | dsmouse | so, it's not like he's utterly pissed me off. also, I'm using the pay-as-you-go for domestic only... |
03:29.12 | lilneon | Qwell: oh cool.. when u said Dca.. i thought that was short for Dominica... the country i am originally from :D |
03:29.20 | dsmouse | I only chose telaix because they report to have 919 area codes... I'd prolly have gone with broadvoice otherwise... |
03:29.29 | dsmouse | but, hey, this is only a hobby for me :) |
03:29.30 | *** join/#asterisk puzzled_ (~patrick@puzzled.xs4all.nl) |
03:29.52 | Nivex | dsmouse: you try voicepulse? |
03:30.03 | lilneon | dsmouse: well this is business for me.. i lvoe asterisk.. and i am trying to set up a lil voip biz |
03:30.10 | dsmouse | Nivex: no 919 :( |
03:30.21 | lilneon | Nivex: yeah so far frm wat i have read voicepulse seems pretty ok |
03:30.26 | dsmouse | lilneon: doing what exactly? |
03:30.49 | *** join/#asterisk Nukemizer (Nuke@66.237.85.58.ptr.us.xo.net) |
03:30.52 | lilneon | dsmouse:reselling minutes.. here still under a monopoly by one telco |
03:31.02 | Nivex | dsmouse: say what?! |
03:31.02 | Nivex | Your official rate center name: RALEIGH |
03:31.03 | Nivex | (Might not match your city name) |
03:31.03 | Nivex | VoicePulse is available in your rate center! |
03:31.04 | sudhir492 | Chuji: Thanks a lot for your help. I will confirm tomorrow with Digium |
03:31.09 | lilneon | so.. ppl welcome paying lower rates to places like US and UK.. |
03:31.24 | dsmouse | ?! |
03:31.50 | Chuji | sudhir492 : No problem, hopefully you get it resolved |
03:32.56 | dsmouse | Nivex: hrms, I must have gotten confused with another carrier... do they do iax? |
03:33.11 | *** join/#asterisk iMediax (lklk@00045a809589.click-network.com) |
03:33.13 | dsmouse | oh |
03:33.14 | Nivex | dsmouse: connect.voicepulse.com does iax, but it's a different rate structure |
03:33.24 | dsmouse | 919 RALEIGH *ON ORDER* |
03:33.42 | Nukemizer | I have turned off my PRI while Digium sends me a new card, my "meetme" rom is now borken. DO I need to have Digium gear in the box in order to do conferencing ? |
03:33.57 | Sedorox | no |
03:34.03 | Chuji | ~ztdummy |
03:34.04 | jbot | ztdummy is probably zaptel timing source which uses a usb-ohci compatible usb controller as source. (part of zaptel cvs) |
03:34.27 | Chuji | ~google ztdummy |
03:34.40 | Nivex | which kinda sucks cuz every controller I've ever had has been uhci |
03:34.52 | dsmouse | beets ntegratedconsulting.com. They wanted a ear to avoid a setup charge. |
03:35.13 | dsmouse | ~rtfw |
03:35.14 | jbot | well, rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
03:40.01 | *** join/#asterisk roamer323 (~sing@Toronto-HSE-ppp4172487.sympatico.ca) |
03:42.28 | greg_work | is it bad to have exten => _123,1,Macro(something..) (pattern matching with no variables) ? |
03:44.28 | *** join/#asterisk doughecka (~Doug@doughecka.user) |
03:45.05 | doughecka | I have 2 2600 routers... |
03:45.10 | doughecka | what can I do with them |
03:45.35 | doughecka | I have 2 T1 DSU/CSU, 1 56/64 DSU/CSU and 2 ISDN BRI cards |
03:45.52 | doughecka | can I get asterisk to talk to them? |
03:46.24 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
03:48.32 | *** join/#asterisk PakiPenguin (~uppal@202.176.230.225) |
03:50.08 | lilneon | ok buh bye guys |
03:50.13 | *** part/#asterisk lilneon (~tj_r3@cuscon10992.tstt.net.tt) |
03:52.21 | Nukemizer | Chuji: jbot: thank you for the ztdummy tip ! |
04:01.14 | dsmouse | jbot: eath Nukemizer |
04:01.19 | dsmouse | jbot: eat Nukemizer |
04:01.21 | jbot | ACTION eats Nukemizer and falls over dead |
04:01.31 | dsmouse | that th is a hard habbith to break |
04:03.06 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
04:03.39 | Legend | does anyone else have greatly varying latency to nufone? |
04:03.58 | tzanger | Legend: you mean jitter? |
04:03.58 | tzanger | no |
04:04.20 | tzanger | mind you I'm 9 hops from them |
04:06.21 | roamer323 | when I call from * SIP extension to SIP extension, the callerid assigned in sip.conf shows up for xlite, but on an extension originating from the ATA - the ATA's callerid shows up... anyone know why? |
04:07.37 | tzanger | http://www.savetoby.com/ |
04:09.33 | Legend | wow, a blank black page |
04:09.34 | Legend | how moving |
04:09.41 | tzanger | it's flash |
04:09.42 | tzanger | wait for it |
04:10.03 | Legend | tzanger: its slow jitter |
04:10.12 | tzanger | slow jitter |
04:10.14 | tzanger | interesting |
04:10.40 | Legend | shit, i know i would be heckled for not singing nufone's praises |
04:11.01 | tzanger | who's heckling? |
04:11.06 | Legend | the latency fluctuates during the day, sometimes above 200ms, but voicepulse usually stays between 50ms and 100ms |
04:11.22 | tzanger | Legend: sounds like you have a better connection to vpc |
04:12.20 | tzanger | I recommend nufone when I feel it's right, as of all the provider's I've tried they Just Work. They've also gone well, well out of their way to help me out with testing the new jitter buffer in -HEAD |
04:12.24 | *** part/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com) |
04:12.28 | tzanger | er not in -HEAD but rather in the bugtracker (2532) |
04:12.34 | tzanger | M2532 |
04:12.42 | tzanger | nope jbot odesn't do the bug tracker link thing |
04:16.20 | tzanger | wow |
04:16.30 | tzanger | 6 people got organs/tissues from one woman |
04:16.35 | tzanger | I think that's amazing |
04:16.46 | Damascene | that must have been some woman |
04:16.52 | tzanger | much better use of a person's body after they're dead than incinerating them or burying them |
04:17.04 | tzanger | http://www.cbc.ca/stories/2005/02/16/rabies-transplants050216 |
04:17.10 | tzanger | 3 of the 6 have rabies now though |
04:17.20 | ManxPower | Death Penalty + Transplants = .... |
04:17.28 | tzanger | ManxPower: hah |
04:17.44 | tzanger | you onbviously have not seen the simpsons episode where Homer gets the dude's scalp for a transplant |
04:18.17 | ManxPower | tzanger, no, but I have read too much Larry Niven |
04:18.30 | tzanger | heh |
04:19.46 | tzanger | Neither the Great Depression, nor two World Wars could prevent the NHL from awarding the Stanley Cup, but with the league and the NHLPA still divided over the issue of a salary cap, Lord Stanley's trophy will not be contested for the first time since 1919, when a Spanish flu epidemic wiped out the finals |
04:19.51 | tzanger | wow |
04:19.57 | tzanger | oh well... nobody there's starving, that's for damn sure |
04:21.32 | tzanger | maybe losing the entire season will wake some people up to reality |
04:24.02 | implicit | :) |
04:24.25 | tzanger | anyway |
04:24.26 | tzanger | 'night |
04:24.44 | tzanger | damn |
04:24.54 | tzanger | all the beer and sauerkraut and saussage has left me quite gassy |
04:24.59 | implicit | anyone know any good SIP termination providers for US only that do NOT have asterisk in the media stream |
04:25.15 | tzanger | that's a good question |
04:25.22 | implicit | i know of a couple |
04:25.24 | tzanger | I am guessing dslreports would have something |
04:25.27 | Moc | any channel driver code + RTP expert ? |
04:25.29 | implicit | but i dont want to use them |
04:25.55 | implicit | just need a provider that has SIP jitterbuffers |
04:25.57 | implicit | and |
04:26.07 | implicit | doesn't restream RTP (thus the need for no asterisk) |
04:26.22 | tzanger | zoa's working on getting 2532 into chan_sip |
04:26.28 | tzanger | not sure about your #2 though |
04:27.05 | tzanger | 'night |
04:27.10 | implicit | #2 will never happen in * afaict |
04:27.59 | *** join/#asterisk numBone (~numBone@c-24-129-204-233.se.client2.attbi.com) |
04:31.09 | *** part/#asterisk didz_ (~omg@200.218.193.30) |
04:38.09 | *** join/#asterisk PTG123 (~PTG123@ip68-106-19-249.ph.ph.cox.net) |
04:38.30 | PTG123 | hey anyone here know a good source for the polycomm phones that is cheap |
04:38.38 | chipig | hey PTG123 |
04:39.33 | PTG123 | hey chi :) |
04:40.40 | *** join/#asterisk andrew` (~andrew@adsl-67-119-26-96.dsl.snfc21.pacbell.net) |
04:41.04 | bjohnson | PTG123: hang on |
04:42.20 | bjohnson | http://www.tritechcoa.com/phone-systems/O1B2.html |
04:42.39 | PTG123 | bj that the phone you would commen? |
04:42.41 | PTG123 | or something else |
04:42.45 | *** join/#asterisk yurpls (~yurplsl@65.114.0.198) |
04:42.47 | PTG123 | need a good reliable cheap phone to demo to business clients |
04:43.00 | bjohnson | that's not what you asked for |
04:43.07 | PTG123 | http://www.tritechcoa.com/product/b-O1B2-80.html |
04:43.13 | PTG123 | is that one any good, anyone know? :) |
04:43.16 | bjohnson | I don't have any voip phones |
04:43.27 | bjohnson | the IP500 is very popular here |
04:43.48 | bjohnson | the Sipura SPA 841 is beginning to be the popular <$100 voip phone |
04:43.49 | PTG123 | hmm |
04:44.13 | bjohnson | a lot of excitment about iax phones becoming available but not many actually in use yet |
04:44.35 | yurpls | Is there a list of IAX phones that are avail? |
04:44.53 | PTG123 | any ide aon agood plae to buy the sipura? |
04:45.12 | bjohnson | yurpls: no .. not a list |
04:45.21 | bjohnson | PTG123: what country are you in? |
04:45.25 | yurpls | atacomm has them. |
04:45.26 | PTG123 | us |
04:45.40 | yurpls | There are a bunch of other places. Try google. |
04:45.45 | bjohnson | PTG123: voxilla (free shipping) .. give me the voip coupons :) |
04:46.11 | bjohnson | PTG123: also check voipsupply .. but shipping cost usually makes them more $$ |
04:46.25 | bjohnson | try atacomm .. but I think they're in about the same range |
04:46.57 | PTG123 | so many choices, so little time :) |
04:47.14 | bjohnson | PTG123: take 5 minutes and I think you'll order from voxilla |
04:47.35 | PTG123 | looking at them now |
04:47.37 | bjohnson | oh yeah !! last I checked, using *users at voxilla got you $10 off |
04:48.15 | PTG123 | voxilla only has one phone for sale :) |
04:50.01 | sudhir492 | Occasionally, I have seen mention of DS3 cards for Linux/Asterisk. Any news on that front? |
04:50.11 | yurpls | No DS3 from digium. |
04:50.33 | yurpls | At that level you'll need more of a card than a non-dsp version. |
04:50.34 | PTG123 | i think the polycomm 300 looks the best/cheapest good combo |
04:50.47 | sudhir492 | I know not from Digium, but is anyone working on other DS3 cards? |
04:51.04 | yurpls | PTG123: I'd go with the 500. We have a pile of 300's here that ppl won't use after using the 500 and 600s |
04:51.18 | PTG123 | why don't they use the 300s? |
04:51.39 | yurpls | Maybe the SS7 folks but there are just a few that are interested in * supporting that level of calls. Including myself. |
04:51.56 | sudhir492 | It may not worthwhile for Digium to develop a DS3 cards as demand for those beasts will be very limited, however it will be cool if someone wrote driver for some other vendor's DS3 card |
04:52.11 | |Vulture| | IP500s are very nice |
04:52.33 | yurpls | PTG123: The phone isn't backlit, none are, but the 500 is better to see and the 600 even sharper. The 300 has a calculator display where the others actually have pixels. |
04:53.01 | yurpls | I think the SBC or maybe saganoma |
04:53.09 | yurpls | SBC definately have it. |
04:53.28 | yurpls | sorry sbs |
04:53.40 | sudhir492 | I have Asterisk installed at multiple locations with T1 cards. Consolidating at 1 place (even for 5 T1) will save me money. |
04:54.22 | PTG123 | well since a dual xeon can only encode 30 g729 streams |
04:54.26 | PTG123 | why would you want more then one t1 per box? |
04:54.28 | yurpls | We using multi 4 port cards with no probs. |
04:54.32 | sudhir492 | The provider is willing to give me as many T1 on a Ds3 interface. |
04:54.41 | PTG123 | not to mention redudance, blah blah |
04:54.43 | yurpls | PTG123: Why encode. ULAW rules. |
04:54.49 | PTG123 | sudhir492: so use a mux |
04:54.57 | yurpls | sudhir492: just get a mux. |
04:55.02 | PTG123 | yurpls: because when you encode most problems with voip disappear :) |
04:55.04 | PTG123 | on a busy network |
04:55.05 | yurpls | $1200 in us + cable |
04:55.05 | sudhir492 | Yes, that is what I am thinking |
04:56.06 | yurpls | PTG123: We are using it for TDM mostly. IP is on a 10Gig backbone with QOS and VLAN for voice. |
04:56.23 | yurpls | Adtran MX unit or telco edgelink. |
04:56.44 | PTG123 | yurpls: well its different if you can control the network, but if you are doing pbxing for a business, or whatever |
04:56.48 | PTG123 | encoding works much better |
04:56.57 | PTG123 | which reminds me i got to buy some g729 licneses now that it works on freebsd :) |
04:57.08 | yurpls | If a DS3 card was avaiblle I'd get it in a heartbeat. I'd love to do * VM on class 5 switching. |
04:57.15 | Qwell | oh, g729 works on freebsd now? |
04:57.32 | PTG123 | yes :) |
04:57.40 | PTG123 | it took mark around a year to implement my code ;) |
04:57.41 | PTG123 | but it works |
04:57.42 | PTG123 | heh |
04:57.46 | Qwell | a year? heh |
04:57.52 | Qwell | You did the mac stuff, right? |
04:57.52 | PTG123 | well a little opver a month :) |
04:57.54 | PTG123 | yah |
04:57.56 | Qwell | a few weeks ago? |
04:58.01 | yurpls | PTG123: yea we do businesses but they pay for upgraded networks. Some are doing GSM and a few have teleworkers that do G729 but mostly ULAW without probs. |
04:58.13 | PTG123 | gsm i still have issues with |
04:58.22 | PTG123 | ulaw i have issues with, but don't seem to have any with g729 |
04:58.23 | PTG123 | for some reason |
04:58.40 | PTG123 | plus it uses 1/10th the bandwidth of ulaw |
04:58.59 | yurpls | Yea but we have like 100's of users on a box. |
04:59.10 | PTG123 | so you can put in a good 30 calls plus have plenty left over for bandwidth on a t1 |
04:59.28 | yurpls | The box has 2 4 port T1 cards with NFAS PRI. |
04:59.30 | PTG123 | someone needs to make a hardware based g729 encoding daughter board |
04:59.41 | yurpls | That would be cool. |
04:59.44 | Qwell | That would be nice |
04:59.53 | Qwell | Take alot of the load off the server |
04:59.59 | yurpls | I'd like a T1 that had DSP for fax, modem, etc. |
05:00.24 | yurpls | We make a killing on fax servers. Hylafax for now. |
05:01.00 | PTG123 | hear that redder :) hold out your hand to yurpls :) |
05:01.00 | PTG123 | heh |
05:01.16 | PTG123 | you know redder in here is the primary hylafax developer? :) |
05:01.33 | yurpls | Hylafax = gooooood |
05:01.49 | Qwell | Whats hylafax exactly? |
05:01.50 | yurpls | Been using it for years. Stupid rightfax servers sucked. |
05:01.55 | Qwell | One of those fax-email gateways? |
05:01.57 | PTG123 | are you terminating t1s directly into the hylfax servers, or handing off via ip? |
05:02.27 | yurpls | PTG123: I work for the LEC so its about 100 feet from a class 5 tandem with DS1 handoffs. |
05:02.51 | yurpls | Dialogic cards. |
05:03.12 | PTG123 | ah gotcha, my next project is to get hylafax or asterisk fax working well with sip handoffs |
05:03.21 | PTG123 | so i can allocate numbers to my clients anywhere in the country |
05:03.44 | yurpls | sip -> fax? |
05:04.20 | PTG123 | ya |
05:04.29 | PTG123 | if t.38 worked in asterisk it would be very simple |
05:04.30 | yurpls | with g729? |
05:04.32 | PTG123 | ulaw is a little iffy |
05:04.34 | PTG123 | nah :) |
05:04.39 | yurpls | oh I was confused. |
05:04.44 | yurpls | Yea t.38 would be nice. |
05:05.02 | PTG123 | if i could find me someone to set me up a test t.38 sip account |
05:05.06 | bjohnson | Qwell: hylafax is linux software to send faxes |
05:05.10 | PTG123 | i'd do the t.38 iface |
05:05.17 | yurpls | and receive faxes. |
05:05.19 | PTG123 | bjohnson: and receieve |
05:05.20 | bjohnson | I believe mgetty is still used to receive them |
05:05.38 | yurpls | Kinda like sendmail for faxing |
05:05.42 | PTG123 | bjohnson: hylafax does all the work |
05:05.55 | PTG123 | it "can" use mgetty to answer the phone |
05:06.17 | yurpls | * seems to work ok with spandsp on the receive. |
05:07.05 | bjohnson | mgetty also auto-senses fax vs data and can autostart ppp for modem calls |
05:07.52 | Moc | I would love a soft modem... that can connect to ppp or something |
05:08.09 | PTG123 | moc: their is such a thing |
05:08.12 | yurpls | Doesn't * do ras? |
05:08.16 | bjohnson | how about connecting to my boot? |
05:08.19 | PTG123 | spandsp guy just came out with it |
05:08.35 | bjohnson | oh you mean for * |
05:08.45 | PTG123 | yah well sort of |
05:08.52 | yurpls | Is that pre10? |
05:08.58 | PTG123 | basically now you can make asterisk emulate a modem to iface with asterisk |
05:09.03 | PTG123 | came out like a week ago |
05:09.08 | Qwell | bjohnson: thanks |
05:10.47 | *** join/#asterisk ScythelX (Fleb@pc-24-181-176-72.sbi.ct.charter.com) |
05:13.11 | *** join/#asterisk ranliv (~ranliv@210.5.98.224) |
05:14.04 | *** join/#asterisk B4 (~B4@202.69.48.245) |
05:14.24 | Moc | PTG123: he did ? |
05:14.24 | B4 | ~seen zx81 |
05:14.26 | jbot | zx81 <matt@222-153-114-115.jetstream.xtra.co.nz> was last seen on IRC in channel #asterisk, 2h 40m 27s ago, saying: 'and henceforth the Asterisk channel dissolves :)'. |
05:14.30 | PTG123 | moc: yah |
05:14.37 | Moc | he rock... |
05:14.59 | Moc | in pre10 ? |
05:15.14 | PTG123 | i don;'t know what pre10 means :) |
05:15.20 | PTG123 | ask redder86 about it he knows more |
05:15.35 | yurpls | pre10 is the current version buts theres no docs. |
05:15.37 | bjohnson | PTG123: Sedorox was looking for that I think |
05:16.07 | Moc | ok well I check it once I get this UNISTIM channel driver working like I want to.. |
05:16.34 | PTG123 | using asterisk i am very much use to the no docs part :) |
05:17.14 | PTG123 | anyone here know about clevo/sager notebooks |
05:18.45 | Sedorox | `whats that? |
05:19.06 | PTG123 | a brand of notebooks |
05:19.09 | Sedorox | PTG123: I have a Sager |
05:19.15 | PTG123 | http://pctorque.com/3790.php |
05:19.18 | PTG123 | thinking about buying that one |
05:19.21 | PTG123 | Sedorox: which one? |
05:19.45 | Sedorox | I have the 8790 (can't get it anymore.. the 9860 or whatever it is replaced it... ) |
05:19.50 | Sedorox | the very high end model... |
05:19.56 | Sedorox | my friend here got that model your looking at |
05:19.56 | PTG123 | where did you buy it from |
05:20.00 | Sedorox | he loves it |
05:20.02 | PTG123 | i am looking for very small but powerful |
05:20.09 | Sedorox | I got it from sagernotebooks.com, but they don't have it listed anymore... |
05:20.13 | Sedorox | yea.. thats why he got it |
05:20.15 | PTG123 | Sedorox: great thats what i want to hear :) any recommendations from where to buy? |
05:20.18 | Sedorox | mine doesn't have any battery life.. maybe a hr |
05:20.20 | PTG123 | i need 1920x1200 too |
05:20.24 | Sedorox | hehe |
05:20.25 | PTG123 | yah this one has 5 hours |
05:20.28 | Sedorox | mine's 1680x1050.. but eh |
05:20.30 | Sedorox | Yup.. centrino |
05:20.33 | Sedorox | he loves it.. |
05:20.35 | Sedorox | umm |
05:20.42 | PTG123 | he get it direct from sager too? |
05:20.50 | Sedorox | ... he used PCTorque first.. then ended up ordering from another site.. let me see if I can find it |
05:21.03 | PTG123 | k |
05:21.07 | PTG123 | why did he not use pctorque? |
05:21.32 | Sedorox | http://www.powernotebooks.com/product.php?itemId=414 |
05:21.44 | Sedorox | he had some ordering problems with his credit card or something I think |
05:21.54 | Moc | :q |
05:21.55 | Sedorox | I think thats where he got it from if I remember correctly |
05:22.25 | Sedorox | Everything works in Linux, 'cept the camera in the screen, and the TV Tuner that I have in mine |
05:22.32 | PTG123 | yah powernotebooks is um $50 more :) |
05:22.36 | Sedorox | but I think they are working on both them soon |
05:22.48 | PTG123 | yah i need to dual boot so that is good news |
05:22.54 | PTG123 | how long did it take you to receive them? |
05:22.57 | Sedorox | yea |
05:23.00 | PTG123 | and do either of you have dead pixels? |
05:23.24 | Sedorox | Mine I got within a week I think... I used standard shipping.. not sure with him, I think he got it over one of the breaks last semester |
05:23.38 | Sedorox | I have a few on mine.. but I don't notice them unless I realllyyyy look for it |
05:23.57 | Sedorox | only noticed it because it was rebooting.. so it was a black screen, and I happened to be close to the screen |
05:24.02 | PTG123 | i wish i could find a place with a dead pixel guarantee |
05:24.39 | Sedorox | Ummm I think Sager will replace the screens for dead pixels.. I think its covered in the warrenty.. not sure.. |
05:25.05 | PTG123 | no |
05:25.08 | PTG123 | only if in center |
05:25.27 | Sedorox | hmmm |
05:26.40 | Sedorox | yea.. overall.. bothof us have been extremely happy with it |
05:27.24 | Sedorox | btw... notebookforums.com is few good for sager stuff.. and also alienware.. dell's.. gateway.. hehe |
05:27.40 | PTG123 | yah |
05:27.44 | PTG123 | thats where i found out about th em |
05:27.57 | PTG123 | i don't like it b/c pctorque kills threads if you ask about places to buy other then them :) |
05:28.07 | PTG123 | i found a plae that had a dead pixel guarantee |
05:28.09 | PTG123 | but loist the url |
05:28.26 | Sedorox | interesting |
05:28.29 | Sedorox | lol |
05:29.36 | Sedorox | I found out about it from a friend of mine in HS.. he got it.. I loved it when I saw it.. and I actually managed to talk my parents into getting it when I went to college.. |
05:29.50 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
05:30.44 | PTG123 | heh |
05:30.49 | PTG123 | well mine notebook just got fried |
05:30.55 | PTG123 | ribbon cable broke so lcd is useless |
05:30.55 | bjohnson | sedorax: weren't you looking for a soft modem? |
05:30.58 | PTG123 | and its pretty beaten up |
05:31.01 | PTG123 | so time to buy a new one |
05:31.14 | bjohnson | PTG123: selling? |
05:31.50 | Qwell | I need to get me a good used laptop |
05:32.40 | PTG123 | bj: selling what? |
05:32.47 | PTG123 | a broken laptop? :) |
05:33.07 | bjohnson | yes |
05:33.16 | PTG123 | Qwell: i got a nice p3 1ghz i'll sell cheap :) |
05:33.17 | Sedorox | bjohnson: no.. I don't think so... |
05:33.23 | PTG123 | bjohnson: for what? |
05:33.24 | Qwell | PTG123: How cheap we talking? |
05:33.30 | Sedorox | lol |
05:33.31 | bjohnson | Sedorox: sorry .. must have been something else |
05:33.36 | Sedorox | and everyone jumps on a broken laptop |
05:33.41 | PTG123 | Qwell: not sure :) i never use it.. its pretty nice though |
05:33.41 | bjohnson | PTG123: you mean for what use? |
05:33.45 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
05:33.53 | Sedorox | bjohnson: I don't think I was asking about anything today.... |
05:33.56 | Sedorox | allwell :-p |
05:33.58 | bjohnson | PTG123: depends on specs .. maybe print server |
05:34.05 | bjohnson | PTG123: maybe router |
05:34.09 | Sedorox | hehe |
05:34.11 | Qwell | laptop router? heh |
05:34.18 | PTG123 | bjohnson: heh why not just use a pc? :) |
05:34.21 | Qwell | makes sense...low power |
05:34.21 | bjohnson | ipcop or monowall |
05:34.31 | Sedorox | hehe |
05:34.33 | bjohnson | PTG123: laptop is smaller and has built in UPS |
05:34.46 | Qwell | hmm, thats actually not a bad idea |
05:34.47 | PTG123 | bjohnson: well its in pieces now :) |
05:34.59 | PTG123 | bjohnson: thats basically what i did is made it into a little computer |
05:35.01 | Sedorox | my old IBM Thinkpad 600 just sits here on my desk out of the way acting as a tunnel endpoint from my house, to allow me access inside my home network, and to give me IPv6 access here at school |
05:35.08 | Qwell | PTG123: If you quote me a price, I'll discuss with the wife |
05:35.14 | Sedorox | lol |
05:35.23 | PTG123 | Qwell: make me an offer, feeling generous today :) |
05:35.29 | PTG123 | let me find the model one sec |
05:35.32 | Sedorox | $20! |
05:35.35 | Sedorox | :-p |
05:35.37 | bjohnson | PTG123: what the heck is it? |
05:35.39 | PTG123 | not that generous |
05:35.42 | Sedorox | lol |
05:35.47 | ScythelX | do I need to setup something so SER handles dtmf for asterisk - or configure asterisk in a certain way? |
05:35.50 | PTG123 | bjohnson: the broken one, p4 2.4ghz or something like that |
05:35.52 | bjohnson | PTG123: specs |
05:35.56 | Sedorox | wow... |
05:36.02 | roamer323 | on an incoming sip call, how does * find the [???] entity in sip.conf to use? does it seach by username, host? anyone? |
05:36.04 | bjohnson | just the lcd is broken right? |
05:36.07 | PTG123 | it was a 5k notebook 2 years ago, or so |
05:36.09 | bjohnson | ram and HD? |
05:36.11 | jbot | IRC Client versions for #debian-bots (4): other - 7 (46.7%) ;; unknown/cloak - 6 (40%) ;; eggdrop - 1 (6.7%) ;; irssi - 1 (6.7%). |
05:36.27 | PTG123 | bjohnson: like i said i gutted it, like 60gig drive or so, 512+128 ram |
05:36.49 | ScythelX | RFC2833? |
05:36.55 | bjohnson | PTG123: you can put it back together? |
05:37.38 | PTG123 | bjohnson: um probably not i kept the bottom intact, i cut out of amettle a custom cpu headsink, etc |
05:37.42 | PTG123 | and was gonna cut plexiglass for the top |
05:37.46 | bjohnson | well, whatever .. $100 in pieces plus shipping cost |
05:37.47 | PTG123 | so it was like a transparent notebook |
05:37.49 | Qwell | PTG123: yeah, if you could get the model number, would be cool to look at it |
05:38.08 | PTG123 | its kind of a project, so probably more fun to keep it |
05:38.28 | bjohnson | PTG123: sure .. now that I've given you ideas for what to use it for |
05:38.31 | PTG123 | besides the ram alone would be worth keeping to put in new notebook :) |
05:38.36 | bjohnson | how about a laptop * server |
05:38.55 | Sedorox | hehe |
05:38.56 | bjohnson | would be okay for homw or small business use |
05:38.59 | Sedorox | good if you travel |
05:39.00 | Sedorox | :-p |
05:39.16 | bjohnson | there's someone out there running * on a P100 with 16M RAM according to the wiki |
05:39.20 | Sedorox | PTG123: I doubt it would go in any new laptop... |
05:39.22 | Qwell | wpw |
05:39.31 | Qwell | wow, rather... |
05:39.38 | Sedorox | I was thinking about getting a Soekris and putting asterisk on it |
05:39.44 | Sedorox | its a 266 Via chip |
05:40.18 | Qwell | heh |
05:41.00 | Sedorox | lol |
05:42.15 | PTG123 | damn trying to figure out model # :) |
05:42.19 | PTG123 | notebook at office |
05:42.21 | Qwell | What brand? |
05:42.42 | bjohnson | Sedorox: read some info about soekris not being great for * |
05:42.48 | PTG123 | sedorox: its pc2100 or whatever ram for notebook so it will :) |
05:43.20 | PTG123 | Sedorox: what do you think my other notebook is worth? :) |
05:43.37 | bjohnson | PTG123: keep the 512 .. give me the rest with the 128M |
05:44.08 | PTG123 | bj: heh |
05:44.24 | PTG123 | then i got to package it up, etc :) |
05:44.37 | PTG123 | alright how about $150 for it i suppose |
05:44.40 | Sedorox | bjohnson: any perticular reason why? |
05:44.44 | Sedorox | PTG123: hehe, ok :-p |
05:44.45 | B4 | feeling generous :) donate the notebook to me |
05:44.51 | Sedorox | what other notebook.. the 1gig? |
05:45.02 | PTG123 | sederox: yah if you want a good cheap notebook |
05:45.06 | PTG123 | its titanium too, and small |
05:45.09 | PTG123 | which is why i liked it :) |
05:45.11 | PTG123 | durable |
05:45.14 | Qwell | PTG123: What brand, etc? |
05:45.16 | Sedorox | lol |
05:45.17 | PTG123 | got 2 batteries, etc |
05:45.20 | PTG123 | its an acer.. :) |
05:45.30 | Sedorox | I'm not really looking for anything.. I don't have any money right now :-/ |
05:45.33 | Qwell | how much/what type of ram? |
05:45.41 | Sedorox | well I little.. but I got bills and stuff |
05:46.13 | PTG123 | qwell: hah why you looking for one? :) |
05:46.16 | PTG123 | welcome to e#ebay :) |
05:46.22 | PTG123 | bjohnson you still alive? |
05:46.25 | bjohnson | PTG123: including shipping or not for $150 USD? |
05:46.30 | B4 | #asteriskbay |
05:46.35 | Qwell | PTG123: Never owned a laptop, but it was really nice when I was able to bring the one from work home. |
05:46.38 | Sedorox | lol |
05:46.40 | bjohnson | B4: abay |
05:46.45 | B4 | lol |
05:46.51 | PTG123 | bjohnson: do you care how its shipped? |
05:47.04 | Sedorox | this is my first _real_ laptop.. the TP600 could never put a battery in it... |
05:47.18 | B4 | me no ... I can even accept shipments via donkey cart |
05:47.20 | bjohnson | PTG123: as long as I can get it working once recieved |
05:47.22 | PTG123 | Qwell: well i never use it, i can't use any res less then 1600x1200 so if you are interested |
05:47.26 | PTG123 | thats my only complaint about it |
05:47.30 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-86-221.sentechsa.net) |
05:47.33 | bjohnson | PTG123: you can ship it camel back for all I care |
05:47.50 | Qwell | PTG123: 15"? |
05:47.55 | PTG123 | bjohnson: well its working for me here :) Well i am beginning to like usps alot for shipping |
05:47.59 | PTG123 | Qwell: yah |
05:48.04 | Qwell | how much/what type of ram? |
05:48.08 | B4 | BTW you can get reaaaaal chep lcds from china :) |
05:48.11 | bjohnson | PTG123: usps is good for me |
05:48.16 | PTG123 | Qwell: its a very nice notebook, um 512 |
05:48.20 | bjohnson | PTG123: shall we move to pm? |
05:48.27 | PTG123 | sure bj |
05:48.54 | Qwell | PTG123: Are you talking $150 for the 1ghz, or the b0rked one? |
05:49.06 | PTG123 | qwell: borked one, i got two of em :) |
05:49.17 | PTG123 | Qwell: i'll take any decient offer for other |
05:49.32 | PTG123 | i also got a g4 powerbook any takers on that :) the ti one |
05:49.38 | B4 | other one also broken lcd? |
05:49.47 | PTG123 | ok 3 books |
05:49.58 | Qwell | PTG123: Mind another PM? heh |
05:50.03 | PTG123 | 1. P4 2ghz-2.4ghz fucked lcd, and kind of in pieces |
05:50.08 | PTG123 | 2. p3 1ghz nice shape :) |
05:50.13 | PTG123 | 3. Powerbook TI g4 :) |
05:50.18 | PTG123 | and sure the more pm the better |
05:50.43 | wasim | PTG123: i'll trade you #2&3 for a farfon :) |
05:50.46 | Sedorox | lol |
05:50.52 | Sedorox | farfon? |
05:50.55 | B4 | lol wasim |
05:50.55 | PTG123 | what is a farfon? :) |
05:51.11 | wasim | ~farfon |
05:51.12 | ScythelX | anyone ever have dtmf problems with client phone - > ser - > asterisk |
05:51.13 | B4 | hey PTG thats a good offer |
05:51.25 | wasim | PTG123: there are only 40 of them in the outside world |
05:51.35 | PTG123 | ScythelX: think you are in the wrongc hannel for asterisk questions |
05:52.00 | B4 | this is #abay ScythelX |
05:52.01 | wasim | farfon.com ... an iax2 hardphone shipping prerelease |
05:52.13 | clive- | wasim, hey, hows the iax,ata comming along? |
05:52.16 | ScythelX | bo yeah |
05:52.22 | B4 | whats the beta price wasim? |
05:52.40 | wasim | B4: ranging between EU 350 and 1000 + vat! |
05:53.00 | B4 | different models? |
05:53.18 | wasim | clive-: its taking a little finangling, coz we're missing a couple of components |
05:53.34 | wasim | B4: nope, shortage of supply, and levels of support |
05:53.44 | *** join/#asterisk scratchrf (~dirk@65-102-181-251.tukw.qwest.net) |
05:53.47 | wasim | B4: you get a 650 EU rebate with the 1000 price tag |
05:54.02 | clive- | wasim, soon youull be selling a iax version on the pa168 ata |
05:54.25 | wasim | clive-: not us, but others will be for sure |
05:54.27 | B4 | hmm well shortage of supply should not raise the price at the manufacturer |
05:54.47 | wasim | B4: we're not the manufacturerr, we're just the r&d shoppe, these are all hand made prototypes |
05:55.00 | wasim | B4: actual vol retail price should be $75 |
05:55.12 | B4 | ah ok ... |
05:55.33 | wasim | B4: these units are for core itsps's to test features etc |
05:55.45 | B4 | outsourced work from some other company? |
05:55.54 | Sedorox | hmmm |
05:56.02 | wasim | B4: outsourced the pcb manufacturing and mounting |
05:56.07 | clive- | :) |
05:56.36 | B4 | no I mean you are r&d ing for for some other manufacturer? |
05:56.38 | wasim | clive-: ofcourse, both products are for different markets |
05:56.39 | *** join/#asterisk pygmy (~pygmy@141.110.15.200) |
05:56.55 | wasim | B4: no, we're licensing out the firmware for other manufacturers |
05:57.25 | wasim | clive-: we're aiming for high end, niche phones ... stuff with encryption and advanced features |
05:57.37 | wasim | and ofcourse, phones for techies :) |
05:57.51 | wasim | we should be releasing a lot of the firmware at lots.ch |
05:58.03 | *** join/#asterisk letherglov (~letherglo@8036aa5e.resnet.ucsd.edu) |
05:58.04 | wasim | as per our comittment to open source |
05:58.15 | B4 | whats the hardware based on? |
05:58.21 | wasim | ti c54xcst |
06:00.06 | scratchrf | has anyone used SixTel/iax.cc? |
06:00.29 | hmodes | has ti released an ata work-alike yet? |
06:00.30 | B4 | seems popular choice so you should have many firmware customers |
06:00.49 | hmodes | i thought someone would have been all over that by now |
06:00.54 | wasim | hmodes: ti? no, we use silvertels slics |
06:01.05 | letherglov | why not ti's slcs |
06:01.13 | letherglov | and are those the same slics digium uses? |
06:01.25 | hmodes | because ti is hopelessly behind the curve perhaps? :) |
06:02.11 | wasim | i really want to get the uberATA out and testing |
06:02.22 | wasim | but we need a little dough before we can do that |
06:02.28 | hmodes | uberata eh? |
06:02.36 | letherglov | it's a shame you're not in afghanistan |
06:02.41 | hmodes | can it school a sipura 3k? :) |
06:02.42 | letherglov | I hear the poppy crop is great this year |
06:02.51 | wasim | hmodes: 30 port modular fxs/fxo iax2 ethernet/dsl "channel bank" |
06:03.00 | wasim | letherglov: yeah, bumper crop |
06:03.01 | hmodes | sooo that would be a yes |
06:03.02 | hmodes | how much? |
06:03.12 | B4 | lol letherglov |
06:03.23 | wasim | hmodes: retail 1200 - 1500 us$ ? |
06:03.54 | hmodes | nice |
06:04.14 | wasim | hmodes: thats what we think to |
06:04.17 | hmodes | does it have its own dsps or does the iax host have to do the codec work? |
06:04.27 | wasim | hmodes: no, it has its own dsps |
06:04.33 | *** join/#asterisk Firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
06:04.34 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
06:04.50 | hmodes | yeah, i conceed that is hotness |
06:04.53 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
06:04.57 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
06:05.07 | wasim | especially if we can get it to retail around $1k, that would be a nice product indeed |
06:05.10 | clive- | wasim so when is the launch date |
06:05.11 | B4 | channel bank in the works wasim? |
06:05.28 | hmodes | hell, i'd buy one for home :) |
06:05.35 | wasim | clive-: we're open sourcing the firmware today i think |
06:05.41 | B4 | yeah book one for me too :) |
06:05.52 | Firestrm | anyone know if engenius is selling their wifi phone yet? |
06:06.35 | clive- | wasim does that mean the hardware is done? |
06:06.43 | wasim | clive-: the hardware has been done for ages now |
06:06.57 | B4 | want beta testers? heh |
06:07.01 | wasim | B4: sure |
06:07.08 | clive- | ah, cool,..send me 2 |
06:07.15 | clive- | does it work? |
06:07.16 | B4 | send me one :) |
06:07.16 | wasim | but we're really running short of units |
06:07.26 | wasim | clive-: ofcourse, its testing all over EU right now |
06:07.39 | wasim | clive-: .de, .ch, .pl. .no, .nl |
06:07.50 | B4 | impressive |
06:07.51 | Firestrm | wasim, i joined half way through conversation, what hardware are we talking about? |
06:08.12 | wasim | Firestrm: the elusive farfon |
06:08.49 | implicit | wasim whats the pricing? |
06:09.20 | wasim | most of the world seems to :) |
06:09.34 | *** join/#asterisk pranav (sameer@202.149.48.200) |
06:10.08 | implicit | Firestrm: you need everything it seems |
06:10.34 | Firestrm | implicit, first the earth.. then the universe.. |
06:10.51 | Mavvie | redder86: still awake? |
06:11.01 | implicit | Firestrm: other way around would save time |
06:11.08 | implicit | first the universe then no need for the earth |
06:11.17 | Firestrm | hmmm.... i like it.. |
06:11.46 | Firestrm | i can obtain the universe, but i cant get my PSTN line working properly.. |
06:11.48 | Firestrm | :( |
06:12.22 | implicit | you dont know if you can obtain the universe until you do it |
06:12.34 | implicit | you can't even really know the essence of what it is to obtain |
06:12.41 | implicit | until you obtain every obtainable thing |
06:12.52 | hmodes | pfft |
06:12.59 | hmodes | everything == nothing |
06:13.26 | implicit | hmodes: that is not a bad assumption |
06:13.28 | pranav | we are not able to connect to fwd |
06:13.32 | implicit | but you can't prove by negation |
06:13.40 | hmodes | damn! |
06:14.02 | hmodes | then the only option is to prove everything or nothing |
06:14.03 | hmodes | wtf?! |
06:14.59 | pranav | hello |
06:15.01 | pranav | we are not able to connect to fwd |
06:15.02 | implicit | hmodes: no |
06:15.39 | *** part/#asterisk B4 (~B4@202.69.48.245) |
06:15.44 | implicit | it depends, is empirical evidence enough for you or do you want to actually prove it? |
06:15.56 | Qwell | pranav: We saw that the first time. Does anything happen that might be remotely useful in debugging the problem? |
06:16.18 | hmodes | eff empirical evidence, what has it ever done for me? |
06:16.50 | implicit | hmodes: a lot? |
06:16.58 | implicit | lol |
06:17.05 | hmodes | damn! foiled again! |
06:17.51 | hmodes | so one should accept what they're told then? |
06:18.18 | *** join/#asterisk ayano (~erik_leee@adsl-66-51-208-150.dslextreme.com) |
06:18.40 | ayano | what are the disadvantages of asterisk@home? |
06:18.56 | Qwell | the extra 5 chars you have to type when discussing it |
06:19.05 | wasim | ayano: you sound like an alien if the connection is made |
06:19.31 | hmodes | haha |
06:19.36 | hmodes | such hate for the noobs |
06:19.44 | hmodes | that's both disturbing and wonderful |
06:20.17 | hmodes | although i suppose he was smart enough to ask |
06:20.37 | ayano | I just tried it, I have always just used source for installing * but that was easy, there has to be some kind of drawback. |
06:21.34 | hmodes | appliance installs are the devil and guis tend to prohibit touching the text configs in any sort of creative manner? |
06:21.49 | ayano | ahh, there we go, so it will only let you configure surtain things? |
06:21.56 | ayano | certain |
06:21.59 | ayano | oops |
06:22.07 | hmodes | via the gui as last i saw it, yeah |
06:22.22 | hmodes | much better to just hack it up yourself if you can manage |
06:22.58 | ayano | I see, but to accually do the install from it is it okay? It only took like 30 minutes on a laptop... |
06:24.17 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
06:25.37 | hmodes | ehhh, probably for a dedicated * box |
06:25.52 | hmodes | tho I'd still be vaguely paranoid about security after the install |
06:26.02 | hmodes | i haven't really played with it personally tho |
06:26.02 | *** join/#asterisk terraHome (~cjs@cs662586-139.satx.rr.com) |
06:26.32 | techie | haha |
06:26.40 | hmodes | also, cvs is always preferred over a package in my mind |
06:27.02 | ayano | okay, I'm going to try to screw it up, and I'll let you know what happens hmodes |
06:27.20 | hmodes | good luck :) |
06:27.47 | ayano | thanks. your my hero for the night.... :) |
06:28.04 | pranav | hello mr.qwall when i call to fwd number it says it is invalid extension |
06:28.38 | terraHome | ugh, why the hell is russia selling syria weapons |
06:28.48 | terraHome | i thought russia hated islamic terrorists. |
06:29.00 | Qwell | terraHome: Because they were getting mad at my making a profit on them |
06:29.06 | terraHome | heh |
06:29.13 | pranav | hello mr.qwall when i call to fwd number it says it is invalid extension |
06:29.22 | wasim | terraHome: so does the US, but that doesn't stop them from arms sales to "terrorists" |
06:29.25 | terraHome | hah |
06:29.41 | terraHome | we do not sell arms to terrorists. |
06:29.43 | Qwell | pranav: again, I saw that the first time. No need to repeat |
06:29.54 | terraHome | not this administration, anyway |
06:29.58 | pranav | ok sorry |
06:31.01 | wasim | terraHome: hah ... you really for that one |
06:31.06 | wasim | terraHome: fall, even |
06:31.25 | terraHome | wasim, where are you located? |
06:32.01 | pranav | Qwell: i have pasted my sip.conf and extentions.conf in pastebin.ca/6001 |
06:32.03 | wasim | terraHome: pk |
06:32.08 | terraHome | hah |
06:32.22 | Qwell | pranav: I can't help you tonight. Sorry |
06:32.42 | Qwell | If it were noon, and I was free for the rest of the day, maybe |
06:33.20 | terraHome | anyone have an IAXy? |
06:33.22 | pranav | ok some one else please tell me whats the mistake |
06:33.24 | terraHome | or play with one? |
06:34.19 | *** join/#asterisk murangd (~nukaidc@pool-162-84-229-224.ny5030.east.verizon.net) |
06:34.54 | terraHome | is it good? |
06:35.42 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
06:35.42 | *** topic/#asterisk is Asterisk: The Open Source PBX || Dev Conf 2PM CST FEB 17th -> IAX2/guest@66.250.68.194/996 |
06:36.20 | `Sauron | I want cheap FXO interfaces |
06:36.49 | Sedorox | how many? |
06:37.03 | pranav | Sedorox: currently i am not using iax to make calls, i have 2 sipura phones , from where i want to make fwd calls |
06:37.18 | terraHome | ugh, the iaxy needs a power supply (not included) |
06:37.59 | Qwell | Sedorox: Sorry for pawning that off, couldn't deal with it, heh |
06:38.04 | SuperMMan | anyone see this http://technology.sympatico.msn.ca/Home/ContentPosting.aspx?contentid=0653c17f5bd348bbb4d3405dc821fa12&show=False&number=4&showbyline=False&subtitle=&detect=&abc=abc |
06:38.45 | pranav | Sedorox:if you don't mind , just have a look at pastebin.ca/6001, i have pasted the sip.conf and the extensions.conf |
06:38.52 | hmodes | terraHome: it's like $10 at radiohack |
06:38.57 | Sedorox | Qwell: 'tis fine :-p |
06:39.04 | Sedorox | pranav: I looked at it.... |
06:39.09 | pranav | ok |
06:39.18 | terraHome | i dread going to radioshack |
06:39.37 | pranav | so is there anything wrong in that? |
06:39.37 | *** part/#asterisk SuperMMan (~graphic@d209-89-191-155.abhsia.telus.net) |
06:39.40 | hmodes | i imagine if you ask nice enough digium can prolly provide powah |
06:39.40 | Sedorox | ummmm |
06:40.10 | Sedorox | the only thing I can think of is your gonna want that format.. the username:password@fwd/${exten} but let me look something up |
06:40.57 | Firestrm | terraHome, digikey is your friend .. |
06:41.16 | Firestrm | terraHome, better than radioscrap any day of the week |
06:41.51 | pranav | when i call fwd number(say 7612) the ring comes but then after say 3 to 4 rings it says "extension invalid" |
06:41.55 | Firestrm | terraHome, www.digikey.com |
06:41.55 | terraHome | digikey... |
06:41.58 | terraHome | ok |
06:42.07 | Firestrm | they have EVERYTHING! |
06:42.11 | Firestrm | allmost.. |
06:43.02 | hmodes | they have an annoying minimum order fee tho ;p |
06:43.06 | Firestrm | they really like me :) |
06:43.11 | hmodes | i keep getting bitten by it when i order flash |
06:43.18 | implicit | hehehe |
06:43.34 | Firestrm | hmodes, never been a problem for me.. all my orders are around 1000.00 ea |
06:43.42 | hmodes | thbbbpt |
06:43.45 | hmodes | lucky you! |
06:43.45 | `Sauron | firestrm: What you order? |
06:43.52 | *** part/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
06:44.18 | hmodes | although even the minimum charge is worth it for most of the obscure shit they can source |
06:44.39 | terraHome | what do you buy from digikey, firestrm |
06:44.49 | terraHome | well, here's the next problem with the IAXy: |
06:44.52 | terraHome | what phone do i use? :) |
06:44.54 | Firestrm | `Sauron, i run a small prototype electonics design/assembly company on the side, i think at some point ive ordered justa bout everthing |
06:45.02 | `Sauron | ah |
06:45.04 | `Sauron | that's cool |
06:45.13 | hmodes | terraHome: i have a pulse dial western electric on mine :) |
06:45.15 | terraHome | i was thinking about one of those really old Bell System desk phones from the late 70s/early 80s |
06:45.23 | terraHome | IAXy can handle pulse? |
06:45.41 | hmodes | yup |
06:45.53 | hmodes | it even converts pulse to dtmf for ivrs |
06:46.00 | terraHome | i'll need a wireless bridge too |
06:46.01 | terraHome | :/ |
06:46.19 | hmodes | linksys is your friend? |
06:46.24 | terraHome | the idea for this phone is as an emergency phoen for home when my cell runs out of batteries |
06:46.31 | terraHome | how much are 802.11b bridges nowdays? |
06:46.51 | hmodes | i don't think anyone even sells b bridges anymore ;p |
06:46.56 | hmodes | a g bridge is prolly about $40 |
06:47.10 | hmodes | multi-mac bridging mebbe $50 or $60 |
06:47.11 | *** join/#asterisk shidan (~shidan@CPE000e08eaf90e-CM014280007905.cpe.net.cable.rogers.com) |
06:47.16 | `Sauron | I have a b/g bridge |
06:47.17 | Firestrm | `Sauron, www.vrl.ca |
06:47.29 | `Sauron | because they wanted $169 for a a/b/g bridge |
06:47.30 | terraHome | http://www2.panasonic.com/webapp/wcs/stores/servlet/vModelDetail?displayTab=O&storeId=11251&catalogId=11005&itemId=70256&catGroupId=20952&modelNo=KX-TG5200M&surfModel=KX-TG5200M |
06:47.30 | `Sauron | Grr |
06:47.32 | terraHome | ugh |
06:47.37 | Sedorox | http://www.newegg.com/app/ViewProductDesc.asp?description=33-156-154&depa=0 |
06:47.41 | Sedorox | B Bridge |
06:47.48 | hmodes | eff a |
06:47.50 | terraHome | i need a super-cheap bridge |
06:47.52 | terraHome | an IAXy |
06:47.56 | murangd | I don't understand why doesn't they make voip using |
06:47.58 | murangd | UDP |
06:48.00 | hmodes | and if you're getting b, g can't possibly be much more |
06:48.00 | murangd | or TCP |
06:48.04 | murangd | why SIP or IAX? |
06:48.16 | `Sauron | I'd rather have A than g |
06:48.23 | `Sauron | G blows |
06:48.29 | terraHome | A does not go through walls as well |
06:48.31 | hmodes | well, in a closed environment |
06:48.34 | Sedorox | I think A blows... |
06:48.36 | hmodes | i have friends tho ;p |
06:48.39 | terraHome | i have a G router |
06:48.43 | terraHome | but only a B card in my mac |
06:48.50 | terraHome | so i'm fine w/ a B bridge |
06:48.59 | Sedorox | I have all G stuff here now... save for a B CF for my zaurus |
06:49.03 | terraHome | of course, if i get that phone, i don't need the bridge |
06:49.10 | terraHome | i'll just plug the base station into my switch |
06:49.12 | `Sauron | the problem with b/g is if you have a G network, and a B device starts talking, ALL the devices drop to B |
06:49.20 | `Sauron | because b/g is on the same spectrum |
06:49.24 | terraHome | yep |
06:49.27 | Sedorox | `Sauron: I didn't have that problem |
06:49.38 | `Sauron | Sedorox: You do, you just don't notice it. |
06:49.42 | Sedorox | when my zaurus is on.. everything still says 54... |
06:49.50 | `Sauron | You can't do both B and G at the same time |
06:49.52 | hmodes | and that's when I kick the b user in the sack and verbally berate them in front of their peers ;p |
06:50.10 | hmodes | and then they upgrade |
06:50.13 | hmodes | and everyone is happy |
06:50.28 | Sedorox | lol |
06:50.47 | letherglov | hmodes, no no no |
06:50.55 | letherglov | first, you steal their music from itunes |
06:51.00 | letherglov | THEN you berate them |
06:51.04 | terraHome | hmmmmm |
06:51.17 | hmodes | but it would be so slooow |
06:51.21 | hmodes | it's almost not even worth it |
06:51.22 | murangd | can someon eanswer my question |
06:51.28 | `Sauron | Firestrm: Done any FPGA stuff? |
06:51.46 | terraHome | so i'm debating on cordless phone or old Bell phone + 802.11[bg] bridge |
06:51.55 | Firestrm | `Sauron, yes, a long time ago.. |
06:51.55 | terraHome | im leaning towards the bell phone |
06:52.09 | `Sauron | webpage says you do asic stuff |
06:52.28 | `Sauron | I'd imagine fpga stuff would be more common, since they were sort of made to take over the asic stuff |
06:53.08 | `Sauron | s/common/recent |
06:53.11 | letherglov | terraHome, you left the ata out of there |
06:53.22 | letherglov | oh, and |
06:53.29 | terraHome | lether, i'll use an IAXy |
06:53.32 | letherglov | you'd better check if your microwave is going to interfere |
06:53.38 | terraHome | oooo, what about a PAYPHONE |
06:53.41 | Firestrm | `Sauron, the asic stuff was when i was doing avionics design. |
06:53.43 | letherglov | you don't want to be making yourself a frozen burrito and loose your girlfriend too |
06:53.48 | terraHome | i could put a payphone in my bedroom next to my bed |
06:53.56 | terraHome | and use it as my talking-in-bed-at-night phone |
06:54.09 | terraHome | hooked up to the IAXy and wifi bridge |
06:54.15 | letherglov | better yet |
06:54.20 | letherglov | you can steal from the x-men |
06:54.22 | letherglov | and get..uhhh.shit |
06:54.26 | letherglov | what's that thing called? |
06:54.34 | `Sauron | Firestrm: Ah. |
06:54.34 | letherglov | memorino; |
06:54.36 | letherglov | ;-) |
06:58.02 | *** join/#asterisk c00ljack (c00ljack@202.69.190.247) |
06:58.19 | c00ljack | hi |
06:58.50 | terraHome | ok, here's a geeky question |
06:58.54 | terraHome | if you take an IAXy |
06:58.56 | c00ljack | to whom can seek assistance re: quiuntum integration? |
06:58.58 | terraHome | and put a pigtail out of it |
06:59.14 | terraHome | and wire that pigtail into your home phone wiring |
06:59.20 | Sedorox | ew |
06:59.27 | letherglov | uh |
06:59.28 | terraHome | is it enough to handle all of the phones at home (albeit one at a time) |
06:59.33 | letherglov | how many REN you think it puts out? |
06:59.44 | terraHome | dunno |
06:59.45 | Sedorox | ummm... as long as you make sure you are disconnected from the PSTN |
06:59.51 | terraHome | im not on the PSTN |
06:59.53 | Sedorox | and yea.. it may not ring all the phones... |
07:00.04 | letherglov | either that or it browns out and reboots |
07:00.05 | terraHome | i have no PSTN service at home |
07:00.09 | letherglov | oh |
07:00.16 | letherglov | live next to ted kazinsky? |
07:00.22 | hmodes | ehh, as long as it's under 2.5 or 3 i bet it can do it with a beefy power brick |
07:00.23 | terraHome | has anybody ever tried this? |
07:00.27 | terraHome | lether, basically |
07:00.38 | letherglov | how do you connect to the internet? |
07:00.41 | terraHome | i have a huge 2br apartment with virtually nothing in it |
07:00.42 | Sedorox | I know people do it with a cisco ATA.. so dunno |
07:00.43 | terraHome | i have: |
07:00.56 | terraHome | a couch, a bed, a chest of drawers, and a desk |
07:00.57 | Sedorox | I'm sure it'll support 2 phones... |
07:00.58 | terraHome | and that's it |
07:01.09 | hmodes | i actually power about 1000sq ft with an iaxy |
07:01.11 | terraHome | sed, is the length of the home wiring gonna be an issue? |
07:01.16 | hmodes | and prolly about 2ren |
07:01.25 | hmodes | mebbe 1.5 or so |
07:01.25 | `Sauron | I know the spa-1001 can do like 10 ren's |
07:01.26 | terraHome | REN = ??? |
07:01.27 | `Sauron | or something |
07:01.32 | letherglov | where's your evil typewriter for your manifesto |
07:01.34 | hmodes | ringer equivilency or some crap |
07:01.37 | letherglov | ringer equivalency number |
07:01.40 | Sedorox | I don't think so... I think it really just matters on how many phones... |
07:01.42 | letherglov | it's the amount of current the phone draws |
07:01.43 | terraHome | lether, heh |
07:01.44 | letherglov | when it rings |
07:01.47 | terraHome | no typewriter |
07:01.55 | `Sauron | you can configure the max. ren load in software |
07:02.01 | terraHome | though i have met a lady who was blown up by the unibomber |
07:02.05 | hmodes | i think 1.0 == oldschool bell phone |
07:02.12 | letherglov | what software? |
07:02.22 | hmodes | so 2-3.0 would be WAY more then enough to ring multiple modern powered ringers |
07:02.32 | hardwire | any idea why linphone under debian shows no codecs? |
07:02.33 | letherglov | well |
07:02.38 | letherglov | if it's an actual ringer |
07:02.39 | terraHome | hmodes, im talking about an old ass Western Electric telephone |
07:02.43 | letherglov | versus the ringer on a powered phone |
07:02.43 | letherglov | sure |
07:02.44 | terraHome | but i will only have one of them |
07:02.48 | hmodes | oh, yeah |
07:02.51 | letherglov | if it's a cordless thing |
07:02.52 | hmodes | i can run one of them :) |
07:02.56 | letherglov | than it's got it's own power supply |
07:02.58 | hmodes | haven't really tried two |
07:03.02 | letherglov | so it probably doesn't draw much |
07:03.07 | terraHome | thee point of using the home wiring is that i dont have to have the IAXy in the room with the phone |
07:03.09 | hmodes | but one oldschool phone and extension to *.modern seems stable enough |
07:03.22 | letherglov | I have one too |
07:03.24 | letherglov | old cortelco |
07:03.26 | letherglov | I use it at home |
07:03.28 | terraHome | would this work: |
07:03.30 | letherglov | and I'm far away |
07:03.36 | letherglov | I can wake anyone in the house with it |
07:03.38 | letherglov | it's loud as fuck |
07:03.49 | terraHome | IAXY <----rj11---> wall jack <--------> wall jack <---- rj11----> telephone |
07:03.51 | `Sauron | terrahome: So you're hooking the iaxy instead of the NT interface box? |
07:04.00 | terraHome | yeah sauron |
07:04.09 | terraHome | i want to plug the IAXy into a wall jack in the computer room |
07:04.16 | `Sauron | I need to find my NT box so I can do that. |
07:04.16 | terraHome | and a regular old phone into the wall jack in the bedroom |
07:04.18 | hmodes | terraHome: almost definately, unless you have the shittiest wiring on earth |
07:04.26 | terraHome | fucking a. word. |
07:04.54 | terraHome | that would save me the cost of the wifi bridge |
07:05.07 | terraHome | i buy a cheap old western electric phone for the bedroom and an IAXy |
07:05.27 | `Sauron | dum di dum |
07:05.30 | `Sauron | wonder where my box is |
07:05.31 | terraHome | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=985&item=6153534553&rd=1 |
07:05.33 | terraHome | that. |
07:06.20 | hmodes | yeah, i have the same thing in rotory |
07:06.25 | hmodes | the iaxy drives it happily |
07:06.43 | terraHome | you have both plugged into different wall jacks in the same house? |
07:06.56 | terraHome | oh, this is going to be sweet |
07:07.02 | hmodes | i think i had a splitter on the iaxy |
07:07.02 | terraHome | i just need to pick out the ideal phone now |
07:07.07 | Sedorox | lol |
07:07.16 | hmodes | but i don't have my wall jack -> apartment nid disconnected |
07:07.26 | Sedorox | hmmm |
07:07.27 | hmodes | which is probably a good 150' of bleeding copper |
07:07.32 | *** join/#asterisk shidan (~shidan@CPE000e08eaf90e-CM014280007905.cpe.net.cable.rogers.com) |
07:07.34 | hmodes | and it's happy with it |
07:08.13 | `Sauron | hmodes: Just watch, there's a line at the CO that rings everytime you get a call... :) |
07:08.39 | hmodes | that would be highly amusing |
07:08.58 | terraHome | i need to make sure to disconnect my apartment from the PSTN |
07:09.00 | hmodes | but i had to find my line when i ran dsl, so i know it goes nowhere |
07:09.17 | `Sauron | nod |
07:09.26 | hmodes | yeeah, driving the pstn takes rather uber power :) |
07:09.27 | `Sauron | I'll find my nid and disconnect it |
07:09.34 | hmodes | granted i would love to light that shit up |
07:09.34 | `Sauron | or wait for the CO to axe it |
07:09.34 | terraHome | hahahah |
07:09.43 | hmodes | but it's cheaper to just run my home |
07:09.46 | terraHome | "why is my nufone balance all gone?" |
07:11.01 | terraHome | dammit! |
07:11.05 | terraHome | i can't find the TNI |
07:11.14 | terraHome | there is a cable junction box for CATV |
07:11.20 | terraHome | but nothing for PSTN |
07:11.26 | `Sauron | terra: at a house, or apt? |
07:11.32 | terraHome | olllllld apartment |
07:11.39 | terraHome | like from teh 50s |
07:11.42 | terraHome | 40s |
07:11.47 | `Sauron | bribe one of the maintenance guys with a 6-pack of beer |
07:11.49 | `Sauron | he'll find it for you |
07:11.50 | `Sauron | :) |
07:12.22 | Firestrm | `Sauron, i find twinkies work best ;) |
07:12.29 | terraHome | will it cause problems to just plug the IAXy into the wall lines? |
07:12.34 | terraHome | having no idea where they go |
07:12.44 | Firestrm | ummmm... yes |
07:12.51 | hmodes | i wouldn't think it would damage the iaxy |
07:12.54 | Sedorox | terraHome: only if they are still connected to the pstn source |
07:12.57 | hmodes | tho you may want to check with mark |
07:13.03 | Sedorox | otherwise... if you know for a fact that they are |
07:13.04 | `Sauron | well, you have to make sure the CO is disconnected from your wiring |
07:13.04 | Sedorox | it shouldn't be |
07:13.10 | hmodes | if it openly doesn't work definately don't leave it plugged in tho |
07:13.14 | `Sauron | dunno what 108V ring voltage would do to an iaxy |
07:13.30 | Sedorox | lol |
07:13.46 | hmodes | well i would assume you're not expecting the wall jack to ring |
07:14.07 | hmodes | if you're plugging one in to a jack that's live, you deserve to have it fried ;p |
07:14.18 | Sedorox | lol |
07:14.31 | terraHome | i'm pretty damned sure that the CO is still physically connected |
07:14.32 | c00ljack | hey guys any idea how can connect quintum to asterisk? |
07:14.38 | terraHome | because its an apartment. |
07:15.00 | `Sauron | Oh well |
07:15.01 | `Sauron | sleep time |
07:15.34 | terraHome | hmmm |
07:15.36 | terraHome | maybe its out back |
07:15.40 | terraHome | behind the building |
07:16.02 | hmodes | tone generators are your friend |
07:16.07 | hmodes | and fairly cheap |
07:16.16 | Sedorox | terraHome: talk with whoever you pay for the reent... |
07:16.22 | Sedorox | check to see if you can get it d/c |
07:16.25 | Sedorox | as it may be in the basement |
07:16.31 | Sedorox | or some room you don't have access to |
07:16.32 | Sedorox | lol |
07:16.48 | terraHome | knowing them, they will disconnect all the interior jacks from each other, too |
07:16.53 | hardwire | ok |
07:16.59 | hardwire | junctionnetworks are neat peeps |
07:17.00 | hmodes | so glad my building has a giant well marked bell box on the outside :) |
07:17.09 | hardwire | anybody here from junction networks? |
07:17.19 | terraHome | oh man |
07:17.25 | Sedorox | lol |
07:17.27 | terraHome | remember those multi-line phones from the 70s |
07:17.32 | terraHome | the ones with the big square buttons |
07:17.34 | terraHome | that would light up |
07:17.36 | hmodes | *thunk* |
07:17.40 | terraHome | *that* is what i need |
07:17.49 | *** join/#asterisk jdmjamboo (jdmjamboo@202.69.190.233) |
07:17.59 | hmodes | i feel a general fondness for them in the same way i feel a general fondness for original ps/2 keyboards |
07:18.13 | hmodes | the 80s was totally the decade of *keys go thunk* |
07:18.18 | Sedorox | lol |
07:18.21 | jdmjamboo | hi people |
07:18.32 | jdmjamboo | hows everybodies doing? |
07:18.37 | Firestrm | terraHome, one apartment place i lived in, didn't even lock the telco room.. Phone Phreakers paradise ;) |
07:18.38 | terraHome | thunk keys ruled |
07:18.47 | terraHome | all mechanical things should work the way those phones did |
07:18.53 | *** join/#asterisk denon (denon@synapse.subneural.net) |
07:18.53 | *** mode/#asterisk [+o denon] by ChanServ |
07:19.13 | terraHome | http://www.pensive.org/jeff/mrfone/collect/WE2563.htm |
07:19.14 | terraHome | pingo |
07:19.16 | hmodes | indeed |
07:19.17 | terraHome | bingo |
07:19.20 | hmodes | long live the thunk |
07:19.31 | terraHome | RED HOLD BUTTON!!! |
07:20.03 | terraHome | hmodes, ever seen an old Fender guitar amp? |
07:20.25 | hmodes | mmm, warm nourishing vacuum |
07:20.33 | jdmjamboo | anybody here had configured Asterisk to Quintum interconnection? |
07:20.35 | jdmjamboo | anybody here had configured Asterisk to Quintum interconnection? |
07:20.39 | terraHome | go away, idiot |
07:20.42 | Firestrm | terraHome, one of the local surplus places has a bunch of the really old style phones with the crank thingy on the side.. i was thinking of turning one into an IP phone.. |
07:20.42 | Sedorox | ..... |
07:20.53 | terraHome | firestrm, we have similar ideas |
07:20.59 | terraHome | i want to take a phone like this: |
07:21.00 | terraHome | http://www.pensive.org/jeff/mrfone/collect/WE2563.htm |
07:21.01 | jdmjamboo | please |
07:21.04 | jdmjamboo | help |
07:21.09 | terraHome | and make it into a multi-line IP phone |
07:21.17 | terraHome | but everything needs to be integrated, in-case |
07:21.21 | Firestrm | terraHome, better yet make it a Wifi phone ;) |
07:21.30 | terraHome | or hidden nearby |
07:21.36 | terraHome | exactly |
07:21.54 | terraHome | i want my phone to look and feel like an old clunk-button phone |
07:21.59 | terraHome | but with modern tech |
07:22.24 | jdmjamboo | hi terraHome.. have you encounter quintum to asterisk interconnection? |
07:22.31 | Firestrm | terraHome, i know where there are a pile of the phones picured in the previous link |
07:22.33 | terraHome | how about a Speak-and-Spell <---> LDAP directory interface |
07:22.41 | hmodes | meh, you could pretty easily rewire a 7960 to drive one of them |
07:22.48 | *** join/#asterisk kks (~kks@203.115.210.253) |
07:22.55 | jdmjamboo | hi terraHome.. have you encounter quintum to asterisk interconnection? |
07:23.05 | terraHome | go away, idiot |
07:23.11 | terraHome | fire: where? |
07:23.14 | terraHome | can you get me one or two? |
07:23.15 | hmodes | the question is, would you want to scrap a 7960 to make it work ;p |
07:23.23 | jdmjamboo | help im trying to be nice here... don't be rude |
07:23.26 | jdmjamboo | help im trying to be nice here... don't be rude |
07:23.34 | terraHome | jdm, pay me |
07:23.37 | terraHome | paypal me |
07:23.39 | terraHome | and i will helpyou |
07:23.40 | Firestrm | i saw them in a pile of electronics and telco rubble at capitol iron in victoria |
07:23.44 | jdmjamboo | act like a professional |
07:23.49 | terraHome | i am. pay me. |
07:23.55 | letherglov | hey now |
07:24.07 | letherglov | I've got my hat on the ground, flipped over |
07:24.11 | letherglov | toss some in mine too |
07:24.17 | jdmjamboo | is you just say that before rather than speaking like idiot one |
07:24.18 | terraHome | fire: i'd love to get ahold of a few |
07:24.39 | jdmjamboo | cupal |
07:24.43 | Firestrm | terraHome, i will go get a price for you and let you know.. |
07:24.47 | Sedorox | Question... trying to help pranav with fwd.. when I dial him.. and when he dials me.. its busy.. anyone else having problems with fwd being busy? (he's SIP and I'm iax2...) |
07:25.14 | Firestrm | terraHome, i need to go there to buy my new gps toy.. |
07:25.28 | terraHome | what toy? |
07:25.30 | terraHome | we sell GPS |
07:25.32 | terraHome | gpstools.com |
07:25.40 | Firestrm | garmin 60cs |
07:25.44 | *** join/#asterisk zoa (zoa@82.103.76.147) |
07:26.08 | `Sauron | terrahome: got any trimble gps receivers? |
07:26.11 | terraHome | http://gpstools.com/components/catalog/product.html?pid=506&cat=375 |
07:26.17 | terraHome | sauron, don't think so |
07:26.28 | `Sauron | bah humbug |
07:26.28 | terraHome | free shipping on that GPS, btw |
07:26.29 | Firestrm | im about to drop some cash down on it.. they are the cheapest in town.. $450.00 |
07:26.37 | `Sauron | garmin can eat my shorts, trimble's where it's at |
07:26.50 | *** join/#asterisk cc (~cc@byte.fedora) |
07:26.53 | terraHome | its hard for us to compete w/ the big box stores on garmin |
07:26.59 | `Sauron | The funny thing is, y'alls $400 garmin GPS doesn't have more than $150-200 worth of parts inside |
07:27.04 | terraHome | theybuy in mega-quantities |
07:27.06 | Firestrm | `Sauron, garmin saved my life more times than i can count.. im a garmin man foreer |
07:27.18 | terraHome | i like garmin just fine |
07:27.21 | terraHome | works good enough |
07:27.28 | terraHome | my buddies in Iraq use them every day |
07:27.32 | Firestrm | when you are here.. www.vrl.ca/ocarc, you want garmin. |
07:27.46 | `Sauron | blah blah blah |
07:27.48 | CMike | oh |
07:28.01 | `Sauron | bust open your garmin unit |
07:28.07 | CMike | nice repeater |
07:28.11 | `Sauron | I bet the gps receiver inside, is made by trimble :) |
07:28.16 | Firestrm | CMike, thanks |
07:28.21 | terraHome | nothing wrong with trimble or garmin |
07:28.25 | terraHome | both are good products |
07:28.44 | `Sauron | I'm about to order a couple Lassen iQ's from trimble |
07:28.46 | CMike | what software did that coveragemap? |
07:28.50 | letherglov | whoa |
07:28.54 | letherglov | what's that huge steel penis? |
07:29.04 | CMike | I have a few lassens.. |
07:29.09 | `Sauron | $46 for a full featured GPS receiver, 12 channels, supplies full PVt and everything |
07:29.12 | CMike | http://www.trimble.com/lassenlp.html thoose I think |
07:29.21 | Firestrm | letherglov, thats what we call it.. except its fiberglass |
07:29.33 | letherglov | radome? |
07:29.45 | CMike | Firestrm: you built that repeater ? |
07:29.50 | `Sauron | CMike: I think they're discontinuing the SQ and LP, and replacing them with the iQ |
07:29.55 | CMike | oh |
07:29.58 | Firestrm | CMike, yes me and one other guy.. |
07:30.14 | CMike | looks like a fun project. |
07:30.20 | `Sauron | the iQ has specs overall, including power drain, which is what the LP was made for |
07:30.36 | CMike | Hm.. darn.. Now I have to go there and try the repeater.. :P |
07:30.41 | *** join/#asterisk schurig (~schurig@p5080A330.dip0.t-ipconnect.de) |
07:30.51 | Firestrm | CMike, till it breaks in the middle of winter, and its the main SAR repeater.. |
07:31.18 | CMike | <-- waiting for my callsign.. |
07:31.18 | CMike | :P |
07:31.19 | terraHome | firestrm, you need a weather station from us :) |
07:31.19 | Firestrm | CMike, the coverage is AWESOME!!.. |
07:31.21 | terraHome | weathertools.com |
07:31.23 | Beirdo | Firestrm: that's about where I wanna move some decade |
07:31.31 | Beirdo | if I do, I'll use yer repeater :) |
07:31.35 | Firestrm | CMike, VE7GEI |
07:31.39 | Beirdo | VA3HGJ (with no radios) |
07:31.49 | CMike | I probably get SA0??? |
07:32.22 | CMike | Sweden just started useing SA as prefix.. SM was apperently full.. |
07:32.25 | CMike | :) |
07:32.40 | Firestrm | Beirdo, enjoy :) its the best repeater for the okanagan valley and beyond, i can pick it up 200km away in nakusp |
07:32.46 | terraHome | firestrm: 866-859-7359 |
07:33.17 | Firestrm | terraHome, I would LOVE a weather station for the repeater.. |
07:33.18 | letherglov | http://www.garmin.com/products/rino/positionReport.html |
07:33.20 | letherglov | now that's cool |
07:33.25 | terraHome | firstrm: call that number |
07:33.28 | terraHome | its toll free |
07:33.33 | Beirdo | My dream (post lottery win) is to move out there and start a winery |
07:33.34 | terraHome | its my rooftop weather station |
07:33.37 | Firestrm | terraHome, now? |
07:33.42 | terraHome | excuse my horrible texas accent |
07:33.43 | terraHome | sure |
07:33.45 | Firestrm | ahhh.. cool |
07:33.53 | `Sauron | text-to-speech? |
07:34.02 | terraHome | call and see :) |
07:34.16 | `Sauron | I'm scared of y'all texans :) |
07:34.33 | terraHome | yes, i have a gun. yes, i like country music. |
07:34.36 | terraHome | yes, i love barbecue |
07:34.41 | CMike | :) |
07:34.46 | terraHome | and yes, i dip copenhagen snuff. |
07:34.54 | Beirdo | Yes, your ex-governor's haunting the world |
07:34.54 | letherglov | I bet you voted for bush too |
07:34.55 | Firestrm | terraHome, cant call it from canada :( |
07:35.01 | terraHome | hahah beirdo |
07:35.09 | terraHome | can't call 1866 from .ca eh |
07:35.11 | terraHome | hrmmmm |
07:35.19 | terraHome | i dont have a local DID |
07:35.29 | terraHome | hrmmm |
07:35.31 | terraHome | maybe i need one |
07:35.31 | Beirdo | not unless you get US50/CDN toll-free |
07:35.34 | `Sauron | you can hop through FWD |
07:35.39 | Beirdo | that one's likely US48 |
07:35.46 | terraHome | how hard is it to get FWD up and running? |
07:35.48 | Firestrm | i might be able to route it though iconnect.. |
07:35.49 | terraHome | does it take long? |
07:35.58 | `Sauron | terra: I had it up in like, 10 minutes |
07:36.01 | terraHome | shit, i need to sign up for a DID, anyway |
07:36.07 | terraHome | FWD url? |
07:36.14 | Firestrm | Beirdo, any idea how much a us50/can tollfree costs? |
07:36.15 | terraHome | nm |
07:36.37 | Beirdo | if you are lucky and can find them, not much |
07:36.40 | `Sauron | Hum |
07:36.40 | Firestrm | terraHome, ya.. i can route through FWD.. let me try |
07:36.44 | `Sauron | that's not bad texas accent |
07:36.45 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
07:36.50 | moonwick | Firestrm: what are the grey tubes in http://www.vrl.ca/ocarc/images/DSC00129.JPG ? |
07:36.54 | moonwick | batteries? |
07:37.05 | Beirdo | rumour has it that livevoip.com has them cheap |
07:37.07 | CMike | Oh well.. gotta work .. local time 08:37 am |
07:37.08 | CMike | ...73 |
07:37.11 | `Sauron | I was expecting the full southern drawl |
07:37.12 | `Sauron | :) |
07:37.23 | Beirdo | he said Texas, not Alabama |
07:37.37 | terraHome | i have a texan accent |
07:37.43 | terraHome | not necessarily southern i guess |
07:37.45 | Beirdo | or like my bud in Tennessee. silly redneck :) |
07:37.46 | kks | hey guy, it may be stupid question, where i can set the SMSC number? |
07:37.48 | terraHome | we have our own accent |
07:38.01 | Beirdo | yes you do |
07:38.06 | terraHome | somebody just got the weather station :) |
07:38.09 | Beirdo | and your own history too |
07:39.27 | `Sauron | I called it |
07:39.44 | hmodes | ... |
07:39.46 | `Sauron | I was going to make something similar |
07:39.53 | `Sauron | 'cept, have it all be text-to-speech |
07:39.54 | terraHome | those weather conditions are updated every 5s |
07:42.04 | Firestrm | terraHome, thats a cool system.. you dont have much of an accent, (or my alberta accent cant diferentiate between alberta accent and texas accent) |
07:43.00 | Firestrm | alberta/texas... brothers. not much different.. except texans dont carry as much firepower :) |
07:43.06 | terraHome | hahah |
07:43.07 | Beirdo | heh |
07:43.18 | Beirdo | Firestrm: Alberta is Canada's Texass |
07:43.23 | Beirdo | er Texas |
07:43.28 | Beirdo | freudian slip |
07:43.29 | Firestrm | absolutly |
07:43.50 | Beirdo | now if only UAlberta would call me for an interview :) |
07:43.58 | terraHome | think again |
07:43.58 | terraHome | http://chrissnell.com/my_guns.jpg |
07:44.17 | Firestrm | you know when your alberta when... all the highway deer crossing warning signs are shot to hell.. |
07:44.37 | terraHome | the *school bus stop ahead* signs are shot to hell here |
07:45.06 | Beirdo | Of course now the Oilers may not exist long enough for me to see a home game SHOULD I get hired out there |
07:45.13 | Beirdo | and screw the Flames very much :) |
07:45.25 | Firestrm | terraHome, albrta too.. except they try to target the picture of the child's head |
07:45.42 | Beirdo | can you tell I'll fit in in Edmonton? :) |
07:45.45 | terraHome | thats my favorite one of those handguns |
07:45.54 | terraHome | because my grandfather carried it in WWII |
07:46.12 | terraHome | its also the most badass of the four |
07:46.20 | Beirdo | my grandfather built bombers |
07:46.27 | terraHome | i want to sell that Sig |
07:46.31 | terraHome | but it won't fetch enough |
07:46.43 | Sedorox | night all.. or should I say morning |
07:46.44 | Sedorox | :-p |
07:46.50 | Beirdo | ye olde Lancaster (dam busters) |
07:47.19 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
07:48.57 | Firestrm | terraHome, this is my gun http://www.snipercentral.com/pm_sm.htm |
07:49.22 | *** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se) |
07:49.29 | Firestrm | $6000.00 and worth every penny |
07:49.44 | terraHome | *CLI> Feb 17 01:49:07 NOTICE[14399]: chan_iax2.c:5869 socket_read: Registration of '616306' rejected: Registration Refused |
07:49.48 | terraHome | ugh, FWD doesn't like me |
07:50.11 | Firestrm | terraHome, i think its FWD.. im getting the same messages lately |
07:50.26 | Firestrm | terraHome, it was working yesterday.. |
07:50.30 | terraHome | nice weapon |
07:51.24 | moonwick | yeah, that's a texan accent :) |
07:51.25 | *** join/#asterisk eivindtr (~Eivind@193.91.146.34) |
07:51.26 | Firestrm | terraHome, dropped a 4point buck from a treestand this year.. 450 yd shot, took out the heart, dead center where i put the shot |
07:52.16 | Firestrm | terraHome, i have the 300 win sm version |
07:52.33 | terraHome | dang. |
07:52.55 | terraHome | im about 4/10 on the M16A4 @ 300yds |
07:53.00 | Firestrm | terraHome, he was just finishing up with his girlfriend, i waited to he could die happy :) |
07:53.39 | Beirdo | Firestrm: take it to Ottawa and see if you can't get rid of some Liberals so the Conservatives can finally rule, eh? |
07:54.14 | Firestrm | Beirdo, don't tempt me.. im just bairly able to restrain myself.. |
07:54.59 | terraHome | dammit FWD |
07:55.01 | terraHome | register, you bastard |
07:55.15 | Beirdo | hehe |
07:55.18 | Firestrm | terraHome, its broken. im not registering either.. |
07:55.32 | terraHome | ok |
07:57.41 | Firestrm | terraHome, if i buy from gpstools, what courier do they use for shipping? |
07:58.01 | terraHome | oh duh, i forgot to activate my iax2 account |
07:58.13 | terraHome | fire, well for your GPS, we'd probably use FedEx Ground |
07:58.18 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
07:58.19 | terraHome | in fact |
07:58.21 | pranav | can anyone help me with the fwd stuff |
07:58.21 | terraHome | i know we would |
07:58.22 | Firestrm | terraHome, that would also do it.. although i still think they are broken.. |
07:58.55 | pranav | when i call to an fwd number it rings but then there is no response |
07:59.02 | terraHome | fire, by default, we don't accept international orders...but if you get ahold of me tomorrow i can hook you up |
07:59.06 | Firestrm | terraHome, ground not the best for canada.. i get assraped for brokerage fees. |
07:59.11 | terraHome | yep |
07:59.16 | terraHome | FedEx International we can ship |
07:59.20 | terraHome | though its not free :) |
07:59.33 | terraHome | our FedEx rates are *very* cheap |
07:59.39 | terraHome | see what we charge people for overnight |
07:59.42 | terraHome | US$7.99 |
07:59.46 | terraHome | won't find that anywhere. |
08:00.19 | Firestrm | ya no kidding.. |
08:00.43 | pranav | hi terrahome |
08:01.58 | terraHome | you just paypal the money and we ship the same day |
08:02.12 | terraHome | oops that was for privmsg |
08:05.41 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
08:06.56 | pranav | hi Zeeek |
08:07.03 | terraHome | firestrm, the Davis Instruments stuff rules |
08:07.08 | terraHome | there is a nice Perl module |
08:07.12 | terraHome | Device::WxM2 |
08:07.16 | terraHome | i contributed code for it |
08:07.24 | terraHome | makes it easy to pull data off the device |
08:07.27 | terraHome | and then you just AGI it |
08:07.44 | Zeeek | good morning gentlemen |
08:07.55 | wasim | morning Zeeek, any packages from .de? |
08:07.55 | terraHome | i wrote a little daemon in perl that pulls the data off the serial port and stores it in a Storable |
08:08.03 | terraHome | and my AGI just reads the storable |
08:08.13 | Firestrm | terraHome, nice, i will have to pull it into a serial stream, so i can transmit it over the radio |
08:08.38 | *** join/#asterisk djin (~marius@62.58.40.196) |
08:08.39 | Firestrm | terracon, does the windspeed go up to 100mph? |
08:09.34 | Zeeek | Is there a way to hit a key during a call to interrupt it, hang up and execute a "wait for DMTF" section? |
08:09.57 | Zeeek | wasim - been waiting for you heh heh - nothing ATM |
08:10.13 | Zeeek | it would be great for today or tomorrow |
08:10.38 | Zeeek | but was something sent? Das ist das Qveschun |
08:11.14 | Zeeek | wasim, Das ist das Qveschun |
08:11.18 | terraHome | firestrm, im not sure |
08:11.27 | terraHome | im sure it does |
08:11.43 | wasim | Zeeek: ok, now its beginning to piss me off to |
08:11.58 | Zeeek | what did I di ? :) |
08:12.00 | Firestrm | terraHome, ive personally been in 60mph winds up there, at that wasnt even with a storm.. clear sky.. |
08:12.11 | wasim | Zeeek: no, no, not you, me |
08:12.20 | Zeeek | wasim email me the phone number and I'll call them if you want |
08:12.22 | terraHome | http://www.weathertools.com/components/catalog/product.html?pid=357&cat=276 |
08:12.26 | terraHome | that's a good one for you |
08:12.38 | terraHome | http://www.weathertools.com/static/products/davis/6150_spec.pdf |
08:13.17 | *** join/#asterisk chaoscon_ (~ph33r@chaoscon.user) |
08:14.06 | Firestrm | terracon, im a little suspious about the tiny strip of what appear to be plastic holding the windspeed/direction unit.. |
08:14.44 | Zeeek | wasim tell me about it! |
08:14.56 | Firestrm | we have had solar panel's ripped off by the wind, that were supported with 1" steel strapping and 5/8 bolts |
08:15.08 | Zeeek | wasim I've enabled queries if you wanna msg me |
08:16.29 | Beirdo | night |
08:17.34 | terraHome | fire, we can get all sorts of long-range radios for the davis stuff too |
08:17.34 | terraHome | firestrm, do you have a computer at your station? |
08:17.34 | terraHome | it would be nice to have a solar-powered Soekris box + Davis station |
08:17.34 | terraHome | firestrm, you'd also want a heated rain bucket ideally |
08:17.34 | terraHome | and a webcam |
08:17.35 | terraHome | to check snow depth |
08:17.38 | terraHome | webcam + yardstick |
08:18.06 | moonwick | heh |
08:18.19 | terraHome | hey, someone with FWD |
08:18.21 | terraHome | try this #: |
08:18.29 | terraHome | 616306 |
08:19.35 | terraHome | anyone? |
08:19.52 | Zeeek | what happens? |
08:19.58 | Zeeek | something automatic? |
08:19.59 | terraHome | its my weather station |
08:20.00 | terraHome | yes |
08:20.02 | Zeeek | ok |
08:20.10 | terraHome | just wanna make sure it works |
08:20.10 | Zeeek | iax or sip? |
08:20.17 | terraHome | IAX..shouldn't matter |
08:20.22 | terraHome | thats a FWD number |
08:20.44 | terraHome | *CLI> Feb 17 02:20:04 NOTICE[14399]: chan_iax2.c:5766 socket_read: Rejected connect attempt from 65.39.205.121, requested/capability 0x4/0x4 incompatible with our capability 0xfa00. |
08:20.46 | terraHome | wierd-o |
08:20.49 | terraHome | hrmmm |
08:20.58 | Zeeek | so far not even a ring on iax |
08:21.06 | terraHome | hang on |
08:21.26 | terraHome | hrmmm |
08:21.27 | terraHome | codec |
08:21.39 | Firestrm | terraHome, all i get is fast busy |
08:21.42 | Zeeek | I'm on zaptel btw tried sip |
08:21.49 | Zeeek | one ring and then death |
08:22.01 | Zeeek | a horrible fiery death ending wioth congestion |
08:22.12 | Firestrm | terraHome, 711 only for FWD |
08:22.32 | Firestrm | disallow all |
08:22.42 | Zeeek | correct |
08:23.12 | Firestrm | allow = ulaw, alaw |
08:23.58 | Firestrm | allow=729 == puke on call. |
08:24.26 | Zeeek | depending on versions you may not be able to put many codecs on single line, yes? |
08:25.27 | terraHome | lemme see |
08:25.27 | terraHome | ugh |
08:25.27 | terraHome | cable just went out |
08:25.27 | terraHome | first time...ever. |
08:25.34 | Firestrm | wierd, i cant call *393 numbers through my spa-3000 into my * box.. another bug to squish |
08:25.58 | terraHome | ok |
08:25.59 | terraHome | hang on |
08:26.49 | terraHome | ok try again please |
08:26.56 | terraHome | 616306 |
08:27.01 | Firestrm | same result |
08:27.28 | terraHome | ok hang on |
08:27.39 | terraHome | ok |
08:27.47 | Firestrm | *393 goes though with my wisip.. but blocked by spa-3000.. must be a dialing plan thing on the spa |
08:28.06 | Firestrm | still death. |
08:28.10 | Zeeek | ok works! |
08:28.22 | Zeeek | ok works! |
08:28.34 | Zeeek | ooops hung up after the humidity reading |
08:28.37 | terraHome | sweet! |
08:28.52 | terraHome | yeah, thats how it works now |
08:28.58 | Zeeek | now, you only need someone who gives a shit about the weather 6000miles away :) |
08:29.05 | terraHome | hahahah |
08:29.13 | terraHome | im happy it works tho |
08:29.20 | Zeeek | and maybe put a "thanks for calling" or a commercial at the end |
08:29.21 | terraHome | it will make a nice test # for new FWD users |
08:29.25 | terraHome | i had one, zeek |
08:29.27 | terraHome | took it off |
08:29.31 | terraHome | lemme put it back on |
08:29.51 | Zeeek | how about wait(1) playback(goodbye) |
08:29.53 | terraHome | try now :) |
08:29.57 | Firestrm | hmmm, im still getting fast busy.. |
08:30.00 | Zeeek | I believe you :) |
08:30.08 | terraHome | it worked for zeeek |
08:30.13 | terraHome | wonder why u get that |
08:30.18 | terraHome | i dont even see your call coming in firestrm |
08:31.01 | Firestrm | its connecting to FWD, i can see it in the logs, but it keeps coming back busy.. |
08:31.03 | terraHome | im registered to FWD via IAX.. |
08:31.06 | terraHome | wierd |
08:31.38 | Firestrm | rings though to my other FWD number |
08:31.53 | terraHome | lemme restart again... |
08:32.13 | terraHome | ok |
08:32.32 | Firestrm | same result... fast busy.. |
08:32.42 | terraHome | yeah, i dont even see any console msgs |
08:32.45 | Firestrm | one ring, then fast busy |
08:33.09 | terraHome | just tried FWD's CallMe application |
08:33.10 | terraHome | and it works |
08:33.24 | Firestrm | it most have somehing to do with fwd-iax.. maybe iax-iax dont work.. let me try my softphone |
08:33.35 | terraHome | that's kind of wierd |
08:33.39 | terraHome | im on fwd-iax |
08:33.44 | terraHome | ok, see you now |
08:33.59 | *** join/#asterisk eipi (eipi@153-218-114-200.fibertel.com.ar) |
08:34.05 | terraHome | i guess thats you |
08:34.07 | terraHome | dunno :) |
08:34.08 | Firestrm | that though my softphone.. ie.. no iax |
08:34.32 | Firestrm | my * should be 23927 |
08:34.36 | terraHome | my * connects to FWD via IAX2 |
08:34.45 | terraHome | <PROTECTED> |
08:34.48 | terraHome | that was u |
08:35.17 | Firestrm | that was fwd gateway.. same ip for me as well.. i think iax to iax over FWD is broken.. |
08:35.31 | terraHome | really? |
08:35.31 | Firestrm | sip to iax works though.. |
08:35.53 | Firestrm | when i call sip it goes though.. when i call iax, fast busy.. |
08:35.58 | terraHome | strange. |
08:36.05 | terraHome | FWD's sip servers? |
08:36.44 | Firestrm | terraHome, yes but dont even bother trying to connect to FWD via SIP.. all you'll get is a migrane headache |
08:36.54 | terraHome | ok :) |
08:36.58 | Firestrm | from asterisk that is.. |
08:37.22 | terraHome | i really need to get us a local DID |
08:37.35 | *** join/#asterisk shaZwaz (~sasda@216-236-205-66.reverse.newskies.net) |
08:37.41 | Firestrm | i tried for a week with no luck, intil someone here called me a dumbass for even trying and pointed me to the iax gateway.. |
08:37.41 | shaZwaz | hi room |
08:37.43 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
08:37.44 | terraHome | i could get that one through nufone |
08:37.46 | terraHome | buts its in michigan |
08:38.08 | Zeeek | get a toll-free from nufone |
08:38.14 | Zeeek | I just used IAX and it worked |
08:38.21 | terraHome | we have a toll free from them |
08:38.23 | Firestrm | terraHome, i will have Victoria, Kelowna and Nelson DID's soon. |
08:38.27 | terraHome | but it wont work for our foreign customers |
08:38.36 | shaZwaz | to register to FWD and other Services I have bind SIP to my Public IP ? |
08:39.00 | shaZwaz | why doesn't it bind to 0.0.0.0 |
08:39.16 | Firestrm | terraHome, ive found an investor that wants to give me $$$ to play with asterisk and DID's |
08:39.22 | terraHome | nice. |
08:39.47 | Firestrm | i love it when ppl throw money at me.. i go buy toys like gps :) |
08:39.56 | shaZwaz | implicit you around ? |
08:40.02 | terraHome | ok night all |
08:40.10 | Zeeek | hey why not hook up a GPS and people can call and see where they are? :) |
08:40.15 | Firestrm | terraHome, ya me too.. gnite.. |
08:40.41 | Firestrm | zeeek, www.canaprs.com, enter in ve7gei-2 |
08:41.03 | Zeeek | n ot found |
08:41.32 | Firestrm | http://www.canaprs.net/locate.php?stn=ve7gei-2 |
08:42.26 | Makenshi | Morning.. first Galileo satellite is going to be launched soon :> |
08:42.26 | Makenshi | i think thats the only good thing to have ever come out of the eu |
08:43.00 | Firestrm | zeek or http://www.canaprs.net/locate.php?stn=ve7gei-1 |
08:43.12 | Firestrm | depends on vehicle im in |
08:44.06 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
08:44.11 | Firestrm | you can even send me messages on my radio, but im not sure where that site is anymore.. |
08:44.38 | *** part/#asterisk jdmjamboo (jdmjamboo@202.69.190.233) |
08:45.17 | *** join/#asterisk DonX (don@tool.sparkhosting.net) |
08:45.42 | DonX | How can I find out what timing device asterisk is using? |
08:46.22 | Firestrm | DonX, good Q, i dont know.. but i want to know as well |
08:46.55 | Zeeek | Firestrm what is that? |
08:47.18 | Firestrm | Zeeek, see DonX's question |
08:47.29 | shaZwaz | whu do I have to bind sip to my public IP to register to FWD and other providers |
08:47.41 | shaZwaz | anyone has any idea ? |
08:48.10 | shaZwaz | that leaves my internal sip phones not working |
08:48.14 | Firestrm | shaZwaz, sip-FWD through NAT.. forget it.. i tried for over a week, use IAX |
08:48.34 | Zeeek | no I mean what is the site and the radio messages? I was too busy to look |
08:48.39 | shaZwaz | but its not only FWD .. |
08:48.48 | shaZwaz | its same for voipuser.org |
08:48.49 | Zeeek | I looked but too fast since I was on another site filling something out |
08:49.19 | shaZwaz | unless I bind sip to my static IP my number doesn't register |
08:49.21 | Firestrm | Zeeek, i cant find the radio message site anymore, server is down :( i guess i will have to set up an aprs message server.. |
08:49.38 | Zeeek | but what IS that site? |
08:49.53 | Firestrm | <PROTECTED> |
08:50.02 | Zeeek | funny I do SIP FWD through NAT with no prb |
08:50.02 | Firestrm | ? |
08:50.14 | Zeeek | but not 0.0.0.0 binding |
08:50.23 | shaZwaz | why not ? |
08:50.39 | Firestrm | its an APRS tracker site.. it tracks my location realtime, using gps and packet radio |
08:50.40 | shaZwaz | I need my internal sip phones |
08:50.50 | Zeeek | wait maybe I do but I have only one etheenet interface |
08:51.09 | Zeeek | Firestorm via your cellphone? |
08:51.21 | shaZwaz | Zeeek thats the problems it doesn't work with 0.0.0.0 binding |
08:51.27 | Firestrm | Zeeek, via 2m packet radio 144.390 mhz |
08:51.37 | Zeeek | I was W0DBJ for years |
08:51.44 | Zeeek | ... .... .. _ |
08:51.59 | shaZwaz | is there a way I can use my internal clients while binding sip on my public ip ? |
08:52.08 | shaZwaz | or an other way around ? |
08:52.10 | Zeeek | let my license expire - my First Class Phone too |
08:52.12 | Firestrm | morris code hurts my brain.. |
08:52.26 | Zeeek | oh that's why you're stuck on 2M |
08:52.28 | Firestrm | Zeeek, license never expires in canada.. lifetime.. |
08:52.49 | Zeeek | where I live that's how drivers licenses work. Much more practical |
08:52.56 | Firestrm | Zeeek, untill they open up nocode HF.. which was supposed to be done allready |
08:53.05 | Zeeek | except for when people get on the road in their 70's |
08:53.22 | Firestrm | i cant drive.. im 55 |
08:53.40 | Firestrm | :) |
08:53.53 | Zeeek | learning code should be mandatory. WHat will you do in STeven Sagal disaster film when the survors are trying to tap out H E L P ? |
08:54.19 | Zeeek | you are 55? SHit you are almost old enough for the dean's title here |
08:54.24 | Zeeek | Almost |
08:54.32 | Firestrm | they should learn ascii or baudot |
08:54.51 | Zeeek | baudot is just an ecoding method like a codec |
08:55.01 | Zeeek | or ascii for that matter |
08:55.06 | Firestrm | Zeeek, no im 30's.. think of the song , i cant drive 55.. change the lyrics.. |
08:55.15 | Zeeek | ok I was worried |
08:55.32 | Zeeek | why would anyone over 50 waste their precious time here? |
08:55.48 | Zeeek | like I'm doing right now... |
08:56.23 | Firestrm | im making sure im extra tired for my charter flight in the morning :).. |
08:56.34 | Zeeek | heh |
08:57.21 | Firestrm | i love the look the passengers get when the pilot falls asleep halfway thorough the checklist :) |
08:57.44 | Zeeek | what no alcohol on the breath? |
08:58.47 | Firestrm | Zeeek, no, im a strict 8 hours bottle to throttle kind of pilot.. fatigue has an antidote.. caffeen, alchol also has an antidote.. time.. |
08:59.11 | Zeeek | actually it depends on how much you drink |
08:59.29 | Zeeek | there was an excel calculator making the rounds before the holidays |
08:59.50 | Zeeek | you entered what you drank at what time and it traced the curve of BA (blood alc) |
09:00.14 | *** join/#asterisk dstevens (~dstevens@cpc3-ches1-4-0-cust87.lutn.cable.ntl.com) |
09:00.18 | Firestrm | im well beyond coffee, ive graduated to sucking on unroasted beans.. %100 caffeen uptake with that method.. just watch the heart rate.. |
09:01.50 | Firestrm | maybe thats why i only get about 3 hours of sleep each night ;) |
09:03.02 | Firestrm | anyhoo.. must run... TTFN.. |
09:03.25 | pranav | hello Zeeek |
09:03.33 | Zeeek | hello again pranav |
09:04.00 | pranav | i am facing a problem in the fwd |
09:04.11 | shaZwaz | why doesn't it bind to 0.0.0.0 dammn |
09:04.35 | pranav | when i call an fwd number it says"sorry invlid extensio" |
09:05.02 | murangd | Inorder for asterisk to call SIP to SIP calls |
09:05.08 | murangd | must I signup with a SIP provider |
09:05.09 | *** join/#asterisk welby (~welby@80-192-119-210.cable.ubr04.dund.blueyonder.co.uk) |
09:05.13 | murangd | and insert the user/pass in my sip.conf? |
09:05.15 | pranav | when i call to 612,or 613 it rings once and then i get no response |
09:05.21 | *** join/#asterisk Mike_TK (~Mike_TK@213.180.245.62) |
09:05.28 | murangd | pranav: do you have asterisk installed on your own server |
09:05.30 | Zeeek | murangd no you need no provider |
09:05.38 | murangd | zeek: ok cool thanks |
09:05.52 | pranav | i have pasted my sip.conf and extensions.conf in pastebin.ca/6001 |
09:05.54 | pranav | yes |
09:06.07 | Zeeek | pranav IAX or SIP ? Sometimes FWD IAX is flaky |
09:06.17 | murangd | zeeel: I |
09:06.17 | pranav | asterisk i sinstalled on my own server |
09:06.28 | pranav | sip |
09:06.33 | murangd | zeeel: I just installed asterisk on my own server but for some reason I can't make any calls |
09:06.40 | murangd | so I am trying to figure out where I went wrong |
09:06.47 | murangd | probagly I configured my .conf incorrectly |
09:07.16 | murangd | Zeeek: do you have an SIP phone number I can do a test call on and tell me if it RINGS on your side |
09:08.13 | Zeeek | try the weather station at 616306 |
09:08.24 | murangd | ok trying that one now |
09:08.25 | pranav | ok i'll try |
09:08.29 | *** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net) |
09:08.39 | murangd | pranav: is your SIp server operational? |
09:08.43 | murangd | if so can I do a test call to you |
09:08.48 | murangd | you don't have to s peak just tell me if it rings |
09:09.12 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
09:09.21 | murangd | anyone alive I can do a test SIP TO SIP call to |
09:09.46 | Zeeek | call pranav! you are made for each other |
09:10.00 | murangd | pranav: give me your SIP number |
09:10.12 | pranav | sip number? |
09:10.15 | murangd | yeah |
09:10.22 | murangd | so I can test call you |
09:10.30 | murangd | I just installed asterisk on my server and want to make sure its worknig |
09:10.39 | pranav | but there is no sip number |
09:11.00 | murangd | well any VOIP number |
09:11.20 | pranav | no |
09:11.27 | murangd | can anyone provide me with VOIP number so I may do a test call to make sure my asterisk setting is working |
09:11.48 | murangd | Zeeek: is there a way I can test asterisk without calling anyone to make sure its working? |
09:12.02 | Zeeek | call 613 the echo test |
09:12.06 | murangd | thanks |
09:12.11 | Zeeek | it's there for that very reason |
09:12.33 | murangd | Couldn't Start Call |
09:12.42 | murangd | should I put * before the number |
09:12.47 | pranav | i tried calling that 613, it rings once but then i get no response |
09:13.06 | murangd | I am getting 'Couldn't Start Call' error |
09:13.20 | pranav | murangd thats for me |
09:13.22 | *** join/#asterisk Starblazer (star@proxy.vfm.extremepcgaming.net) |
09:13.50 | Starblazer | Hello, I just noticed something weird with the Agents system, specifically AgentCallbackLogin |
09:14.13 | Starblazer | if you specify the device directly via queues.conf (EG. SIP/100), you will be able to transfer calls blindly w/o waiting for them to pick up) |
09:14.26 | Starblazer | and you get musiconhold for when you put the device on hold |
09:14.39 | Starblazer | however, if you use an extention from your extentions.conf, it will not give you the hold music |
09:14.54 | Starblazer | nor will it let you blind-transfer-before-connection |
09:15.16 | pranav | Zeeek:any guesses |
09:15.49 | Zeeek | about what? |
09:15.55 | pranav | if you can check , i have pasted my sip.conf and the extensions.conf in the pastebin.ca/6001 |
09:16.15 | pranav | i am not able to call any fwd number |
09:16.50 | Zeeek | pranav I think I do see something |
09:16.57 | Starblazer | you use IAX for fwd now |
09:17.13 | Starblazer | not SIP |
09:17.13 | Zeeek | Dial(SIP/${EXTEN:1}@fwd.pulver.com) |
09:17.26 | Zeeek | [fwd] |
09:17.30 | Zeeek | pranav |
09:17.32 | pranav | ya |
09:17.42 | Zeeek | you see above the two lines do not correspond IMO |
09:17.42 | murangd | Ok I have a question |
09:17.49 | murangd | where can I register to get a VOIP phone number |
09:18.00 | Starblazer | muranged: you're talking about a regular phone number? |
09:18.01 | *** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net) |
09:18.04 | soulz- | hello all |
09:18.06 | Zeeek | pranav change [fwd] to [fwd.pulver.com] |
09:18.07 | murangd | Starblazer: no a voip number |
09:18.22 | soulz- | does anyone know why i get this error message? |
09:18.23 | soulz- | channel.c:2173 ast_channel_make_compatible: No path to translate from IAX2 |
09:18.24 | Starblazer | what do you mean, VoIP number, those are assigned by your provider |
09:18.24 | pranav | ok i'll change it |
09:18.29 | Zeeek | or Dial(SIP/${EXTEN:1}@fwd.pulver.com) to @fwd |
09:18.46 | soulz- | http://pastebin.ca/6009 |
09:18.47 | murangd | Starblazer: I've just installed asterik on my server and I want to have a friend do a test calling to me |
09:18.55 | murangd | Starblazer: should I signup here https://www.e164.org? |
09:18.57 | Starblazer | murangd, via his land-line phone? |
09:19.06 | murangd | Starblazer: no he also has a soft phone client |
09:19.14 | murangd | Starblazer: he uses broadband |
09:19.21 | Starblazer | okay |
09:19.37 | Starblazer | if he has a softphone client, then he can just call your system at [startingcontext]@yourip |
09:19.49 | murangd | Starblazer: you mean like |
09:19.53 | murangd | userid@myip |
09:20.05 | murangd | for example |
09:20.07 | murangd | -- Registered SIP '1001' at 162.84.229.224 port 5060 expires 3600 |
09:20.12 | murangd | 1001@myip.com |
09:20.19 | soulz- | starblazer: u seemed to be the one with the answers now, can u help my problem when u get a chance? |
09:20.48 | murangd | Starblazer: btw I'm a php programmer so if you need any php scripts done.. I can help you in that department free of charge |
09:20.55 | Starblazer | soulz-, ugh. |
09:21.09 | soulz- | starblazer: thanks dude |
09:21.31 | Starblazer | murangd, example. If you were to call my asterisk server, using a softphone client, you can dial s@asterisk.myip.com |
09:21.35 | Starblazer | and my system will pick up |
09:21.45 | Starblazer | does your friend subscribe to FWD? |
09:21.51 | Starblazer | you can tie your asterisk system into FWD also |
09:21.56 | Starblazer | soulz-, lemmie check |
09:22.19 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
09:22.35 | murangd | Starblazer: I've just tried to do a test call to myself.. I've got this error |
09:22.37 | murangd | Feb 17 04:34:56 NOTICE[14107]: pbx.c:1358 pbx_extension_helper: Cannot find extension context 'from-sip' |
09:22.42 | Starblazer | ah |
09:22.51 | Starblazer | murangd, calling from where? |
09:22.57 | murangd | Starblazer: myself to myself |
09:22.59 | Starblazer | soulz-, are you trying to bridge calls? |
09:22.59 | soulz- | murangd: looks like ur context is not right |
09:23.05 | soulz- | yes i am |
09:23.17 | murangd | soulz-: can you explain a little bit more what you meant about context not being correct |
09:23.19 | Starblazer | it looks like the two systems dont have compatable phone codecs |
09:23.25 | Starblazer | murangd, in your sip.conf |
09:23.28 | Starblazer | you have a line called |
09:23.32 | Starblazer | context=[whatever] |
09:23.39 | Starblazer | under YOUR device |
09:23.46 | soulz- | starblazer: so i can't make a system talking via g729 to talk with a tdm? |
09:24.02 | Starblazer | soulz-, I dont know that much about asterisk my friend. About as much as I've played with is pure data |
09:24.15 | Starblazer | not bridging from an actual hardline-type-thing to software |
09:24.29 | soulz- | ok dude |
09:25.13 | murangd | Starblazer: can I msg you to tell you what my context settings are |
09:25.18 | pranav | zeeek:i get nothing when i dial the fwd number |
09:25.29 | Starblazer | pranav, why dont you change your fwd number to asterisk? |
09:25.32 | Starblazer | iax |
09:26.17 | pranav | but my configuration is correct then y does'nt the call go? |
09:26.21 | murangd | Starblazer: I've msg you context details |
09:26.34 | Starblazer | pranav, http://www.freeworlddialup.com/content/view/full/1501 |
09:26.41 | Starblazer | that's how I have my asterisk system bridging to FWD |
09:26.55 | pranav | but you have through iax or sip |
09:27.17 | Starblazer | iax |
09:27.23 | Starblazer | I used to have it thru sip |
09:27.40 | pranav | ok so was it working properly with sip |
09:27.48 | Starblazer | months ago |
09:27.57 | Starblazer | now I have it working properly thru IAx |
09:27.58 | Starblazer | IAX |
09:28.00 | pranav | if you can check , i have pasted my sip.conf and the extensions.conf in the pastebin.ca/6001 |
09:28.14 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
09:30.30 | *** join/#asterisk Stonekeeper (~c252e507@server0.expresshosting.net) |
09:36.44 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
09:36.56 | *** join/#asterisk _PiGreco_ (~a@adsl-215-48.38-151.net24.it) |
09:40.19 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
09:40.47 | Delvar | morning all |
09:44.42 | murangd | has anyone ever gotton this error == Spawn extension (default, s, 5) exited non-zero on 'SIP/2 |
09:44.50 | JunK-C | isnt an error. |
09:45.02 | murangd | JunK-C: I receive this message when someone tries to call me |
09:45.08 | JunK-C | <PROTECTED> |
09:45.08 | JunK-C | <PROTECTED> |
09:45.13 | murangd | oh |
09:45.17 | JunK-C | see? :) |
09:45.20 | murangd | yes |
09:45.45 | JunK-C | isnt when some1 is calling ya, its when a context reach "his end" |
09:46.04 | murangd | I had another asterisk user dial my sip number and he got this message Got SIP response 404 "Not Found" back from 65.125.228.1 |
09:46.14 | murangd | JunK-C: in which configration file do I set the user? |
09:46.33 | murangd | I've added myself in the sip.conf file and I am able to log in but for some reason when someone calls me they get 404 error |
09:46.37 | JunK-C | its cause ya dont have that extension in ur context that user is calling. |
09:46.44 | JunK-C | sip.conf |
09:46.56 | murangd | JunK-C: ok I'm in my sip.conf what should I look for |
09:46.59 | JunK-C | ya did a "reload" command after? |
09:47.06 | JunK-C | context=blah line |
09:47.29 | *** join/#asterisk ozJames79 (~james@CPE20320889-1842-1.gex.ncable.net.au) |
09:47.35 | murangd | [1001] |
09:47.35 | murangd | type=friend |
09:47.35 | murangd | username=1001 |
09:47.35 | murangd | secret=1001 |
09:47.35 | murangd | context=default |
09:47.36 | murangd | host=dynamic |
09:47.40 | murangd | allow=ulaw |
09:47.41 | murangd | JunK-C: you mean that line |
09:48.23 | murangd | JunK-C: or I am totatly off |
09:48.24 | JunK-C | in ur default context, add a line exten => 1234,1,NoOp(blah); |
09:48.29 | JunK-C | if 1234 is ur sip number |
09:48.34 | JunK-C | and do a reload |
09:48.49 | JunK-C | so 1234 would be 1001 |
09:49.05 | murangd | JunK-C: you must forgive me.. in my sip.conf file |
09:49.11 | murangd | you want me to put |
09:49.11 | ozJames79 | hi can anyone help i have setup another * box and its behind a firewall i have put it on the dmz and ported forwarded just in case everything works except on calls i have no audio codec is set to ulaw any ideas .....thanks in advance |
09:49.21 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
09:49.26 | murangd | JunK-C: context=default exten => 1234,1,NoOp(blah); |
09:49.29 | murangd | like so? |
09:49.32 | JunK-C | no |
09:49.50 | JunK-C | in ur extensions.conf, in ur default context, add exten => 1001,1,NoOp(blah); |
09:50.07 | JunK-C | ur dialplan is ur extensions.conf |
09:50.20 | JunK-C | ya should read more infos on www.voip-info.org |
09:51.11 | *** join/#asterisk Xander77 (~Alex@exten-halls-243.soton.ac.uk) |
09:51.55 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
09:53.38 | ozJames79 | hi can anyone help i have setup another * box and its behind a firewall i have put it on the dmz and ported forwarded just in case everything works except on calls i have no audio codec is set to ulaw any ideas .....thanks in advance |
09:53.56 | JunK-C | murangd: so it works? |
09:54.04 | murangd | JunK-C: one sec |
09:54.37 | murangd | ;[context] |
09:54.37 | murangd | exten => 1001,1,NoOp(blah); |
09:54.49 | murangd | no wait one sec |
09:55.33 | Starblazer | nah |
09:55.49 | Starblazer | I clued him into it, methinks |
09:56.05 | JunK-C | Starblazer: he'll be able, let him just few times. :) |
09:56.34 | Starblazer | yeah, he's taking the typical 'new person' approach' of just because it's decalred @ sip.conf, doesn't mean that it's declared everywhere yet |
09:56.46 | Starblazer | declared* |
09:56.47 | murangd | Starblazer: ok added the line |
09:56.56 | Starblazer | kks, did you restart your server? |
09:57.55 | Starblazer | Got SIP response 481 "Call Leg Does Not Exist" |
09:58.23 | murangd | Starblazer: yeah I 've restarted |
09:58.30 | murangd | it says |
09:58.38 | murangd | <PROTECTED> |
09:58.39 | murangd | <PROTECTED> |
09:59.26 | Starblazer | do you have your softphone to register with your asterisk box? |
09:59.34 | *** join/#asterisk TeLLuS (~johan@h187n2fls31o858.telia.com) |
09:59.45 | murangd | ok now for some reason |
09:59.47 | murangd | when I do |
09:59.49 | murangd | show channels |
09:59.54 | murangd | it's not showing me logged in |
10:00.01 | JunK-C | isnt in show channels |
10:00.05 | JunK-C | its sip show peers |
10:00.26 | murangd | Name/username Host Dyn Nat ACL Mask Port Status |
10:00.26 | murangd | 1001/1001 162.84.229.224 D 255.255.255.255 5060 Unmonitored |
10:00.27 | murangd | 1 sip peers [1 online , 0 offline] |
10:00.29 | murangd | it shows me connected |
10:00.46 | JunK-C | ya need 2 sip no? |
10:01.27 | murangd | JunK-C: I don't understand what you mean |
10:02.25 | JunK-C | tell us how ya making ur test exactly. |
10:02.33 | Starblazer | okay, here's the "test" |
10:02.41 | Starblazer | I've got a dial line in my asterisk server |
10:02.41 | murangd | JunK-C: I had starblazer call my sip number |
10:02.58 | Starblazer | exten => 1236,1,Dial(SIP/1001@his-ip-number) |
10:03.18 | *** join/#asterisk Stonekeeper (~c252e507@server0.expresshosting.net) |
10:03.19 | Starblazer | doing a Dial(SIP/s@his-ip) brings up the "Cool, you've got it installed" message |
10:03.41 | JunK-C | its exten => 1236,1,Dial(SIP/1001@his-HOSTNAME); no? |
10:03.43 | *** join/#asterisk slav_jb (~k@pirus.securax.be) |
10:03.46 | JunK-C | oups, IP :) |
10:03.52 | JunK-C | bit tired huh |
10:03.59 | Starblazer | you? lol |
10:04.03 | JunK-C | whatcha get when dial his 1001 ? |
10:04.07 | JunK-C | on ur side. |
10:04.09 | Starblazer | I get a 481 |
10:04.12 | Starblazer | Got SIP response 481 "Call Leg Does Not Exist" |
10:04.24 | JunK-C | murangd: ya did a reload? |
10:04.37 | JunK-C | murangd: do a show dialplan 1001@default |
10:04.38 | murangd | JunK-C: yes, before starblazer would get a 404 user does not exist |
10:05.03 | murangd | [ Context 'default' created by 'pbx_config' ] |
10:05.03 | murangd | <PROTECTED> |
10:05.03 | murangd | |
10:05.04 | murangd | -= 1 extensions (1 priorities) in 1 contexts. =- |
10:05.36 | *** join/#asterisk qwerp (~abc@219.95.105.74) |
10:05.46 | JunK-C | i told ya to put exten => 1001,1,NoOp(blah); |
10:06.03 | Starblazer | NoOp? |
10:06.23 | JunK-C | for a test |
10:06.43 | Starblazer | alright |
10:06.48 | Starblazer | I see what you mean |
10:06.56 | *** join/#asterisk bowman (~bowman@snert3.tal.de) |
10:07.04 | Starblazer | ooooo |
10:07.09 | *** join/#asterisk mak_ (~mak@privat.ua-online.net) |
10:07.11 | Starblazer | now i get a "wrong password on auth" |
10:07.11 | murangd | ok done |
10:07.16 | murangd | Starblazer: try now |
10:07.17 | mak_ | hi |
10:07.26 | murangd | Starblazer: I've just restart asterisk |
10:07.33 | JunK-C | murangd: and in ur sip.conf, in ur 1001 class, add a qualify=yes |
10:07.34 | JunK-C | and reload |
10:07.47 | Starblazer | Forbidden - wrong password on authentication for INVITE to |
10:08.08 | Starblazer | okay |
10:08.46 | murangd | ok just reloaded |
10:08.54 | JunK-C | murangd: now do a sip show peers |
10:09.13 | murangd | Name/username Host Dyn Nat ACL Mask Port Status |
10:09.13 | murangd | 1001/1001 162.84.229.224 D 255.255.255.255 5060 UNREACHABLE |
10:09.26 | JunK-C | its UNREACHABLE |
10:09.59 | murangd | how come? |
10:10.07 | Zeeek | hello again |
10:10.08 | JunK-C | dunno, its a soft-phone? |
10:10.08 | murangd | I mean.. what would cause a 'unreachable' error |
10:10.11 | JunK-C | lo Zeeek. |
10:10.14 | *** join/#asterisk ZX81 (matt@222-153-114-115.jetstream.xtra.co.nz) |
10:10.17 | murangd | JunK-C: yes a soft-phone "firefly" |
10:10.26 | JunK-C | close it, and re-open it. |
10:10.32 | *** part/#asterisk slav_jb (~k@pirus.securax.be) |
10:10.42 | Zeeek | anyone know anything about WIndows networking? I just added a LAN card so I can be connected to asterisk and the offcie LAN |
10:10.45 | JunK-C | murangd: past the last line of sip show peers too. |
10:10.59 | murangd | ok just closed |
10:11.01 | murangd | and re-opend it |
10:11.03 | Zeeek | I'm trying to figure out how to specify which connection to use for ssh and other clients |
10:11.27 | murangd | Name/username Host Dyn Nat ACL Mask Port Status |
10:11.27 | murangd | 1001/1001 162.84.229.224 D 255.255.255.255 5060 UNREACHABLE |
10:11.37 | murangd | JunK-C: still un-reachable.. what is it suppose to say? |
10:11.51 | JunK-C | its suppose to give ya a "time" |
10:12.09 | JunK-C | like: |
10:12.15 | JunK-C | gate1*CLI> sip show peers |
10:12.15 | JunK-C | Name/username Host Dyn Nat ACL Mask Port Status |
10:12.15 | JunK-C | 102/102 (Unspecified) D N 255.255.255.255 0 UNKNOWN |
10:12.15 | JunK-C | 101/101 (Unspecified) D N 255.255.255.255 0 UNKNOWN |
10:12.15 | JunK-C | 100/100 192.168.1.208 D N 255.255.255.255 5060 OK (5 ms) |
10:12.16 | JunK-C | 3 sip peers [1 online , 2 offline] |
10:12.17 | Starblazer | 100/100 209.103.209.231 D 255.255.255.255 5060 OK (88 ms) |
10:12.31 | JunK-C | Starblazer: help him to get his stuff connected. |
10:12.42 | Starblazer | I've never used his SIP client before. |
10:13.00 | Starblazer | plus it's 4am here, and I should really head off to bed |
10:13.13 | mak_ | when I putting 1.call to /var/spool/asterisk/outgoing Can I put to 1.call two Application: ? |
10:13.16 | JunK-C | Starblazer: its 5:13 here :) |
10:13.37 | JunK-C | mak: i dont think so. |
10:13.59 | modulus_ | penis |
10:14.04 | Zeeek | moose |
10:14.08 | modulus_ | oops wrong window! |
10:14.11 | JunK-C | mooo |
10:14.12 | JunK-C | :) |
10:14.13 | JunK-C | hehehe |
10:14.15 | Zeeek | who knows anything about windows here? |
10:14.24 | mak_ | JunK-C: but if I need for example SetLanguage and then SayDigit what should I do ? :) |
10:14.37 | JunK-C | Zeeek: ive just learned how to click here and there, but after all this :) |
10:14.49 | Zeeek | no, seriously |
10:14.51 | JunK-C | mak_: priorities! |
10:14.58 | JunK-C | 1,setlanguage(fr) |
10:14.59 | Zeeek | I have this great ASTERISK |
10:15.03 | JunK-C | 2,SayDigits(123); |
10:15.04 | Zeeek | on a linux box |
10:15.12 | modulus_ | Zeeek, i once did something cool on windows |
10:15.32 | Zeeek | but I have 2 LAN cards on Win box and can't figure out how to tell ssh client to use the connection to talk to asterisk |
10:15.35 | Delvar | deltree c:/windows ? |
10:15.36 | mak_ | JunK-C: tnx :) |
10:15.47 | wasim | Zeeek: route |
10:15.59 | Zeeek | wasim in a cmd line option? |
10:16.01 | modulus_ | Zeeek, that's more of a networking issue no? |
10:16.05 | wasim | Zeeek: yep |
10:16.06 | Zeeek | yes it is |
10:16.08 | Delvar | Zeeek: assuming they are on different subnets use route ad bla.... |
10:16.33 | Zeeek | errr hmmm both routers are 192.168.1.1 |
10:16.43 | Delvar | thats the problem then :) |
10:16.44 | Zeeek | is that... bad? |
10:16.49 | Delvar | that IS bad |
10:16.53 | Zeeek | shit |
10:16.58 | modulus_ | indeed |
10:17.05 | Zeeek | ah but wait |
10:17.08 | Delvar | your por windows box wouldnt have a clue where to end teh packet |
10:17.12 | JunK-C | mak_: instead of print OUTPUT "Application: Wait\n"; |
10:17.12 | JunK-C | <PROTECTED> |
10:17.17 | JunK-C | just use |
10:17.22 | *** join/#asterisk qwerp (~abc@219.93.57.58) |
10:17.23 | JunK-C | <PROTECTED> |
10:17.23 | JunK-C | <PROTECTED> |
10:17.23 | JunK-C | <PROTECTED> |
10:17.27 | JunK-C | inside ur perl script. |
10:17.38 | Zeeek | it works the other way - asterisk knows to send me stuff on 192.168.1.60 |
10:17.39 | JunK-C | with Priority |
10:18.05 | Delvar | if you have 2 nics int eh same computer they HAVE to be on different subnets or weird things happen |
10:18.17 | *** join/#asterisk murangd (~nukaidc@pool-162-83-240-155.ny5030.east.verizon.net) |
10:18.21 | murangd | sorry I got disconnected |
10:18.22 | murangd | <Starblazer> 100/100 209.103.209.231 D 255.255.255.255 5060 OK (88 ms) |
10:18.24 | murangd | *** Disconnected from IRC.freenode.net |
10:18.27 | Zeeek | shit |
10:18.28 | murangd | that was the last message I saw |
10:18.40 | Zeeek | I hate changing network stuff |
10:18.46 | Zeeek | although on linux it's easy |
10:19.00 | Delvar | i know windows can be a bitch |
10:19.06 | Zeeek | can they be one apart like 192.168.2.x ? |
10:19.17 | Zeeek | windows SUCKS! |
10:19.17 | Delvar | i still havnt found out how to perminantly modify the routing table on my windows box yet... |
10:19.32 | Delvar | yes |
10:19.40 | Zeeek | yes /8 ? |
10:19.49 | Zeeek | (or is that /24) |
10:19.54 | Delvar | one on e 192.168.1.x and the other on 192.168.2.x with netmasks of 255.255.255.0 |
10:19.56 | murangd | JunK-C: still alive |
10:20.09 | Delvar | i can never rmeber :/ |
10:20.12 | JunK-C | murangd: yes, get ur peers connected with ur sip-phone |
10:20.14 | Delvar | remember* |
10:20.14 | Zeeek | me neither |
10:20.27 | mak_ | JunK-C: why two |
10:20.28 | mak_ | 12:17 < JunK-C> #print OUTPUT "Context: default\n"; |
10:20.28 | mak_ | 12:17 < JunK-C> #print OUTPUT "Context: in\n"; |
10:20.30 | Zeeek | <PROTECTED> |
10:20.35 | Delvar | i think its a /24 |
10:21.04 | JunK-C | mak: cause i've mal-pasted |
10:21.09 | murangd | JunK-C: I am not sure WHY its says UNREACHABLE |
10:21.09 | JunK-C | <PROTECTED> |
10:21.09 | JunK-C | <PROTECTED> |
10:21.09 | JunK-C | <PROTECTED> |
10:21.23 | JunK-C | murangd: cause ur peers cant "talk" to * |
10:21.32 | mak_ | :) |
10:21.47 | Zeeek | so I change the router to be 192.168.2.1 - chenge that on linux in the rc and everything in one and 42 ? |
10:21.50 | murangd | JunK-C: ok so where would that error lie in.. is it one of the .conf files? |
10:22.09 | Zeeek | "all is one" |
10:22.11 | JunK-C | non, in ur firefly configs |
10:22.29 | JunK-C | mak_: sorry its like 5:30am here, a bit tired huh? :) |
10:23.09 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
10:23.25 | Delvar | Zeeek: so your * box is ont eh 192.168.2.x range now?... |
10:24.12 | Zeeek | long story short: got a new static ip connex, swithed the office LAN stuff to the older dynamic one - the networks were never connected to each other |
10:24.20 | Zeeek | but now I wanna talk directly to asterisk |
10:24.32 | Zeeek | instead of thru the internet to go 10 meters |
10:25.12 | murangd | can someone try to connect to my asterik server |
10:25.16 | murangd | to see if everything is working |
10:25.19 | Zeeek | I hope there's only one place I need to change the gateway in linux? |
10:26.46 | *** join/#asterisk Mike_TK (~Mike_TK@213.180.245.62) |
10:26.57 | murangd | does anyone know alterantive softphone other than firefly |
10:27.56 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
10:28.00 | *** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
10:28.05 | Delvar | bah mIRC br0ked |
10:28.17 | JunK-C | murangd: x-lite |
10:28.57 | Delvar | Zeeek: what did you say last? |
10:30.01 | JunK-C | look at this: |
10:30.02 | JunK-C | http://pastebin.ca/6019 |
10:30.05 | JunK-C | really strange. |
10:40.18 | *** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au) |
10:41.26 | *** join/#asterisk meppl (~mephisto@62.158.37.97) |
10:51.39 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
10:52.06 | *** join/#asterisk [ro]nic3try (~nic3try@p3.pub.ro) |
10:52.08 | cjk | hi, was anyone of you able to get a price quotation from pulver.com for their pulver communicator software? |
10:52.11 | [ro]nic3try | re all |
10:52.49 | [ro]nic3try | has anyone know how to set the default codec wich asterisk use ? |
10:53.47 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
10:54.07 | oej | Codec for which protocol? |
10:54.15 | fishboy1669 | morning guys |
10:54.21 | [ro]nic3try | sip |
10:54.53 | CMike | hiyas oej |
10:54.53 | [ro]nic3try | defaul is g711 |
10:55.00 | JunK-C | [ro]nic3try: its ulaw by default. |
10:55.03 | JunK-C | right |
10:55.10 | JunK-C | g711u or a |
10:55.28 | [ro]nic3try | and i want to use g729 |
10:56.04 | PakiPenguin | [ro]nic3try: buy it |
10:56.44 | [ro]nic3try | ??? |
10:57.44 | JunK-C | RO: ya need to pay for that codec |
10:58.20 | PakiPenguin | [ro]nic3try: if you dont want to pay , use gsm! |
10:59.16 | [ro]nic3try | but if i give : show codecs |
10:59.22 | [ro]nic3try | i see i have g729 |
10:59.42 | [ro]nic3try | how do i set it to be default on all cals to asterisk |
11:01.29 | PakiPenguin | show codecs , just give a detail about codecs |
11:02.54 | fishboy1669 | has anyone ever tried this |
11:02.55 | fishboy1669 | v |
11:02.56 | fishboy1669 | http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20forwarding |
11:03.57 | fishboy1669 | is anyone awake here?! |
11:03.58 | fishboy1669 | lol |
11:04.12 | Delvar | no |
11:04.32 | Delvar | looks interesting why not give it a go? |
11:04.47 | [ro]nic3try | ok, so .. just to understand.. asterisk does not suport g729? or i cann't use it as default codec |
11:05.06 | Delvar | asterisk does support g729 you just need to buy licances |
11:05.17 | fishboy1669 | i am trying to get it working |
11:05.29 | *** join/#asterisk cc (~cc@byte.fedora) |
11:05.43 | Delvar | to set as default codec for all calles in your [general] context add disallow=all allow=g729 |
11:05.44 | fishboy1669 | the cli shows stuff happning but when i dial the extention i have put the frowaring on it still dials |
11:06.18 | [ro]nic3try | that i do in sip.conf ? |
11:06.48 | *** join/#asterisk Jnel (~bit-logic@c2-239-1.pta.dial.mweb.co.za) |
11:08.14 | Jnel | hi all; I need help with a sip phone config |
11:08.36 | Jnel | I am trying to get 2 S3020's talking, but.. |
11:08.57 | Jnel | the phones ring and disconnect on answer.. |
11:09.30 | Jnel | The error message is that the frame type requested is type 1 and the native type is 4.. |
11:10.08 | Jnel | Further i says that there is no patch to convert from type 1 to 4.. |
11:10.20 | Jnel | any solutions or suggestions ?? |
11:11.32 | *** join/#asterisk HjemmeRoyK (~roy@83.80-203-29.nextgentel.com) |
11:12.43 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
11:12.49 | Jnel | Any ideas on this problem?? |
11:13.02 | Zeeek | human stupidity has no limits - especially my own ! |
11:13.39 | Jnel | lol.. tell me about it..I feel the same way |
11:14.44 | Zeeek | So I changed the gateway address on one router and couldn't ping it. It took all of 20 min torealize that changing the gateway wasn't enough, I had to change the ip itself. INCREDIBLY STUPID! |
11:15.36 | *** join/#asterisk usam (~usam@203.156.37.115) |
11:15.52 | Jnel | mmmm...sounds like you're in a mood to ponder things...try and help me with this.. |
11:15.55 | usam | is it possible to change a codec on the fly when using SIP ? |
11:15.57 | cjk | which stund do you recommend, the one from vovida or the mystun? |
11:16.26 | Jnel | I've got 2 S3020's which disconnects on anser... |
11:16.30 | Zeeek | stun.e164.org |
11:16.43 | Zeeek | member of the asterisk community - one of US ;) |
11:16.44 | HjemmeRoyK | when is it you'll really need stun? |
11:17.11 | Zeeek | RoyK Hej! just the man for my windows network question |
11:17.13 | murangd | can someone explain to me what is DID |
11:17.15 | HjemmeRoyK | we're running SIP behind all sorts of NAT..... |
11:17.21 | HjemmeRoyK | Zeeek: hehe |
11:17.22 | murangd | in relations to phone numbers |
11:17.23 | Jnel | The error says that the frame type requested is 1 and not 4 as the native... |
11:17.34 | Zeeek | DID is a number people can call to ring into your * server |
11:17.34 | murangd | is DID a kind of phone number? |
11:17.42 | Zeeek | Direct Inward Dial |
11:17.43 | murangd | zeek: ok cool thanks |
11:17.49 | murangd | ah ok I see |
11:17.55 | cjk | Zeeek, you might be right that that stun is great, but i want a stun server where i have the control. if it goes down i know home to blame |
11:18.01 | Zeeek | Roy you know how to tell a particular program to use a certain route? |
11:18.03 | Jnel | setting is... and no patch to convert... any solution?? |
11:18.12 | murangd | Zeeek: do you know where I can get some cheap DID numbers.. like a provider that provides DID numbers? |
11:18.14 | HjemmeRoyK | Zeeek: as in ip route? |
11:18.18 | Zeeek | cjk ok, I don't use stun anyway |
11:18.38 | Zeeek | RoyK yeah, I have a second card now and want to talk ssh directly to the asterisk network box |
11:19.15 | Zeeek | and maybe even connect an IAXY or SIP phone some day (router has a couple of extra slots) |
11:19.17 | murangd | Zeeek: why are you buying hardware cards to support asterisk? your making outgoing calls? |
11:19.39 | Zeeek | murangd nothoing to do with that - but I do have three Digium cards |
11:19.51 | murangd | Zeeek: what exactly are the purpose for Digium cards |
11:19.59 | murangd | Zeeek: they are used ONLY for PSTN dialing right? |
11:20.13 | Zeeek | mine are to connect three regular phones (FXS) and to connect two phone lines (FXO) |
11:20.13 | *** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au) |
11:20.26 | Zeeek | yes you don't n,eed hardware murangd |
11:20.26 | murangd | ah I see |
11:20.41 | murangd | well Junk-C help me setup my asterik server |
11:20.50 | murangd | and now I can receive VOIP to VOIP calls |
11:21.01 | murangd | no I want to do VOIP to PSTN calls |
11:21.09 | murangd | so I need to get a VOIP termination provider |
11:21.15 | murangd | but I don't really know any good ones |
11:21.25 | murangd | also I am looking for a place that provides DID numbers |
11:21.36 | Zeeek | yes so why not go looking for one? nufone, voicepulse, iconnecthere, voiptalk, voipjet |
11:22.10 | Zeeek | just add .com to any of the above and stir |
11:22.20 | Jnel | hi all; I need help with a sip phone config |
11:22.29 | Jnel | I am trying to get 2 S3020's talking, but.. |
11:22.37 | Jnel | the phones ring and disconnect on answer.. |
11:22.45 | Jnel | The error message is that the frame type requested is type 1 and the native type is 4.. |
11:22.55 | Jnel | any solutions or suggestions ?? |
11:23.07 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
11:23.18 | Zeeek | sounds like the codec curse |
11:23.33 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) |
11:23.36 | Jnel | meaning? |
11:23.52 | Zeeek | meaning look at your codec definitions |
11:24.06 | libpcp | would it be possible setup ? if (!($row[10] > 6)) { ?> |
11:24.06 | libpcp | <PROTECTED> |
11:24.06 | libpcp | <?php |
11:24.06 | libpcp | } |
11:24.11 | libpcp | ops sorry |
11:24.25 | libpcp | SIP server ---> E1 Channel Bank ---- PSTN |
11:24.47 | *** join/#asterisk MuppetMaster (~muppetmas@a82-92-73-185.adsl.xs4all.nl) |
11:25.05 | libpcp | sorry guys, i thought i was not on my clipboard |
11:28.26 | Zeeek | s'ok RoyK I found it and now everything is cool as in 42 ! |
11:30.55 | *** join/#asterisk TheEmperor (TheEmperor@218.111.48.89) |
11:31.03 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
11:32.49 | Zeeek | who was it said something about not being able to make persistant routes? I found that too |
11:33.03 | HjemmeRoyK | route -p add |
11:33.05 | HjemmeRoyK | iirc |
11:33.31 | Zeeek | yep - and I just looked, it adds the route to the regisrty. But why didn't you answer me whan I was looking ? :) |
11:34.07 | HjemmeRoyK | trying to do some work as well as helping you :) |
11:34.20 | Zeeek | I'll accept that this time |
11:34.27 | Mavvie | what is an persistant route? |
11:34.28 | HjemmeRoyK | "this time" :) |
11:34.30 | Zeeek | ~lart RoyK |
11:34.45 | HjemmeRoyK | Mavvie: it's saved over a reboot |
11:34.52 | Zeeek | persistant means it stays between boot |
11:34.59 | Zeeek | ~lart RoyK even more |
11:35.05 | jetscreamer | one tha tpersists over reboots |
11:35.06 | Mavvie | HjemmeRoyK: oh. euhm. windows? |
11:35.07 | jetscreamer | iirc |
11:35.11 | HjemmeRoyK | Mavvie: yeah |
11:35.17 | Zeeek | ~lart RoyK leaving nothing but a smalll greasy spot |
11:35.20 | HjemmeRoyK | Mavvie: I used to be an MCSE on 3.51 and 4.0 :P |
11:35.23 | Mavvie | aha. no miracle I wasn't familiar with the term. |
11:35.25 | HjemmeRoyK | ~kill Zeeek |
11:35.27 | jbot | ACTION shoots a inverse meson gun at Zeeek |
11:35.28 | HjemmeRoyK | ~LART Zeeek |
11:35.30 | jetscreamer | route -a or something in windows i think |
11:35.41 | HjemmeRoyK | jetscreamer: route -p, as stated above |
11:35.45 | jetscreamer | o |
11:35.46 | HjemmeRoyK | ~lart jetscreamer |
11:35.47 | jetscreamer | nm |
11:35.49 | jetscreamer | :/ |
11:35.52 | fishboy1669 | anyone know of a advance extentions.conf dial plan documnetation i need to understand it propper now |
11:35.54 | Zeeek | you have to understand that most of the world's offices run WIndows |
11:36.04 | Zeeek | so a minimum network knowledge is nec |
11:36.09 | fishboy1669 | doing fancy stuff like call redirect config from phone |
11:36.13 | HjemmeRoyK | ~docs |
11:36.14 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
11:36.14 | fishboy1669 | hi zeek |
11:36.14 | Zeeek | but now, I rule the world with two LAN cards |
11:36.27 | Mavvie | Zeeek: don't worry about my network knowledge :-P |
11:36.28 | Zeeek | ~lart docs |
11:36.53 | Zeeek | I'm not worried - only my own is worrying |
11:36.54 | fishboy1669 | wish jbot could be more specific |
11:36.59 | Zeeek | Starter tutorial: |
11:36.59 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
11:36.59 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
11:36.59 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
11:36.59 | Zeeek | THE reference of the moment: |
11:36.59 | Zeeek | http://www.asteriskdocs.org |
11:37.05 | fishboy1669 | ~docs extentions.conf |
11:37.14 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. |
11:37.14 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
11:37.16 | Zeeek | that one |
11:37.19 | fishboy1669 | aha last one is new |
11:38.00 | Zeeek | so the problem of CallerID notification on the Windoze boxes is solved |
11:38.08 | fishboy1669 | cheers zeek |
11:38.13 | fishboy1669 | ~dov |
11:38.18 | Zeeek | until the next ip change :( |
11:38.21 | fishboy1669 | ~docks |
11:38.25 | fishboy1669 | ~docs |
11:38.26 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
11:38.31 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. |
11:38.31 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
11:39.11 | fishboy1669 | dont i know it lol |
11:42.41 | *** join/#asterisk Igor-BZ- (~root@62.123.121.61) |
11:43.31 | *** join/#asterisk Luhiwu (~marsosa@200.63.89.209) |
11:45.26 | *** join/#asterisk pranav (sameer@202.149.48.200) |
11:45.27 | Igor-BZ- | hi :) |
11:45.34 | pranav | hi |
11:46.24 | murangd | is this site voipjet.net |
11:46.26 | murangd | loading for anyone? |
11:46.39 | *** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl) |
11:46.45 | HjemmeRoyK | murangd: works for me *tm( |
11:46.51 | HjemmeRoyK | murangd: works for me (tm) |
11:46.56 | jerlique | does anyone have any experience with agent logouts? |
11:47.25 | HjemmeRoyK | murangd: er. no |
11:47.30 | HjemmeRoyK | murangd: it doesn't |
11:48.03 | pranav | i am not able to make calls with fwd , i can make calls to pstn, mobile, internlly |
11:48.10 | libpcp | from what ive experience, voipjet.net has a noise on the voice quality |
11:48.40 | pranav | when i make calls to fwd numbers it says "sorry its an invalid extension |
11:49.13 | *** join/#asterisk trym (~trym@linux.debian.us) |
11:49.26 | pranav | i have pasted my sip.conf and the extensions.conf in the pastebin.ca/6001 |
11:49.52 | pranav | can anyone tell me what is the mistake |
11:50.22 | trym | I have installed spandsp to have asterisk receive faxes. When a fax call is made to asterisk, asterisk starts whining about RFC3389. I also notice that the volume spandsp/asterisk is communicating with varies.. which is not normal for a fax session. Any suggestions? |
11:50.25 | murangd | what's the best PSTN termination provider in terms of COST and Quaility |
11:50.32 | HjemmeRoyK | pranav: change your passwords :} |
11:51.01 | murangd | what's the best PSTN termination provider in terms of COST and Quaility |
11:51.14 | pranav | but then i have registered with that password to that number |
11:51.31 | pranav | u mean to say shold i register again |
11:51.41 | trym | HjemmeRoyK: have you used spandsp with asterisk ? |
11:51.49 | jerlique | Can asterisk do a desktop->fax gateway? |
11:51.49 | HjemmeRoyK | nope |
11:52.00 | HjemmeRoyK | jerlique: should work, on a LAN |
11:52.12 | HjemmeRoyK | otherwice you'll need t.38, which isn't finished |
11:52.14 | Igor-BZ- | I'm trying www.wipphone.com I have try with my BGT |
11:52.15 | jerlique | are there some docs? |
11:52.27 | HjemmeRoyK | jerlique: see spandsp |
11:52.31 | jerlique | thanks |
11:52.47 | HjemmeRoyK | ~spandsp |
11:52.54 | HjemmeRoyK | ~fax |
11:52.55 | jbot | Well, apperantly the fax was concieved of by Napoleon Bonaparte. He commissioned a system of devices that could transmit a traced image electrically over telegraph lines to a remote device that would redraw the image identically. |
11:53.48 | jerlique | hehehe |
11:54.03 | Makenshi | the first fax was an interesting mechanical contraption |
11:54.21 | Makenshi | making use of a swiging arm and a counterbalance |
11:54.24 | Igor-BZ- | When asterisk call pvt->read in a channel? channel il ANSWERED, and asterik call pvt->write rigth... |
11:54.40 | pranav | when i make calls to fwd numbers it says "sorry its an invalid extension |
11:55.12 | *** join/#asterisk [Hug] (~ss@195.244.154.200) |
11:56.05 | jerlique | Are ppls here integrators of asterisk systems, or users within their business? |
11:58.23 | *** join/#asterisk hajekd (~hajekd@mail.idoox.com) |
11:59.57 | hajekd | Whats up to VoipJet? Their DNS is screwed. |
12:00.33 | pranav | can someone tell me what is the mistake |
12:01.46 | *** join/#asterisk Weezey (Weezey@lan6.LO.iasl.com) |
12:02.01 | Weezey | if I'm connected to asterisk CLI, how do I exit without bringing down the server? |
12:03.04 | tzafrir | pranav, is that your reall password in the file? |
12:03.21 | tzafrir | or have you bothred modifiying it? |
12:03.34 | JunK-C | Weezey: exit |
12:03.55 | murangd | does anyone know any Public VOIP Gateway |
12:03.57 | tzafrir | OK, already noted |
12:04.19 | Weezey | JunK-C: hmm, that must not have worked before because I wasn't using -r to connect to it. thanks. |
12:04.34 | JunK-C | if ya've started it with -c |
12:04.40 | JunK-C | there's no way i know |
12:04.47 | JunK-C | stop now, then safe_asterisk |
12:04.52 | JunK-C | and asterisk -rv |
12:04.59 | JunK-C | then exit gonna works |
12:05.23 | pranav | yes thats what i have registered with |
12:05.26 | Weezey | okay, thanks. |
12:05.40 | pranav | no i have never modified it |
12:06.28 | pranav | is there a problem of NAT |
12:06.32 | bowman | thanks for giving out your password. |
12:07.44 | pranav | bowman:are u telling me |
12:07.52 | Weezey | I have two SPA-3000s, one connected to POST and the other connected to a Norstar ATA, for some reason the norstar ATA just keeps ringing when there's an incoming call and it's configured the same way. Outgoing calls throught the norstar ATA work fine however. |
12:08.16 | Weezey | s/POST/POTS/ |
12:09.13 | *** join/#asterisk Specky[W] (~sspecken-@p508EC9F0.dip0.t-ipconnect.de) |
12:09.37 | *** join/#asterisk GodThor (~ninja@212.110.95.139) |
12:10.25 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
12:10.28 | pranav | tzafrir: is that a mistake of NAT |
12:10.47 | *** part/#asterisk Specky[W] (~sspecken-@p508EC9F0.dip0.t-ipconnect.de) |
12:10.55 | GodThor | when i start asterisk there is not h323 protocols, i have installed pwlib, openh323 |
12:11.12 | GodThor | do i must install something else? |
12:11.51 | tzafrir | pranav, NAT will generally cause issues with the RTP data channels and not with the SIP control channel |
12:12.13 | tzafrir | IIUC |
12:12.22 | pranav | ok fine |
12:12.46 | tzafrir | do you send them the correct number? |
12:13.11 | pranav | yes |
12:13.24 | miller7 | anyone here familiar with zapras and dial in access to * box? |
12:13.38 | tzafrir | How have you verified that? |
12:16.22 | GodThor | anyone help with h323? |
12:17.48 | pranav | see i know a few numbers like 613,612,55555 so tried calling them |
12:18.08 | *** join/#asterisk Tornad (~regis@81.56.183.143) |
12:18.22 | Tornad | hi |
12:18.27 | pranav | when i call them it rings once or twice but then i get no further response |
12:19.13 | Weezey | GodThor: don't quote me or anything, but I think you have to change the asterisk source to have the h323 code from the openh232 site, then re-compile. |
12:19.24 | pranav | but when i call to somebody's fwd number it says "invalid extension" |
12:20.36 | Igor-BZ- | there is a channel... che chan_h323 or chan_oh323 is a wrapper from openh323 and asterisk... |
12:20.54 | *** part/#asterisk [ro]nic3try (~nic3try@p3.pub.ro) |
12:21.37 | Igor-BZ- | U can found all here: http://www.inaccessnetworks.com/projects/asterisk-oh323 |
12:22.32 | Igor-BZ- | use pwlib and openh323 version ONLY FROM inaccess-network... |
12:22.52 | Igor-BZ- | there is howto on this site ^^^^^^ |
12:23.39 | GodThor | because i didnt install them manually it comes with asterisk package /from cvs |
12:24.58 | GodThor | ok i would try inaccess-network, thanks |
12:27.42 | Igor-BZ- | ...mmm.... I'm using this driver... :) |
12:29.45 | *** join/#asterisk Banter (Banter@209.119.214.81) |
12:30.19 | GodThor | thanks Igor, thats my real name also |
12:30.40 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
12:30.44 | pranav | see when i call a few fwd numbers like 613,612,55555 so tried calling them |
12:30.52 | pranav | but when i call to somebody's fwd number it says "invalid extension" |
12:31.02 | pranav | see when i call a few fwd numbers like 613,612,55555 so tried calling them |
12:31.16 | pranav | when i call them it rings once or twice but then i get no further response |
12:31.50 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
12:32.47 | pranav | hi igor |
12:34.59 | Igor-BZ- | hi pranav :) |
12:35.22 | *** join/#asterisk mrempire (~user1@h71032.upc-h.chello.nl) |
12:35.26 | Igor-BZ- | do U have a correct configuration on extensions.conf? |
12:36.25 | pranav | tell me what to so |
12:37.04 | pranav | i have pasted my sip.conf and extensions.conf in the pastebin.ca/6001 |
12:38.24 | pranav | and the calls are going to pstn, mobile and internally only to a fwd number they are not going |
12:38.38 | pranav | do you want to see what comes on the cli screen when i dial the fwd number |
12:40.42 | *** part/#asterisk pranav (sameer@202.149.48.200) |
12:40.56 | *** join/#asterisk pranav (sameer@202.149.48.200) |
12:41.20 | pranav | sorry i got disconnected |
12:41.30 | pranav | now i am back |
12:42.27 | Igor-BZ- | ok I'm looking... |
12:42.38 | pranav | ya |
12:45.34 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
12:47.13 | Igor-BZ- | there is a problem on your conf.... |
12:47.20 | pranav | ya tell me |
12:47.23 | Igor-BZ- | use this "template" http://www.voip-info.org/wiki-Asterisk+FWD+NAT+Config+Example |
12:48.42 | pranav | ok i'll go through this site |
12:49.25 | pranav | but is there a mistake of NAT |
12:49.50 | Igor-BZ- | why? |
12:50.08 | Igor-BZ- | can U open port on your router? |
12:50.51 | pranav | bcos someone yesterday gave me a command like "traceroute_ _ |
12:51.32 | pranav | it was a address after this and he told that your asterisk server is behind a NAT |
12:52.35 | murangd | Igor-BZ-: have you tried out voipuser.com ? |
12:52.48 | Igor-BZ- | never... |
12:53.29 | pranav | see the wan is connected to the router and from router to the switch and switch to the asterisk pc |
12:53.49 | Zeeek | what is the NAT problem this time? |
12:54.15 | pranav | i don't know whether it is a NAT problem |
12:54.22 | *** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
12:54.28 | Zeeek | well, are we still on the same FWD stuff after all these hours? |
12:54.36 | pranav | ya |
12:54.51 | pranav | the calls are not going |
12:54.52 | Zeeek | what is the problem -I've been disconected a while) |
12:55.21 | Zeeek | pranav look at this |
12:55.25 | Zeeek | http://willypick.mindsay.com/?entry=10 |
12:55.35 | pranav | see i am not able to make fwd calls |
12:55.35 | Zeeek | ^^^^^^^ The asterisk config that dare not speak its name: Double NAT! ^^^^^^^^^^^^ |
12:55.43 | *** join/#asterisk amer (~aaa@203.99.60.27) |
12:56.26 | amer | my setup is *sip ----- *IAX------*IAX------*SIP |
12:56.34 | Zeeek | aha |
12:56.43 | Zeeek | I've never had a problem |
12:56.44 | amer | my setup is A----*sip ----- *IAX------*IAX------*SIP-----B |
12:56.48 | pranav | ya i'll gothrough this |
12:56.50 | *** join/#asterisk mcukstorm (~mcukstorm@neo.matrix-lan.net) |
12:56.51 | Zeeek | what is your router setup on both ends |
12:57.03 | amer | when I call from A to B, caller ID is messed up |
12:57.15 | amer | I am unable to figure out whats going on |
12:57.28 | Zeeek | aha |
12:57.34 | mcukstorm | hi all, does any one know a pinout i can use for connecting an RJ45 from the 400P card (FXO module) to a BT PSTN Line (UK) |
12:57.41 | Zeeek | and you are using setcallerid etc in extensions to FWD |
12:57.52 | amer | nope |
12:58.10 | amer | i dont want to change the callerID |
12:58.16 | Zeeek | amer speaking for the 12455 people here? |
12:58.25 | *** join/#asterisk didz_ (didz_@200.218.192.52) |
12:58.27 | Zeeek | who is amer? |
12:58.32 | Zeeek | the other pranav? |
12:58.41 | amer | no |
12:58.44 | amer | amer is amer |
12:58.53 | Zeeek | oh got confused with pranav |
12:59.02 | amer | np |
12:59.13 | Zeeek | so why is callerid messed? |
12:59.15 | Zeeek | amer |
12:59.16 | amer | so do u see any problems with my setup? |
12:59.17 | pranav | what happened? |
12:59.26 | amer | i just see 00000000 |
12:59.30 | Zeeek | amer I use that all the time |
12:59.43 | amer | instead of the caller ID of the SIP phone A |
12:59.51 | Zeeek | what does arterisk CLI see ? |
13:00.11 | Zeeek | insert a NoOp(${CALLERIDNUM}) in the dialplan |
13:00.19 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
13:00.27 | amer | hmmmm |
13:00.42 | Zeeek | to see what is coming in from your phone, ya see? |
13:00.43 | amer | I see the correct caller ID |
13:00.56 | amer | ok let me try that |
13:01.02 | Zeeek | which is like Hey Now <2000> |
13:01.05 | Zeeek | ?? |
13:05.17 | *** join/#asterisk h3x (~Justino@adsl-065-013-150-019.sip.msy.bellsouth.net) |
13:05.31 | h3x | hi |
13:05.42 | *** join/#asterisk meshugga (philip@loeblich.linuxteam.at) |
13:05.56 | h3x | guess who was at a convention i just went to |
13:06.00 | h3x | sysmaster |
13:06.01 | meshugga | hi |
13:06.02 | h3x | hahahahaha |
13:06.13 | h3x | i so wanted to give them a bunch of crap |
13:06.18 | Zeeek | shiksa? |
13:06.24 | h3x | "hi id like to buy an asterisk box from you for $50k" |
13:08.21 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
13:09.17 | Weezey | can I play musiconhold in the background while transferring a call? |
13:12.33 | Zeeek | it should do that automagically |
13:12.54 | h3x | if the moh is set up right anyway |
13:12.55 | h3x | heh |
13:12.55 | Zeeek | call yourself and see! |
13:22.48 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
13:28.03 | *** join/#asterisk sangee (~rkuru@207.188.77.82) |
13:28.56 | murangd | Zeeek: I've just setup an account at FWD and it registeries correctly |
13:29.09 | murangd | Zeeek: but I am unable to make a call.. I am using X-lite softphone software |
13:29.24 | murangd | Zeeek: do I have to dial 394 then the number inorder to make a call? |
13:30.03 | Zeeek | no you dial the number if registered directly to FWD |
13:30.16 | Zeeek | if not you need a dialplan extension |
13:30.20 | netsurfer | http://www.theregister.co.uk/2005/02/17/spam_gets_vocal_with_voip/ <-- ffs that takes the piss |
13:30.21 | Weezey | nope, no ringing, no moh, not nuthin'. Hrmmm... On a completely new topcic, how come sometimes I can press buttons when the biatch is talkin' but sometimes it just ignores them? |
13:30.22 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. |
13:30.22 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
13:30.44 | murangd | Zeeek: in my extenstion.conf I have this |
13:30.44 | murangd | exten => _394.,1,SetCallerId,voip@goldenbucks.biz |
13:30.44 | murangd | exten => _394.,2,Dial(IAX2/13569@fwdOUT/${EXTEN:3},60,r) |
13:30.47 | murangd | exten => _394.,3,Congestion |
13:30.49 | sangee | does asterisk supportx g723 and g729 codec? |
13:30.49 | murangd | sorry I mean |
13:31.25 | murangd | Zeeek: is that what you are refering to when you say DIALPLAN? |
13:31.25 | Zeeek | Setcalleridnum(YOURFWDNUM) |
13:31.26 | *** join/#asterisk _Brian (brian@unix01.voicenet.com) |
13:31.33 | Zeeek | and set the name to whatever |
13:32.00 | Zeeek | The number one answer is FWD site has this info that is what they did that page for |
13:32.22 | Zeeek | Intyerested in FWD? FreeWorlDialup? |
13:32.23 | Zeeek | http://www.freeworlddialup.com/content/view/sitemap/2 |
13:32.28 | Zeeek | This is their site map |
13:32.39 | Zeeek | http://www.freeworlddialup.com/support/configuration_guide |
13:32.47 | Zeeek | This is the CONFIGURATION GUIDE |
13:33.19 | _Brian | tired of people asking Zeeek? |
13:33.22 | Zeeek | Hereis all you need to know about FWD IAX: |
13:33.22 | Zeeek | http://www.freeworlddialup.com/advanced/iax |
13:33.32 | Zeeek | no, do I appear tired? :) |
13:33.57 | bjohnson | sangee: I think g723 (is that gsm?) .. g729 is supported if you buy licenses (I think $10 each) |
13:33.59 | _Brian | just a little bit :) |
13:34.14 | murangd | http://www.fwdout.net = freeworlddialup.com? |
13:34.18 | sangee | where can i buy the license? |
13:34.20 | bjohnson | ~docs |
13:34.21 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
13:34.23 | jetscreamer | i could ask what is freeworldialup? :o |
13:34.31 | murangd | Zeeek: http://www.fwdout.net = freeworlddialup.com? |
13:34.35 | jetscreamer | j/k |
13:34.45 | bjohnson | sangee: from digium or whoever owns g729 |
13:34.58 | Zeeek | So my new macro, let's see: |
13:35.00 | Zeeek | http://www.freeworlddialup.com/content/view/sitemap/2 |
13:35.00 | Zeeek | http://www.freeworlddialup.com/support/configuration_guide |
13:35.00 | Zeeek | http://www.freeworlddialup.com/advanced/iax |
13:35.00 | Zeeek | http://www.freeworlddialup.com/support/forum |
13:35.06 | sangee | thx |
13:35.15 | HjemmeRoyK | http://hampage.hu/pdp-11/kepek/1103sys.jpg |
13:35.23 | bjohnson | murangd: all sorts of info on the wiki listed by ~docs |
13:35.27 | Zeeek | fwdout is Bellster after the lawsuit threat |
13:35.42 | bjohnson | I thought fwd was included in the samples |
13:35.43 | *** join/#asterisk CpuID (~none@CPE-203-45-152-22.qld.bigpond.net.au) |
13:35.47 | Zeeek | new asterisk box RoyK ? |
13:36.03 | Zeeek | it is but now they do IAX and all, why not go to the source |
13:36.32 | HjemmeRoyK | Zeeek: hehe. with an 18bit cpu |
13:36.38 | Zeeek | http://www.freeworlddialup.com/advanced/service_numbers |
13:36.45 | HjemmeRoyK | PDP-11 |
13:37.12 | Zeeek | I used to program PDP-11 in RSX and RT11 |
13:37.17 | Zeeek | under, not in |
13:37.28 | Zeeek | used assembler and some Fortran |
13:37.33 | felipex | i have 2 * box connected via iax2 trunk |
13:37.39 | Zeeek | so far so good |
13:38.10 | felipex | when i try to place a call from *1 to *2 i have this messages |
13:38.33 | felipex | Call rejected by 192.168.255.23: No authority found |
13:38.45 | felipex | this message in *1 |
13:38.59 | jetscreamer | i had star trek on papertape |
13:39.21 | felipex | in *2 i have Rejected connected attempt from 192.168.0.5 |
13:41.52 | *** join/#asterisk mjmac (~mjmac@cpe-68-175-244-78.maine.res.rr.com) |
13:42.15 | bjohnson | felipex: make sure the username and secrets are the same on both boxes .. use the host arg if you have any that are static ip addresses (then you don't need to register on that machine) |
13:45.06 | *** join/#asterisk TheEmperor (TheEmperor@218.111.50.241) |
13:50.19 | ariel_ | morning all |
13:50.44 | oej | Morning |
13:50.58 | oej | ls -la |
13:51.19 | oej | Oops, wrong window |
13:51.35 | *** join/#asterisk zeedo (~notroot@www.bsrf.org.uk) |
13:52.45 | amer | :) |
13:52.56 | Moc | oej hey what up ? |
13:54.19 | Zeeek | madonna naked |
13:54.23 | Zeeek | oops wrong window :) |
13:54.37 | amer | -- Executing NoOp("IAX2/edge@edge/4", "19790709709777") in new stack |
13:55.02 | tzanger | werd to the goatherd |
13:55.29 | amer | callerid is correct but when its passed to a the SIP server from usr is chaged to asterisk |
13:55.53 | amer | who can I make asterisk not to do this and pass on the actual callerID |
13:56.00 | murangd | amer: you have your caller id working correctly? |
13:56.14 | murangd | amer: when I dial a local number, it says 'Unknown Number' |
13:56.51 | amer | Zeeek any ideas |
13:57.04 | oej | Stable or head? |
13:57.14 | oej | Callerid is broken in head (personal opinion) |
13:57.27 | junky[work] | oej: since when? |
13:57.38 | ariel_ | actually callerID is also strange on stable. |
13:57.53 | amer | CVS-v1-0-02/10/05-15:59:23 |
13:58.08 | ariel_ | I get it fine on the NoOp line but when it's sent to the phones like it shows the name but unknown number. |
13:58.45 | junky[work] | amer: isnt head. |
13:59.00 | amer | no but my problem is different |
13:59.06 | ariel_ | I am going to upgrade to the stable release as of today. I have had it for over 1 week and read on the cvs that he removed the new id to the older stuff that was working. |
13:59.18 | amer | my setup is A----*sip ----- *IAX------*IAX------*SIP-----B |
13:59.40 | amer | my setup is A----*sip ----- *IAX1------*IAX2------*SIP-----B |
14:00.20 | ariel_ | amer, your still using iax1? |
14:00.40 | bjohnson | we'll use the candlestick in the library. Don't tell Colonel Mustard |
14:00.42 | amer | IAX2 received the ccorrect callerID but when it passes the call to *SIP it puts from user as asterisk@10.0.9.1 |
14:00.44 | bjohnson | oops wrong window |
14:01.07 | amer | no thats just to differentiate b/w 2 servers |
14:01.17 | ariel_ | bjohnson, so funny "NOT" |
14:01.37 | bjohnson | now you're just being mean |
14:02.44 | `Sauron | YEah |
14:02.45 | ariel_ | bjohnson, really now.... I was trying to be funny. |
14:02.49 | `Sauron | quit being mean to my buddy... |
14:02.53 | `Sauron | ;) |
14:03.19 | amer | hey guys what about my problem |
14:03.36 | amer | oej, you know whats wrong here |
14:04.50 | Essobi | Hmm. I got like 5 sip peers I use to dial out. Anyone have a suggestion how to fail them over to each other with outbound dialing? |
14:04.52 | ariel_ | amer, I don't use head but it's a problem as well in my stable |
14:05.51 | murangd | amer: how does your extenstion.conf look |
14:05.56 | murangd | amer: mine is exten => _394.,1,SetCallerId,voip |
14:05.59 | murangd | amer: what's yours? |
14:06.45 | amer | I dont want to set any callerID, I just want asterisk to pass whatever callerID it gets |
14:06.55 | *** join/#asterisk Ubuz (~momo@DSL217-132-49-219.bb.netvision.net.il) |
14:07.26 | greendisease | Ubuz: manyanim |
14:08.02 | Ubuz | greendisease: hakol tov |
14:08.16 | amer | english please |
14:08.27 | Ubuz | ok, hello everybody |
14:09.10 | Ubuz | I have a question about playing non gsm files with agi. I managed to record a vox file, but how can i play it? |
14:09.24 | Ubuz | Or any other kind of file. |
14:09.27 | junky[work] | Ubuz: STREAM ? |
14:09.32 | junky[work] | STREAM FILE |
14:09.44 | Ubuz | STREAM FILE ignores the file |
14:10.00 | junky[work] | huh? |
14:10.38 | mtqh | convert to gsm them stream |
14:11.18 | Ubuz | Why should I convert? If it can record, surely it can play. |
14:11.31 | mtqh | did you do a |vox afterward |
14:11.36 | mtqh | are you in head or stable? |
14:11.39 | greendisease | hey can someone donate a conference room for some sessions at linuxworld today |
14:11.43 | mtqh | brb |
14:12.32 | Ubuz | I didn't do a |vox after, because I don't know after what I should do it. |
14:12.38 | Weezey | All my incoming calls to my set are coming in with the proper callerid, but with asterisk as the number, no the number they're coming from. SETCIDNUM is set correctly, how do I pass that to the phone? |
14:12.51 | Weezey | err CALLERIDNUM |
14:13.18 | `Sauron | <PROTECTED> |
14:13.21 | greendisease | JerJer: ping |
14:13.31 | ariel_ | greendisease, I think bkw_ said he would last night. |
14:14.13 | greendisease | he did but hes not here now, and we need to get it set up soon |
14:14.20 | TheEmperor | what's the best softphone to use? iax2 and messenging |
14:15.32 | ariel_ | TheEmperor, I use xlite for sip and diax for iax in windows. |
14:16.05 | TheEmperor | ariel_ : where can i get diax from? |
14:17.18 | ariel_ | TheEmperor, http://www.laser.com/dante/ |
14:18.00 | TheEmperor | ariel: thanks :) |
14:18.58 | murangd | ariel_: why do you alterante? |
14:19.09 | murangd | I mean couldn't you just do SIP -- IAX via your asterisk setup |
14:19.14 | murangd | no need to use two different softphones |
14:19.36 | TheEmperor | would it be better to use iax2 rather than sip on the softphone? |
14:19.36 | fishboy1669 | can anyone shead light on what this does |
14:19.40 | fishboy1669 | exten=s,2,Dial(Local/${temp}@pbx/n) |
14:19.53 | murangd | does anyone have their called ID working successfully? |
14:20.16 | Zeeek | everyone murangd |
14:20.22 | fishboy1669 | lol |
14:20.24 | murangd | Zeeek: well not I |
14:20.25 | murangd | lol |
14:20.36 | murangd | can you paste me an example of your settings in extenstion.conf |
14:20.42 | murangd | where you have your caller id proberly setup |
14:20.43 | Zeeek | callerid from where? a SIP caller to your asterisk? |
14:20.47 | fishboy1669 | exten=s,2,Dial(Local/${temp}@pbx/n) |
14:20.49 | fishboy1669 | ? |
14:20.53 | ariel_ | murangd, I use xlite or most of the setups. I use diax due to sometimes I run into a system that blocks my rtp stream. |
14:21.16 | murangd | Zeeek: Yes a sip caller to asterisk to fWD to PSTN number |
14:21.43 | Zeeek | and where are you looking to see the cid? |
14:21.55 | Zeeek | on what phone? |
14:21.55 | murangd | Zeeek: on local PSTN phone caller id's menu |
14:21.59 | TheEmperor | how's xten's eyeBeam? |
14:22.05 | amer | can I set fromuser in Extensions.conf file |
14:22.07 | murangd | Zeeek: is that possible or no? |
14:22.20 | Zeeek | FWD to PSTN? How are you doing that? |
14:22.50 | murangd | woah |
14:22.55 | murangd | I'm just a newbee |
14:23.05 | murangd | Zeeek: how do you dial PSTN numbers? |
14:23.16 | bjohnson | greendisease: fwd? |
14:23.18 | Zeeek | from FWD? You don't usually |
14:23.31 | greendisease | bjohnson: huh? |
14:23.33 | Zeeek | except at certain holiday promo times |
14:23.38 | bjohnson | fwd has conf rooms |
14:23.43 | murangd | ok my orginal question |
14:23.48 | greendisease | whats fwd? |
14:23.52 | Zeeek | http://www.freeworlddialup.com/content/view/sitemap/2 |
14:23.52 | Zeeek | http://www.freeworlddialup.com/support/configuration_guide |
14:23.52 | Zeeek | http://www.freeworlddialup.com/advanced/iax |
14:23.52 | Zeeek | http://www.freeworlddialup.com/support/forum |
14:23.52 | Zeeek | http://www.freeworlddialup.com/advanced/service_numbers |
14:23.53 | greendisease | or whose fwd? |
14:23.57 | greendisease | ah fwd |
14:24.00 | greendisease | that fwd |
14:24.39 | Zeeek | murangd I still don't see how you are calling a PSTN line from FWD |
14:24.43 | murangd | exten => _8.,1,SetCallerId,voip |
14:24.45 | murangd | is that correct |
14:24.46 | murangd | or no |
14:24.47 | Zeeek | so I don't know about your prob |
14:25.02 | murangd | Zeeek: well answer this exten => _8.,1,SetCallerId,voip <-- is that correct for caller id setup? |
14:25.05 | Zeeek | show application setcallerid |
14:25.11 | bjohnson | greendisease: I haven't used the fwd conf rooms .. but they say they are available |
14:25.36 | *** join/#asterisk Tornad (~regis@81.56.183.143) |
14:25.36 | Zeeek | murangd you need to read the docs to see how comannds work - try to wiki there is a complete list |
14:25.41 | bjohnson | murangd: you're trying to set outgoing callerid right? |
14:26.04 | bjohnson | murangd: look at the superdial macro on the wiki .. it does a bunch of things you will want |
14:26.12 | Zeeek | to talk to fwd you can set callerid to your FWD number |
14:27.06 | GodThor | i have problem to compile pwlib from inaccess ,any other solution? |
14:27.31 | `Sauron | YEah, don't use h.323? :) |
14:27.54 | GodThor | :))))))))) |
14:27.55 | murangd | bjohnson: cool thanks |
14:28.21 | GodThor | for quintum any other protocols? |
14:28.49 | murangd | bjohnson: do you happen to have the URL for wiki |
14:29.06 | Zeeek | you mlight want to read this too |
14:29.07 | Zeeek | http://www.voip-info.org/wiki-Asterisk+cmd+SetCIDNum |
14:29.10 | *** join/#asterisk HitTop (~Jack@host6614613596.biz.tor.fcibroadband.com) |
14:29.44 | GodThor | to change my panasonic 1232 , what you propose protocols in asterisk, i user 10 analog lines |
14:29.53 | murangd | Zeeek: thanks |
14:31.24 | *** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com) |
14:36.04 | amer | can I set "fromuser" in Extensions.conf file |
14:37.06 | bjohnson | what's fromuser? |
14:37.36 | amer | or better set fromuser in sip.conf to fromuser=${callerIDnum} |
14:37.56 | amer | sip FromUser |
14:38.13 | Delvar | ~time |
14:38.15 | jbot | methinks time is 1 dimensional, or everlasting |
14:38.51 | Delvar | what is jbot anyway? |
14:39.44 | tclark | GodThor: check the cost of 10 analog plus inet data vs fractional t1/data to get get rid of the 10 analog lines then just use asterisk with a t1 interface |
14:39.48 | Delvar | an imp with too much time on his hands? |
14:40.52 | bjohnson | ~jbot |
14:40.53 | jbot | i heard jbot is the shipboard computer, but you may call me eddie if it helps you relax |
14:41.02 | bjohnson | hi eddie |
14:41.34 | Delvar | i see |
14:42.08 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
14:43.28 | *** join/#asterisk bill522 (~bill522@182-30.201-68.swfla.rr.com) |
14:43.31 | *** join/#asterisk Syrus_ (~pascal@tahiti.mpl.rullier.net) |
14:43.56 | murangd | bjohnson: ok I've just read the entire history of caller id |
14:44.09 | murangd | they should really provide working example of how to implement this |
14:45.02 | bjohnson | did you look at the superdial macro? |
14:46.40 | shaZwaz | implicit u there ? |
14:46.58 | murangd | bjohnson: reading it now |
14:47.40 | bjohnson | one caveat .. setting name usually doesn't work unless you go out through your own PRI (and it supports it) |
14:48.02 | bjohnson | often you can set CIDNUM if going out through a voip provider |
14:48.06 | shaZwaz | anyone has successfully used outbound providers and local LAN phones while binding on 0.0.0.0 ? |
14:48.42 | bjohnson | ? |
14:48.59 | bjohnson | binding what on 0.0.0.0? |
14:49.05 | Ubuz | anyone knows what happend to xvoip? are there any other forums for asterisk? |
14:49.06 | shaZwaz | SIP |
14:49.37 | bjohnson | well what do you know .. I guess I bind to 0.0.0.0 |
14:50.03 | bjohnson | I connect to LAN but I think all my internet connections use IAX |
14:50.04 | shaZwaz | bjohnson: the prob is that if I use 0.0.0.0 my outbound services dont register me |
14:50.41 | shaZwaz | well sure it works fine on IAX but not on SIP |
14:50.54 | greendisease | does anyone use kphone? |
14:51.02 | shaZwaz | 0.0.0.0 listens to all |
14:51.45 | bjohnson | greendisease: better try some alternatives .. you're running out of time |
14:52.06 | bjohnson | btw .. most of the people here will tend to connect to FWD from * .. not softphone clients |
14:52.38 | shaZwaz | that what I am doing |
14:53.05 | bjohnson | oops .. forgot one thing when you decided to try iaxcomm .. you need to sign up for a iax account |
14:53.17 | shaZwaz | but I dont understand why doesn't it register me when listening on 0.0.0.0 |
14:53.23 | ^Fenris | I have a POTS line and a VOIP line coming into my * box, in the [default] section of extensions.conf, how do I differentiate between them? (they need to be handled differently) |
14:54.26 | murangd | bjohnson: I've read the page on super macros but I don't think that will help me |
14:54.38 | bjohnson | greendisease: follow answer 4 here http://www.freeworlddialup.com/content/view/full/1501 |
14:54.40 | tzanger | ^Fenris: first off, please don't use [default] |
14:54.45 | bjohnson | murangd: works for me |
14:54.56 | tzanger | and make your [default] section nothing more than exten => s,1,Hangup |
14:55.04 | ^Fenris | tzanger: alright |
14:55.06 | tzanger | it saves you from unintended operation |
14:55.18 | tzanger | then have your POTS line come into its own [POTS] or [fxo] context |
14:55.25 | bjohnson | ^Fenris: next .. get rid of that symbal from the front of your nick |
14:55.31 | tzanger | and yrou VOIP line into whatever the provider name's next |
14:55.54 | murangd | bjohnson: can you paste me a line in your extenstion.conf where you have caller id probaberly setup and visible on a PSTN phone |
14:56.13 | bjohnson | I use the superdial macro |
14:56.22 | bjohnson | that is why I pointed you to it |
14:56.23 | ^Fenris | bjohnson: heh, someone else has the decarrotized ver of my nick already registered on this network |
14:56.48 | shaZwaz | any SIP guru around ? |
14:57.15 | ^Fenris | tzanger: okay, I'll work on that, thanks |
14:57.17 | bjohnson | the "context" lines in zapata.conf, sip.conf, iax.conf set where incoming calls go to in extensions.conf |
14:57.30 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
14:57.50 | PBXtech | why am i getting these messages == Primary D-Channel on span 2 up PRI DEBUG shows nothing |
14:58.11 | tzanger | ^Fenris: if you want them both to do the same thing hten just create a common context and include it from those other specific contextsx |
14:58.29 | tzanger | it's far far far safer than throwing everything and everyone in [default] since you are CONSCIOUSLY doing it |
14:58.41 | tzanger | PBXtech: are you getting it over and over and over? |
14:58.46 | PBXtech | yes |
14:58.47 | tzanger | it sounds like your D channel is bouncing |
14:58.49 | bjohnson | ^Fenris: an overview is available on the wiki pages about extensions.conf |
14:58.51 | tzanger | you won't find anything in the debug |
14:59.03 | PBXtech | yea, that because of SLIP? |
14:59.08 | tzanger | becuase the PRI isn't actually sending anything since hte D channels' up and down like a bride's nightie |
14:59.29 | murangd | anyone have caller ID working |
14:59.32 | murangd | on their system |
15:00.05 | bjohnson | yes. I do |
15:00.05 | PBXtech | is there any way to test the prob? |
15:00.15 | bjohnson | both for incoming AND outgoing |
15:00.22 | murangd | heh |
15:00.23 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
15:00.33 | murangd | yes I know bjohnson but what you've suggested I don't understand how to implement |
15:00.53 | bjohnson | what is the url for the superdial macro? |
15:01.02 | murangd | http://www.voip-info.org/tiki-index.php?page=Superdial%20macro |
15:01.19 | bjohnson | so you can successfully dial out correct? |
15:01.27 | murangd | correct |
15:01.28 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
15:01.28 | amer | is there an application like setFromUser? |
15:01.45 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
15:01.51 | bjohnson | murangd: give me a one of your working outgoing dial command lines |
15:02.09 | miller7 | anyone here familiar with zapras and pppd? I need some minor help. I have compiled, installed and tested it and I need help with pppd settings |
15:02.24 | *** join/#asterisk cc (~cc@byte.fedora) |
15:02.35 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
15:02.53 | bjohnson | amer: what is that? |
15:02.54 | PBXtech | [tzanger]: is there any way to trouble shoot the D channel? |
15:03.37 | amer | I want to change the "from field" in SIP header to the callerID |
15:04.30 | amer | so what I will do is _X.,1,setFromUser=${callerIDNum} |
15:04.37 | murangd | bjohnson: check msg |
15:04.57 | amer | this way actual callerID will be displayed on the called phone |
15:06.10 | bjohnson | amer: look at the superdial macro |
15:06.13 | bjohnson | on the wiki |
15:06.16 | amer | ok |
15:06.19 | bjohnson | ~docs |
15:06.20 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
15:06.33 | bjohnson | it uses SetCIDNum() |
15:08.18 | *** join/#asterisk yurpls (~yurplsl@65.114.15.70) |
15:08.39 | yurpls | Heloo |
15:10.28 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
15:11.20 | murangd | is macro support enabled on asterik by default? |
15:11.31 | NormAst | PBXTech: What kind of problems you having with your D-Channel |
15:11.34 | sivana | yes |
15:11.49 | *** join/#asterisk eKo1 (~bernd@207.42.191.66) |
15:12.56 | yurpls | Anyone know what it means when I attach a trunk (analog) to a FXO port on digium card and it stays busy when connected? |
15:13.41 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfndi.dialup.mindspring.com) |
15:14.41 | *** join/#asterisk [Latre] (~latre@148.233.19.133) |
15:14.43 | *** part/#asterisk GodThor (~ninja@212.110.95.139) |
15:14.49 | *** part/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4118784.sympatico.ca) |
15:16.25 | bjohnson | reasons why a person might want more than one voip provider account: 1. auto failover if one is not available (same reason why you often would like to failover to a pstn line if voip is not available) 2. even if not auto failover, having an existing account with another provider means you can manually change your outgoing quickly if you need to for some reason 3. provider1 has DID you want but the provider you use for outgoing does not 3. |
15:16.49 | tzanger | I made my own superdial macro based on Manxpower's excelent macro |
15:17.20 | dsmouse | bjohnson: you were cut off at 3 |
15:18.35 | bjohnson | that was a typo |
15:18.55 | bjohnson | 4. you temporarily get a special deal or freebie |
15:19.11 | vaewyn | 5. You like to screw around with your setup more than you should |
15:19.16 | vaewyn | ;P |
15:19.24 | bjohnson | tzanger: I loosely based mine on his .. but posted it on the wiki |
15:19.33 | tzanger | *nod* |
15:19.36 | murangd | bjohnson: do you know any good DID providers |
15:19.39 | dsmouse | 6. you like testing out other providers to make sure your service is as good as you think it is |
15:19.42 | bjohnson | murangd: yes |
15:19.45 | bjohnson | hundreds |
15:19.56 | murangd | bjohnson: could you paste a few |
15:20.08 | bjohnson | what country .. to what country? |
15:20.22 | murangd | USA&Canda to Usa&Canda |
15:20.30 | bjohnson | outgoing only? |
15:20.39 | tzanger | the only problem wiht IAX is that I can receive CONGESTION or CHANUNAVAIL |
15:20.41 | murangd | outgoing/incoming |
15:20.48 | tzanger | I believe I should only receive CONGESTION if the far end TELLS me it's congested |
15:20.50 | bjohnson | livevoip, voipjet, teliax, iax.cc, nufone |
15:20.53 | *** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
15:20.57 | bjohnson | aleph-com.net |
15:20.57 | tzanger | if it's too busy to take my call I should get CHANUNAVAIL |
15:20.59 | murangd | thanks |
15:21.24 | murangd | bjohnson: a DID is a number where someone can call you and that number can be hooked up to your VOIP service or am I mistaken? |
15:21.25 | bjohnson | murangd: incoming DIDs to Canada are hard to find. |
15:21.32 | fishboy1669 | is there anyone that can tell me what the oej is in this exten=7001,1,Macro(stdexten,7001,SIP/oej) |
15:21.33 | bjohnson | that's right |
15:21.38 | fishboy1669 | its doing my head in |
15:21.54 | fishboy1669 | what is SIP/oej ???????? |
15:21.55 | dsmouse | fishboy1669: it should match a context in sip.conf |
15:21.55 | bjohnson | fishboy1669: it's an arg to that macro |
15:22.21 | tzanger | bjohnson: you got that right |
15:22.25 | fishboy1669 | ok i understand the arg to to the macro |
15:22.31 | tzanger | iax.cc has 'em through Group but as you've seen on the list their service is spotty |
15:22.40 | bjohnson | fishboy1669: if it's the stdexten macro I copied form an example, the number is the voicemail box and the SIP/oej goes to a SIP device |
15:22.43 | fishboy1669 | but what is SIP/oej to do |
15:22.56 | *** part/#asterisk Banter (Banter@209.119.214.81) |
15:23.02 | fishboy1669 | yes its stdeexten macro |
15:23.11 | bjohnson | SIP/oej goes to a SIP device |
15:23.18 | dsmouse | fishboy1669: it'll get passed to dial later, which will look up oej in sip.conf |
15:23.31 | fishboy1669 | oh |
15:23.32 | fishboy1669 | mmm |
15:23.34 | fishboy1669 | ey |
15:23.37 | bjohnson | ioej must be a SIP device configured in sip.conf |
15:23.39 | fishboy1669 | confused |
15:23.56 | fishboy1669 | the sip.conf is not mentioned in the example |
15:24.03 | fishboy1669 | no wonder i couldnt figure it |
15:24.04 | bjohnson | make your own |
15:24.10 | fishboy1669 | yup |
15:24.12 | fishboy1669 | cheers guys |
15:24.39 | bjohnson | likely that arg is used in a dial command |
15:24.53 | *** join/#asterisk brettnem (~brettnem@208.54.232.29) |
15:25.05 | yurpls | Anyone have the pinouts on the 400P card with FXO? Is it 4&5 or 3&4? |
15:25.07 | dsmouse | fishboy1669: well, it could be replaced by anything, eg "Zap/1" for line one off some zaptel thing, or "IAX2/guest@66.250.68.194/996" to get the conf today at 2 |
15:25.14 | fishboy1669 | u stars |
15:25.41 | fishboy1669 | i changed it to a current sip phone extention and it seems to be worknig now |
15:25.46 | fishboy1669 | thanks again guys |
15:26.21 | riksta | has anyone got a system, where their agents can log into like an "after call work" or "lunch" area, and hop back into the queue? |
15:27.35 | HitTop | hi all |
15:27.53 | HitTop | i wonder if there's any Ser user arround? |
15:29.58 | fishboy1669 | im just starting with it |
15:30.40 | HitTop | fishboy1669: do u use mysql with ser? |
15:30.52 | ariel_ | yurpls, are you asking about the plugs in the TDM400B card it uses the 4&5 for the active pair |
15:31.48 | fishboy1669 | i have it installed with that but not used them together yet |
15:31.50 | fishboy1669 | sorry |
15:31.56 | fishboy1669 | i am using mysql with * |
15:32.13 | *** join/#asterisk TrevorSHarrison (~trevorsha@24.49.36.218) |
15:34.52 | HitTop | oic.. ~ |
15:38.24 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
15:38.24 | *** mode/#asterisk [+o anthm] by ChanServ |
15:40.57 | *** join/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net) |
15:41.04 | *** join/#asterisk slav_jb (~k@pirus.securax.be) |
15:41.43 | *** part/#asterisk slav_jb (~k@pirus.securax.be) |
15:49.22 | *** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc) |
15:50.53 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-18-115-202.buff.east.verizon.net) |
15:51.45 | SuPrSluG | hello |
15:53.35 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
15:57.39 | greendisease | anyone know how to create a conference room with fwd? |
15:57.50 | murangd | can someone explain to me what is Radius protocol |
15:58.03 | greendisease | remote authentication dial in user service |
15:58.15 | murangd | greendisease: what is it use in relation to VOIP |
15:58.34 | murangd | authentication method? |
15:58.44 | greendisease | most likely |
16:00.07 | murangd | how can I encoperate radius with my asterik server for user authentication |
16:00.45 | *** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net) |
16:01.12 | ManxPower | murangd, You do a lot of hacking. |
16:02.22 | *** part/#asterisk mcukstorm (~mcukstorm@neo.matrix-lan.net) |
16:06.14 | *** join/#asterisk HitTop (~Jack@host6614613596.biz.tor.fcibroadband.com) |
16:06.36 | bprice20 | is anyone else having trouble with extensions.conf mysql db using realtime? |
16:07.17 | *** join/#asterisk brazil (~cleber@200.198.105.37) |
16:07.53 | brazil | hello all |
16:09.17 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
16:09.36 | jsolares | is it me, or is iaxtel really spotty |
16:10.25 | `Sauron | iaxtel blows |
16:10.41 | jsolares | oh ok, i was beginning to think i was screwing up |
16:13.12 | hajekd | how you handle outgoing faxes with asterisk? Getting a card with analog ports? |
16:13.26 | ManxPower | hajekd, yes. |
16:13.32 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
16:14.06 | hajekd | ManxPower - any recommendation on good card? |
16:14.56 | ManxPower | You mean like the TDM400P with FXS and FXO interfaces? |
16:16.30 | hajekd | if that one is working with fax then yes ;) |
16:16.49 | PatrickDK | heh, fax is a different story |
16:17.04 | greendisease | hmm, kphone doesnt have a number pad |
16:17.08 | PatrickDK | they all work with fax, but you need software to understand fax |
16:17.45 | hajekd | I can receive fax with asterisk. |
16:18.03 | PTG123 | hey is their a way to know in extensions.conf which sip account is dialing, and use the first character in it in the dial string? |
16:18.06 | hajekd | But now what to connect a fax machine to an analog port of asterisk... |
16:18.11 | hajekd | s/what/want |
16:18.33 | *** join/#asterisk Uajal (~icechat5@ool-182e86f3.dyn.optonline.net) |
16:20.36 | ManxPower | hajekd, Then install an analog port in Asterisk |
16:20.44 | yurpls | Anyone have a TDM400P and a couple of minutes? |
16:21.33 | Uajal | I read documentation. Still I didn't understand idea of extensions. I have asterisk installed. And now I configure it for broadvoice. There in register command I should set extension. Have no Idea. I have SIP phone. I now its IP address. Can be this sip phone be mapped to extension? |
16:22.05 | *** part/#asterisk djin (~marius@62.58.40.196) |
16:24.27 | *** join/#asterisk klicTel (~Claude@207.107.208.137) |
16:24.34 | klicTel | hi all |
16:25.44 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
16:25.53 | HitTop | i want to ask about sliming asterisk |
16:25.58 | PBXtech | what is the zttest tool used for |
16:26.17 | ManxPower | PBXtech, checking interrupt latency |
16:26.25 | ManxPower | well at least for zaptel cards |
16:26.48 | *** join/#asterisk clint_ (~clint@snap.helixsystems.com) |
16:26.55 | HitTop | if i only use sip and iax chan, can i remove all other chan*.so? |
16:26.59 | klicTel | is anyone aware of issues connecting Cisco call manager to * using SIP? |
16:27.12 | PBXtech | so has nothing to do with the bandwith, speed of the card? to help trouble shoot echo? |
16:27.35 | clint_ | Does anyone here understand the difference between busy and congestion in asterisk? |
16:27.43 | HitTop | if i only use ulaw, can i rm all other codec_*.so? |
16:28.17 | HitTop | because right now, codec_ilbc.so and codec_lpc10.so seems to be running very slowly |
16:29.11 | PatrickDK | ilbc is always slow |
16:29.19 | HitTop | wat is it for? |
16:29.21 | PatrickDK | you can rm any you want |
16:29.29 | PatrickDK | ilbc has error correction and stuff in it |
16:29.35 | PatrickDK | it goes slow, but handles packet loss |
16:29.59 | HitTop | but if i've set every channel to use just one codec, asterisk will bridge calls for me right? |
16:30.07 | brazil | can you help me about QOS? I using an appliance that having HTB and SFQ with Asterisk.. My question is if HTB and SFQ can used to QoS with Asterisk? |
16:31.33 | HitTop | (because right now, im trying to install a sip router to a linksys, but SER is just too complicated for me.. i'd just stick with asterisk, and try to slim it) |
16:31.55 | PatrickDK | if you just want bridging, ser would be better |
16:32.33 | *** join/#asterisk easydone (~notdone@eksel.demon.nl) |
16:32.41 | HitTop | i guess i'd have to bridge for calls, because the performance for the router isn't high~ |
16:32.43 | JerJer | bridging and ser |
16:32.46 | JerJer | mmkay |
16:32.54 | JerJer | don't you mean proxy? |
16:33.17 | PatrickDK | I think he just want to let one phone call another phone |
16:33.21 | PatrickDK | nothing else interesting |
16:33.28 | HitTop | right patrickdk |
16:33.37 | brazil | anybody had an answer? |
16:33.47 | fishboy1669 | hi |
16:33.54 | fishboy1669 | anyone know what im doing wrong |
16:34.03 | JerJer | so a SIP proxy |
16:34.25 | fishboy1669 | i have setup a main menu with exten => s,background,welcome |
16:34.28 | fishboy1669 | but it dont work |
16:34.36 | JerJer | u need a priority in there |
16:34.40 | *** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk) |
16:34.47 | fishboy1669 | sorry thats in there |
16:34.53 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
16:34.54 | fishboy1669 | s,1,back .... |
16:35.03 | *** join/#asterisk plappy (~asdf@64.56.147.94) |
16:35.06 | fishboy1669 | its getting something to point at it thats the issue |
16:35.27 | fishboy1669 | i have sip ext 117 context=main-menu but it dont do anything |
16:35.30 | Weezey | can someone telnet to 206.210.111.28 port 4800 |
16:36.17 | Weezey | I just need to know if it connects (not refused) |
16:36.24 | vagwin | it connects |
16:36.27 | HitTop | it connects |
16:36.40 | fishboy1669 | IT connects people |
16:36.43 | HitTop | Connected to flowers.loit.ca (206.210.111.28). |
16:36.50 | Weezey | cool, thanks. |
16:37.03 | vagwin | wow. ddostastic. |
16:37.14 | vagwin | :P |
16:37.31 | *** join/#asterisk stepcut (~redlion@ip68-107-21-88.sd.sd.cox.net) |
16:37.32 | fishboy1669 | jerjer any idea how i set something up to point at my exten => s,1,back ..... |
16:38.07 | fishboy1669 | when i dial 117 it is still looking for 117 context in the extentions |
16:38.10 | fishboy1669 | .conf |
16:39.09 | stepcutHM | I am trying to decide whether to get a toll-free number from iax.cc or teliax, any one want to share their experiences? |
16:39.12 | HitTop | JerJer: do u use SER? Im reading SER howto, but I think the documentation teaches base on environment of ser with mysql, but I just have SER by itself. Could you guild me to any website or guildline just to start SER without mysql? |
16:39.33 | Weezey | fish: exten => t,1,Goto(s,1) ? |
16:40.07 | fishboy1669 | weez whats that do? |
16:40.16 | fishboy1669 | t ? |
16:41.12 | fishboy1669 | i have exten => t,1,goto,0|1 at the end of the main menu context |
16:41.49 | HitTop | fish: u want sip client to dial 117 to go into ur welcome context? |
16:41.54 | fishboy1669 | yes |
16:42.06 | fishboy1669 | i dial 117 on sip phone |
16:42.09 | bjohnson | for everyone asking about how to send a call from one to another or answer a call .. this is for you |
16:42.10 | bjohnson | ~docs |
16:42.11 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
16:42.11 | HitTop | fish: within ur client context, add this line |
16:42.41 | HitTop | fish: exten => 117,1,Goto(welcomeContext,s,1) |
16:43.02 | fishboy1669 | aha ok |
16:43.11 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
16:43.12 | fishboy1669 | mmm |
16:43.27 | bjohnson | stepcutHM: my experience is that both can be tested with an investment of < $20 .. sign up to or both and test them. Make notes on the wiki so the next person doesn't have to go through the same |
16:44.13 | HitTop | fish: read the asterisk tips in wiki, it helps a lot to begin with asterisk~!! |
16:44.13 | stepcutHM | bjohnson: yeah, I was thinking of using iax.cc for toll-free (no monthly fee), and teliax from a local DID (half the cost of iax) :) |
16:44.16 | HitTop | ^_^ |
16:44.17 | fishboy1669 | so whats the exten => s,1, ........ used for |
16:44.29 | HitTop | fish: s stands for start i think |
16:44.43 | fishboy1669 | yes |
16:45.00 | fishboy1669 | so if its start then how do i get an incommming call to point to it |
16:45.02 | bjohnson | stepcutHM: no problem with that .. I currently have accounts with 4 voip providers due to that type of thing |
16:45.09 | shido6 | where is the call coming from fishboy1669? |
16:45.11 | HitTop | is there any SER user arround? I need some help~_~ |
16:45.16 | fishboy1669 | sip phone |
16:45.25 | bjohnson | reasons why a person might want more than one voip provider account: 1. auto failover if one is not available (same reason why you often would like to failover to a pstn line if voip is not available) 2. even if not auto failover, having an existing account with another provider means you can manually change your outgoing quickly if you need to for some reason 3. provider1 has DID you want but the provider you use for outgoing does not 4. you t |
16:45.25 | bjohnson | emporarily get a special deal or freebie |
16:45.38 | bjohnson | someone should put that on the wiki |
16:45.47 | shido6 | do you want EVERY sip user to contact your box and get this or just a certain group? or only from a specific endpoint? |
16:45.52 | bjohnson | I'll stop pasting it to the chan .. I know it's annoying |
16:46.21 | fishboy1669 | i got the code out of my * book but it doesnt say how to point all incomming calls to it |
16:46.35 | fishboy1669 | at moment just from one phone |
16:46.38 | oej | HitTop: There is a #ser channel |
16:46.46 | HitTop | there's no one there in that channel |
16:47.13 | shido6 | fishboy1669 yes, but who do u want incoming calls to come from? one specific sip endpoint, a group or all? |
16:47.14 | HitTop | i wonder if there's that little ppl using ser compare to asterisk? |
16:47.22 | vaewyn | hmm... anyone from abptech on here? |
16:47.23 | bjohnson | fishboy1669: what interface do you have incoming on? find it in the conf files and change the context to point to where you want incoming calls to go. Put your 's' exten lines there |
16:47.37 | SuPrSluG | i have dundi working. but am having nat issues. on one end it changes the port. any way to force a cable/dsl router to not change the prot when port forwarding |
16:47.41 | fishboy1669 | one specific sip endpoint |
16:47.43 | fishboy1669 | for testing |
16:47.46 | oej | HitTop: So what's your question? |
16:48.11 | shido6 | then in your sip.conf in the user stanza put the context with your "s" extensions there so for example |
16:48.20 | shido6 | check ur pm |
16:49.15 | fishboy1669 | i have pointed that sip extention context at the mainmenu but the extentions.conf dont play with it |
16:49.18 | fishboy1669 | jsut cuts it off |
16:49.28 | shido6 | ok |
16:49.34 | shido6 | pastebin.ca your sip.conf and extensions.conf |
16:52.41 | HitTop | oej: i want to know if there's any beginner guide for ser that doesn't require mysql. I want to set ser as a proxy, so it routes internal calls and external to asterisk server |
16:53.42 | fishboy1669 | if i change the s to 117 then it works |
16:53.50 | fishboy1669 | god this is frustrating |
16:53.54 | HitTop | oej: and I got using serctl, when i do serctl restart, it gives this error: "Stopping SER : No PID file found! SER problably not running" |
16:53.58 | __Sparks_ | If I have multiple SIP routes, for example three for Sipgate, and one for Free World Dialup, do I need to specify different ports somwhere in sip.conf? |
16:54.04 | clint_ | Busy vs Congestion, anyone? |
16:54.31 | vaewyn | fast vsslow :} |
16:54.34 | clint_ | How does asterisk decide whether a call gets busy or congestion? |
16:54.36 | vaewyn | vs slow even |
16:54.50 | oej | hittop: There's a good admin guide on iptel.org/ser |
16:55.05 | HjemmeRoyK | clint_: the dial plan |
16:55.05 | vaewyn | busy == busy... congestion == deep shy0t is hapenning with the phone system |
16:55.06 | ManxPower | ~docs |
16:55.08 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
16:55.18 | clint_ | Okay, got that part. |
16:55.31 | clint_ | But how is it that when dialing a phone directly connected to asterisk box... |
16:55.48 | clint_ | (asterisk should know that the station is busy, not inexplicably unavailable...) |
16:55.54 | clint_ | that congestion is the normal response? |
16:56.08 | clint_ | ('m sure this is an I.O. error, but I'm stumped) |
16:56.51 | shido6 | clint_, what? |
16:57.03 | shido6 | some phones report a busy |
16:57.07 | shido6 | when in use |
16:57.09 | shido6 | or DND |
16:57.19 | shido6 | are u talking about PRI's? |
16:57.32 | clint_ | shido6: config: two phones on channel bank, on asterisk. |
16:57.36 | HitTop | ls |
16:57.45 | shido6 | through a T1 interface on asterisk |
16:57.46 | shido6 | ? |
16:58.02 | clint_ | shido6: take #1 offhook, dial it from #2, we get congestion by default, not busy. |
16:58.11 | shido6 | or is asterisk running on your channel bank? |
16:58.20 | shido6 | congestion? |
16:58.21 | shido6 | ok |
16:58.24 | shido6 | whats in ur dialplan |
16:58.26 | clint_ | shido6: asterisk box -> channel bank -> phones |
16:58.28 | shido6 | what does the CLI error say |
16:58.28 | shido6 | ? |
16:58.36 | shido6 | pastebin.ca your extensions.conf |
16:58.41 | shido6 | and we'll help ya out |
16:59.03 | *** join/#asterisk Christopher1 (KRS1@68-233-58-6.atlsfl.adelphia.net) |
16:59.10 | clint_ | exten -> 1,1,Dial(Zap/1), exten-> 1,2,Busy(); likewise for exten2. |
16:59.15 | __Sparks_ | Hello! - Do i need multiple port = lines for each SIP account in my sip.conf file? |
16:59.15 | Christopher1 | Hello. |
17:00.01 | clint_ | (standby) |
17:00.27 | shido6 | err |
17:00.33 | shido6 | noooo |
17:01.37 | Zeeek | so wasim... and so it goes |
17:01.40 | SuPrSluG | anyone else have problems w/ asterisk.xvoip.com . i get a blank page. no errors. |
17:02.28 | *** join/#asterisk zoa (zoa@82.103.76.147) |
17:04.26 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
17:05.50 | Christopher1 | what digium hardware would you recommend to use asterisk with 4 analog phone lines? |
17:06.18 | Zeeek | TDM400 with 4FXO |
17:06.22 | PatrickDK | only 4, I would say tdm400 |
17:06.39 | PatrickDK | though, I do like sipura 2000 for that too myself |
17:06.42 | Christopher1 | thanx |
17:07.01 | Christopher1 | asterisk is l33t |
17:10.14 | hajekd | Is there are EU store for digium cards? |
17:10.50 | loud | country ? |
17:10.55 | hajekd | Czech Republic |
17:10.57 | loud | i know there's one in france |
17:11.03 | zoa | in belgium too |
17:11.08 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
17:11.20 | mutilator | mpg321 won't work instead will it? |
17:11.24 | hajekd | URLs? |
17:11.29 | loud | what a beautiful country Czech Republic, want the french l ink ? |
17:11.38 | hajekd | loud: yes, please |
17:11.47 | Zeeek | eikonex.com ? |
17:11.48 | *** join/#asterisk yashax (~yasha_x@69.15.218.218) |
17:11.56 | hajekd | don't wanna pay duty when buying at US store... |
17:11.59 | loud | https://shop.eikonex.net/catalog/default.php |
17:12.03 | zoa | hey i know that eikonex guy |
17:12.29 | Zeeek | I bought twice from eiko,nex and once from digium direct |
17:12.40 | Igor-BZ- | I have a little problem with asterisk... some one know when and where core call pvt->read in a channel? |
17:13.01 | hajekd | i bought my quadbri from Junghanns ;) |
17:14.32 | __Sparks_ | Wonder if anyone can help mere here! - If I make a call via my asterisk box, using my SIP phone to PSTN Number, i get a ringing tone on the phone way before the phone the other end actually rings. is this normal, or can it be corrected, as it is annoying! |
17:15.37 | zoa | im wondering if i should make ss7 for asterisk |
17:15.47 | brettnem | yes, pleaser |
17:15.49 | brettnem | -r |
17:16.01 | *** join/#asterisk HitTop (~Jack@host6614613596.biz.tor.fcibroadband.com) |
17:16.28 | ManxPower | zoa, Someone already has. |
17:17.03 | brettnem | ManxPower: it doesn't work in the US |
17:17.04 | zoa | i know but he is so slow :( |
17:17.09 | zoa | i still dont know his price |
17:17.23 | brettnem | no A-Link capability |
17:17.37 | hajekd | I hope I need FXS when want to connect fax to 400P :) |
17:17.54 | shido6 | take out the ||r or the ,r but that may just cut out a few rings |
17:17.58 | *** join/#asterisk tedh (nobody@angry.mob.net) |
17:17.59 | shido6 | __Sparks_ |
17:20.41 | zoa | guess it all depends on the price the others will ask |
17:20.48 | __Sparks_ | shido6, Is this in extensions.conf (Sorry I am new to this!!) |
17:20.48 | zoa | if its affordable, i wont do it |
17:20.58 | zoa | heya shido |
17:21.52 | tedh | Hello. I've read this is the place to ask my digium/asterisk questions. Whats the best way to go about it other than blurting out the questions? |
17:22.30 | zoa | hey kran |
17:22.32 | zoa | kram |
17:23.09 | wasim | tedh: go read the wiki at www.voip-info.org |
17:23.10 | vaewyn | tedh: blurt out the questions... and use pastebin.ca for large posts of configs/stuff |
17:23.26 | outtolunc | tedh, make sure they are well thought out questions with enough info someone here can help |
17:23.45 | tedh | I've been through much of voip-info already. This is actually a problem I'm having with the zaptel module. |
17:24.01 | *** join/#asterisk damnsure (~damnsure@wbs-146-171-124.telkomadsl.co.za) |
17:24.21 | Zeeek | <PROTECTED> |
17:24.22 | tedh | Here's the short version of the blurt: I am running Debian with a 2.6 kernel. Is it possible to get the zaptel module to work? I can't get it to recognize my linux source. |
17:24.29 | Zeeek | did you check with DE land? |
17:25.30 | tedh | Or should I give up and go to 2.4 kernel. the voip-info says its possible with 2.6. |
17:25.40 | vaewyn | tedh: yes it is VERY possible.. as in I do it :P I would shy away from the kernel packages and use the kernel.org tarballs |
17:26.13 | vaewyn | make sure you have a link from /usr/src/linux-26 to your source tree and voila! |
17:26.16 | tedh | Can you tell me whats specifically looking for when it says it can't find the linux source? |
17:26.23 | tedh | oh |
17:26.30 | tedh | i just have /usr/src/linux. |
17:26.33 | vaewyn | sorry linux-2.6 not linux-26 |
17:26.36 | tedh | maybe thats my problem. |
17:26.45 | vaewyn | hmm.. that should work also... but |
17:26.48 | vaewyn | *shrugs* |
17:27.11 | tedh | brb, going to try that. |
17:27.45 | yashax | hwo do I restart astersik from command line? |
17:27.59 | outtolunc | i'd remove your symlink |
17:28.08 | outtolunc | then use make clean; make linux26; make install |
17:28.32 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-18-115-202.buff.east.verizon.net) |
17:28.49 | outtolunc | it does a uname -r to find your source |
17:29.08 | vaewyn | you shouldn't have to 'make linux26'the makefile senses 26 automagically |
17:29.38 | outtolunc | we know how 'magical' the makefile can get |
17:30.06 | tedh | No luck vaewyn. It appears to be looking for /lib/modules/2.6.8-1-386/build on my system. Which isn't there after running the 'make dep' that the instructions instructed me to |
17:30.20 | tedh | Again, this is still using the debian kernel package. |
17:30.22 | __Sparks_ | okay, now I have another problem with Sipgate - If I call a PSTN number via my Asterisk box then hang up, the phone the other end rings for about 5 seconds longer - is this somthing I can fix! |
17:30.27 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
17:30.51 | damnsure | hi there, i'm playing with * for 2 days now and i got 2 voip phones running but i struggel with isdn on fc3, the card itself connects to the internet and i can go surfing but while dialing a number allways lacking dialtone appears?! |
17:30.54 | *** join/#asterisk keith778 (~kobrien@ool-4355f47e.dyn.optonline.net) |
17:31.17 | keith778 | Hello. Is anyone here running a Digium T400P card in a Dell 1750 server? |
17:31.26 | yashax | guys, how do I restart astersik from command line? |
17:31.28 | keith778 | I am having trouble finding a way to power the card |
17:31.31 | damnsure | reload |
17:31.53 | *** join/#asterisk PakiPenguin (~uppal@202.176.230.225) |
17:31.56 | *** join/#asterisk jobi (~jobi@lauga.ssvl.kth.se) |
17:32.18 | tedh | yashax: you can do an "asterisk -r" and then 'reload' |
17:32.23 | outtolunc | power the card? sure you aren't talking about the TDM400P |
17:32.38 | keith778 | yes, sorry the TDM400P |
17:32.45 | outtolunc | (note there is a 4 port t1 card that is a T400P) |
17:32.52 | yashax | thank you |
17:32.53 | vaewyn | yashax: asterisk -rx 'reload' |
17:32.56 | PakiPenguin | hello everyone , how can i have an extension number for my iax client? , i have this exten => 1075,1,Dial(IAX2/meow@meow/1075) , but it always give the userbusy |
17:33.16 | keith778 | Has anyone figured out a way to power the tdm400p in a dell server? |
17:33.21 | damnsure | anyone any idea what "lacking dialtone" means |
17:33.22 | Sedorox | remove the /1075 |
17:33.34 | shido6 | ZzZzzz |
17:33.49 | Sedorox | the iaxy doesn't have knowledge of the exten.. so just call it.. not pass a extention |
17:33.49 | shido6 | err |
17:33.56 | shido6 | u dont even need @meow , really |
17:33.59 | johnnyb | Is there a way to get the TDM400P to pick up the phone earlier? It seems to wait two rings until it actually decides to pick up the phone. |
17:34.08 | PakiPenguin | Sedorox: its a soft iax client |
17:34.18 | vaewyn | johnnyb: only if you don't want callerid |
17:34.19 | shido6 | exten => 1075,1,Dial(IAX2/meow|20|r) |
17:34.26 | Sedorox | PakiPenguin: shido6 is currect too... and you still don't need that |
17:34.29 | johnnyb | vaewyn: I don't. |
17:34.32 | outtolunc | keith778: there is a TDM400P install doc on digium.com |
17:34.33 | shido6 | exten => 1075,2,Voicemail,u1075 |
17:34.37 | Sedorox | the only time you need the pass the extention is when your doing to another server |
17:34.42 | shido6 | exten => 1075,103,Voicemail,b1075 |
17:34.45 | Sedorox | follow shido6 |
17:34.50 | outtolunc | http://www.digium.com/downloads/tdm_inst.pdf |
17:34.53 | PakiPenguin | got it |
17:34.53 | klicTel | is anyone aware of issues connecting Cisco call manager to * using SIP? |
17:34.56 | Sedorox | is it 103 or 102 for busy...? |
17:35.03 | johnnyb | vaewyn: how do I get it to answer sooner? |
17:35.04 | vaewyn | johnnyb: then set immediate=yes |
17:35.08 | johnnyb | Thanks! |
17:35.12 | vaewyn | no prob |
17:35.16 | PakiPenguin | thanks |
17:35.30 | fishboy1669 | night |
17:35.31 | *** join/#asterisk bobx (~bobx@lowfreq.trancemitter.org) |
17:35.31 | shido6 | klicTel, good luck! u can do it |
17:35.40 | vaewyn | johnnyb: that is in the zapata.conf btw... forgot that :} |
17:35.40 | *** part/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
17:35.42 | shido6 | callmanager needs the sip load tho |
17:35.58 | klicTel | shido6: does it take luck? |
17:36.15 | keith778 | outtolunc: yes, I have read that but it doen't talk about how to power the card in a dell server. As far as I can tell there is no 4 pin power on dell servers |
17:36.25 | keith778 | unless there is some adapter that I need |
17:36.52 | vaewyn | keith778: got to be 4 pin unless you are running diskless or SATA drives |
17:37.12 | PakiPenguin | <PROTECTED> |
17:37.12 | PakiPenguin | <PROTECTED> |
17:37.21 | jobi | I'm using asterisk as a GW from SIP to the PSTN, and would like the GW only to accept SIP calls coming from my SIP proxy (SER) |
17:37.25 | keith778 | Nope. The server is hot swap scsi so there is no 4pin power on the hd |
17:38.13 | jobi | can I have asterisk check for the source IP address of the incoming SIP calls? |
17:38.25 | jobi | or have SER authenticate itself somehow |
17:38.31 | vaewyn | keith778: bummer then... I think you are SOL |
17:38.35 | damnsure | anyone any idea what "lacking dialtone" means and how i can fix it, with i4l and a W6692 based card... |
17:38.46 | keith778 | yeah, thats what I was afraid of |
17:38.47 | Weezey | is there any way to detect a fax machine? |
17:38.52 | shido6 | keith778 |
17:38.53 | shido6 | what? |
17:39.10 | brettnem | wiki! |
17:39.25 | outtolunc | keith778: i'll trade you a 600sc for your 1750 <G> {ducks} |
17:39.36 | damnsure | i searched the wiki but nothing helped... |
17:39.53 | damnsure | i'm struggeling for a day now... |
17:40.14 | keith778 | shido6: yeah, hot swap scsi drive have a high density power adapter that is wired directly to the motherboard. Since the power supplies are also hot swap all |
17:40.33 | keith778 | of the power connectors are proprietary |
17:41.15 | keith778 | I was hoping someone hacked together a special cable to pull the power off of these connectors. Seems like I am the one that will be making the cable ;) |
17:42.32 | SuPrSluG | anyone using nagios w/ * |
17:42.42 | Uajal | My asterisk demo works with broadvoice. Congratulation to me! |
17:42.59 | *** join/#asterisk amer (desikukar@210.56.9.213) |
17:43.07 | brettnem | SuPrSluG: I setup asterisk and nagios.. hate the nagios setup.. |
17:43.20 | brettnem | actually, I prefered argus |
17:43.49 | shido6 | nagios is a pain |
17:43.52 | SuPrSluG | pia for certain. do u use it to call u if * goes down? |
17:43.56 | Uajal | yeah, I remember nagios setup. 2 days. But it worked. |
17:44.02 | shido6 | but so is getting out of bed |
17:44.03 | brettnem | call you with what? heh |
17:44.06 | *** join/#asterisk _Raptor_ (~RaptorBlu@p5480548A.dip.t-dialin.net) |
17:44.07 | _Raptor_ | hi# |
17:44.23 | yashax | which SIP softphone would you recommend to use for asterisk testing....? |
17:44.23 | brettnem | I'm using argus and midas.. midas is a nice setup |
17:44.28 | __Sparks_ | Ok, I seem to be able to use the "Flash" button on my Grandstream BudgeTone to have two calls on the go at the same time. - Is it possible to conferance these two calls? |
17:44.32 | amer | is there anyway I can set SIP "fromuser" in extensions.conf? |
17:44.33 | shido6 | xlite |
17:44.33 | brettnem | nice xmls and all |
17:44.37 | SuPrSluG | brettnem:never tried argus. |
17:44.44 | brettnem | it's nice |
17:44.48 | brettnem | midas is too |
17:45.00 | SuPrSluG | brettnem:is less bulky |
17:45.11 | brettnem | I'm less bulky? |
17:45.15 | brettnem | hmm.. thanks |
17:45.21 | Uajal | how to build autodialer from asterisk? can it be done through config files or I should use AGI? |
17:45.48 | yashax | shido6: thank you |
17:45.59 | _Raptor_ | i have a problem with speex codec: i have compiled speex and asterisk and i can load the modules codec_speex but when i add the line: codec=speex to oh323.conf then i get: |
17:46.00 | _Raptor_ | <PROTECTED> |
17:46.17 | *** join/#asterisk garyitcom (~Tech@119-114.8-67.tampabay.rr.com) |
17:46.21 | damnsure | Feb 17 19:45:22 WARNING[6900]: chan_modem_i4l.c:374 i4l_read: Device '/dev/ttyI1' lacking dialtone -- Hungup 'Modem[i4l]/ttyI1' what does that mean, couldn't find anything at voip-info |
17:46.25 | PTG123 | is their a way to force asterisk to transcode all calls, instead of doing a native bridge? |
17:47.24 | SuPrSluG | brettnem:apparently argus is a landfill monitor. argus.org |
17:47.42 | brettnem | haha |
17:47.42 | brettnem | I have a landfill I guess |
17:48.03 | SuPrSluG | me too. toxic pc parts everywhere |
17:48.52 | brettnem | SuPrSluG: http://www.voip-info.org/wiki-Asterisk+monitoring |
17:49.10 | SuPrSluG | brettnem:i'm there thanx |
17:49.21 | PTG123 | its anyone in here based in california? |
17:49.25 | brettnem | chck out midas too.. |
17:50.37 | Jlau515 | hi, anybody got time to help me troubleshoot my dialplan when dialing a zaptel channel? |
17:50.46 | _Raptor_ | has anyone an idea concerning my speex problem or a link to read about it? |
17:51.20 | Weezey | my PSTN comes in via a FXO on a SPA-3000, goes out the same way, how do I create a zapata channel for that? |
17:51.29 | Jlau515 | internal sip to sip calling works, when dialing a zaptel channel it fails |
17:51.42 | Jlau515 | not sure how to troubleshoot zaptel issues |
17:52.24 | amer | is there anyway I can set fromuser=callerIDNUM? (sip fromUser) |
17:53.34 | yashax | is there a quick doc that tells me how to set the Xlite with asterisk? |
17:53.41 | Jlau515 | my extensions.conf can be found here, http://pastebin.ca/6025 |
17:54.06 | amer | yashax: its just another SIP phone |
17:54.12 | amer | very easy to setup |
17:54.13 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
17:54.16 | Weezey | amer: http://www.voip-info.org/wiki-Asterisk+cmd+SETCIDNUM |
17:54.29 | SuPrSluG | brettnem:can u monitor dundi w/ argus? |
17:54.49 | amer | thanks weezey, this isn't what I want |
17:54.56 | yashax | amer: can you please tell me how? |
17:55.14 | Weezey | amer: sorry |
17:55.15 | amer | create a sip user in asterisk |
17:55.25 | amer | Weezey: np |
17:55.28 | Uther_P | you guys know where to point me for information about a really bad and a really LOUD echo problem going from SIP to ZAP. (sometimes I can hear myself VERY loud, and the other party not at all... and sometimes its fine) |
17:55.56 | junky[work] | Uther_P: ya've echocancelwhenbridged=yes ? |
17:56.07 | Uther_P | ahh, lemmie check |
17:56.10 | Zeeek | amer you want many users dynamically possible? |
17:56.22 | yashax | amer: you mean the extension? |
17:56.38 | amer | Zeeek: yes |
17:56.48 | amer | I am running asterisk in proxy mode |
17:56.51 | Zeeek | I don't think you can do it |
17:56.58 | amer | :( |
17:57.15 | yashax | k... created the extension, now...? |
17:57.16 | junky[work] | <PROTECTED> |
17:57.20 | Zeeek | but what do I know? |
17:57.22 | junky[work] | yashax: reload :) |
17:57.34 | damnsure | anyone experience with * on fedora core 3 and isdn? |
17:57.40 | Uther_P | junky[work]: thanks... but I don't find any documetation in the wiki :/ |
17:58.06 | *** join/#asterisk Secretive (~polarisx@c-67-161-5-149.client.comcast.net) |
17:58.08 | Secretive | Hey |
17:58.10 | amer | another IAX or SIP proxy fwds all calls to my asterisk but when asterisk fwds the call its sets the FROM field to asterisk@10.0.0.1 |
17:58.31 | amer | yashax: did u create a sip user first? |
17:58.38 | junky[work] | uther |
17:58.38 | junky[work] | echocancel=yes |
17:58.39 | junky[work] | echocancelwhenbridged=no |
17:58.43 | junky[work] | both both at yes |
17:58.54 | junky[work] | and ya'll have to restart (not reload) ur * |
17:58.58 | junky[work] | try and gimme feedbacks |
17:59.04 | Uther_P | ok, thanks |
17:59.14 | junky[work] | amer: SetCallerID |
17:59.17 | yashax | amer: sip user = extension? |
17:59.18 | Secretive | Can anyone make sense of this situation: I can dial out with my VOIP Phones, but when I try to call between phones...the phone will ring (sometimes) but when I answer it, neither party can hear anything. |
17:59.27 | Uther_P | I'll try it.... but I was just mentioning that I can't seem to find the documentation for that option in the wikui |
17:59.29 | Zeeek | amer you have no luck, we all wanna set your callerid! |
17:59.34 | Uther_P | s/kui$/ki/ |
17:59.34 | junky[work] | FROM field? ya mean the source? |
17:59.36 | amer | hahahaha |
17:59.42 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
17:59.47 | amer | its not the callerID |
18:00.00 | Zeeek | amer you do C programming? |
18:00.01 | amer | sip:fromuser is different |
18:00.14 | amer | a lil bit, I can hack code |
18:00.20 | Jlau515 | can anybody help me with my dialplan? |
18:00.26 | amer | yashax: yes |
18:00.29 | Zeeek | amer prolly your only answer |
18:00.29 | HitTop | wat is it for for chan_agent.so? |
18:00.35 | yashax | amer: done.... next? |
18:00.39 | junky[work] | describe ur FROM field exactly? |
18:00.43 | amer | now create an entry in extensions.conf |
18:00.51 | Zeeek | maybe write a tiny app to manipulate that data - if you can find it :) |
18:00.56 | tzanger | grrr |
18:01.00 | tzanger | my tr08 channel bank won't ring |
18:01.05 | tzanger | it worked before |
18:01.51 | HitTop | where can i find the description for each modules ? |
18:01.51 | Uther_P | junky[work]: I have echocancel and echocancelwhenbridged already both set to yes |
18:02.08 | junky[work] | which version? |
18:02.13 | stepcutHM | hrm, do I have to edit the source code to change the text of the email that is sent when you get voicemail ? |
18:02.23 | junky[work] | its only from sip to zap? zap to sip ? or both? |
18:02.27 | jsolares | stepcutHM: no |
18:02.36 | Uther_P | I think both |
18:02.41 | Secretive | Anyone know what this means: Feb 17 12:07:14 WARNING[26440]: chan_sip.c:728 retrans_pkt: Maximum retries exceeded on call 40364e1a7270f78d7bdf16ed4ef6514e@66.55.69.242 for seqno 102 (Critical Request) |
18:02.41 | Secretive | Feb 17 12:07:14 WARNING[26440]: chan_sip.c:1168 find_peer: Looking for SIP: 1.301 |
18:03.06 | Uther_P | it doesn't happen sip to sip, I'm sure of that much... and I've bridged a call zap to zap, and I didn't notice there |
18:03.09 | jsolares | stepcutHM: serveremail= in voicemail.conf |
18:03.17 | junky[work] | try it and msg me after. |
18:03.38 | vaewyn | anyone from voipsupply on here? |
18:03.44 | stepcutHM | jsolares: thanks, I see it now :) |
18:03.45 | Uther_P | Secretive: I've been getting those messages for a while now... noone could tell me what they ment either... I just ignoret hem |
18:03.47 | Zeeek | Secretive it means it ain't finding to phone |
18:04.10 | *** part/#asterisk tedh (nobody@angry.mob.net) |
18:04.12 | Uther_P | junky[work]: its inconsistent though |
18:04.51 | damnsure | Anyone knows what this means: WARNING[6900]: chan_modem_i4l.c:374 i4l_read: Device '/dev/ttyI1' lacking dialtone |
18:04.52 | damnsure | <PROTECTED> |
18:04.53 | Secretive | Zeeek: Why wouldn't it be able to find the phone |
18:05.56 | Zeeek | try an experiment. Turn off or unplusg a sip phone that's registered while watching CLI |
18:07.02 | Secretive | Zeeek, I have 3 phones setup all on the same network behind a firewall. Could this be the problem? |
18:07.30 | Zeeek | where is asterisk? Behind he same fw? |
18:07.43 | Secretive | Nope, on a dedicated server at a remote location. |
18:08.01 | Zeeek | what SIP ports in the phones? |
18:08.11 | Zeeek | how are they forwarded? |
18:08.41 | Secretive | Zeeek: I'm not quite sure I understand you're question? They go through a Linksys Router. |
18:08.52 | Secretive | *your |
18:09.00 | Zeeek | and are ports being forwarded to the phones? |
18:09.28 | vaewyn | anyone have the polycom firmware and bootrom images? @#$@#$ polycom won't let end users download them |
18:10.22 | vaewyn | and evidently voipsupply doesn't hand them out as the VAR either |
18:11.08 | HitTop | when asterisk starts, it will load everything under /usr/lib/asterisk/modules (except those that are listed noload in modules.conf) right? |
18:11.29 | HitTop | so.. including those app_*.so and format_*.so etc? |
18:12.42 | junky[work] | HitTop: right |
18:13.10 | Jlau515 | having issue with sip to zap configuration please help |
18:13.38 | HitTop | is there any place where I can find wat each modules is for? (other than in the actual source code, because the descritpion is just a name in the source) |
18:15.09 | junky[work] | HitTop: make the doc. |
18:15.10 | hardwire | anybody have a snom 220? |
18:15.20 | hardwire | why when I put a call on hold.. does it give me dialtone :) |
18:15.27 | ManxPower | Ugh! The "Stealth Asterisk Install" was anything but "stealth" |
18:15.52 | ManxPower | Nobody bother to tell us that there were other customers on the CT1, nor that they were running data, nor that they had multiple hunt groups. |
18:17.44 | HitTop | Junky: im now trying to make asterisk small as possible so that i will run well in a linksys router |
18:17.52 | vaewyn | ManxPower: I think that qualifies as "ooops!" |
18:17.54 | *** join/#asterisk nwhit (~chatzilla@65.107.59.67.ptr.us.xo.net) |
18:18.53 | nwhit | help.... I loaded the currect cvs and now the voicemail system is putting a B in front of all the passwords, and not accepting them. Any suggestions? |
18:19.34 | ManxPower | vaewyn, Yeah, nut we designed the setup so NO changes were required on the CT1, the channel bank, or the corporate PBX. So to back out we just had to patch the t-1 cables back to where they were and everything was back the way it was. |
18:19.46 | amer | hey manxPower |
18:19.54 | *** join/#asterisk ionix (~ioni@66.38.219.151) |
18:19.55 | ionix | Hey |
18:20.21 | ionix | anyone has a way for asterisk to pickup the name of the caller from the ANI ? |
18:20.27 | ionix | like RBOC database or something |
18:20.40 | ManxPower | ionix, um, ANI does not provide NAME. |
18:20.44 | ionix | I know |
18:20.47 | ManxPower | CLID provides name and that works just fine. |
18:20.48 | ionix | ANI provides number |
18:20.58 | ionix | but I want to get the CLID from the ANI |
18:21.04 | ionix | trying to figure a way |
18:21.17 | *** join/#asterisk CybreWulf (cybre@killcybre.org) |
18:21.23 | Uther_P | does anyone see a problem arising from having 4 analog POTS running over a Cat-5, each on its own twisted pair? |
18:21.32 | HitTop | I want to ask for format_*.so, they're there for converting right? during a call, if both sides are using the same codec (let say ulaw), then asterisk will bridge the call so asterisk would just pass packets over right? |
18:21.40 | shido6 | think thats been done, Uther_P |
18:21.46 | Uther_P | I ask because I'm having an echo problem, and wonder if that could be a possible cause |
18:22.00 | Uther_P | I didn't think it would be a problem, I've done it before |
18:22.13 | Uther_P | just reaching for the answer... but I think I know its from our provider |
18:23.21 | Uther_P | a SIP to SIP call internally has virtually no delay and no echo... if I hook up a regular analog phone to the lines from our provider and call a cell phone, there is anywhere from 400-800ms delay :/ |
18:23.39 | Uther_P | I think my dumbass provider is overselling |
18:23.54 | *** join/#asterisk eipi (eipi@153-218-114-200.fibertel.com.ar) |
18:24.15 | ionix | whos the dumbass provider ;) |
18:24.46 | damnsure | Anyone knows what this means: WARNING[6900]: chan_modem_i4l.c:374 i4l_read: Device '/dev/ttyI1' lacking dialtone |
18:25.04 | amer | there is no dialtone |
18:25.06 | nwhit | I loaded the currect cvs and now the voicemail system is putting a B in front of all the passwords, and not accepting the passwords. Any suggestions? |
18:25.28 | damnsure | thx :) any idea how to fix it? |
18:25.29 | ManxPower | nwhit, Don't use the developement version if Asterisk. That's my suggestion. |
18:25.51 | damnsure | i'm struggeling for 2 days on this, can't find anything |
18:25.55 | ManxPower | damnsure, almost nobody uses i4l with Asterisk. They use CAPI or ZapBRI |
18:26.21 | damnsure | so i should stick to capi |
18:26.27 | __Sparks_ | IOf I have more than one SIP Provider registering in sip.conf, i seem to be getting errors like "chan_sip.c:6801 handle_response: Failed to authenticate on REGISTER to '<sip:ACCOUNT@fwd.pulver.com> - Do I need to specify ports? |
18:26.36 | nwhit | manxpower: ok... not the answer i was looking for |
18:27.30 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
18:27.35 | Secretive | Okay guys, here's the problem. I can call extensions remotely through Asterisk and my Hard-phones. But if I try to call a hard phone that is on the same network as me, it doesn't work. |
18:28.25 | Secretive | They all dial out just fine, and recieve calls from remote networks.. |
18:28.39 | Secretive | ...but if I try to call a phone that is on the same network (behind the same router) as me....it just doesn't work |
18:28.41 | *** join/#asterisk santiago (~santiago@201.245.167.88) |
18:28.44 | Tall-guy | Gents, I have a need for 8 pstn's and 16 sets......I'm thinking T1 to a channel bank of fxo's and fxs's....whats the best channel bank recommendation? |
18:28.55 | Secretive | Just get this: Feb 17 12:31:05 WARNING[26440]: chan_sip.c:1168 find_peer: Looking for SIP: 1.303 |
18:28.55 | Secretive | -- Executing Dial("SIP/1.303-cb17", "SIP/1.301") in new stack |
18:28.55 | Secretive | Feb 17 12:31:05 WARNING[26440]: chan_sip.c:1168 find_peer: Looking for SIP: 1.301 |
18:28.55 | Secretive | -- Called 1.301 |
18:30.23 | ManxPower | Secretive, Do you have a [1.301] section in sip.conf? |
18:31.56 | shido6 | Secretive do you have peers in your sip.conf or friends? |
18:32.11 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
18:33.05 | shido6 | answer Manxpowers ? first :) |
18:33.10 | *** join/#asterisk paulc (~paulc@S010600062586a0b4.vc.shawcable.net) |
18:33.15 | Zeeek | Secretive is being ... secretive |
18:33.23 | Zeeek | he won't say! |
18:33.41 | Secretive | shido6: I have friends. |
18:33.44 | amer | can I use Variables in sip.conf? like callerIDNum |
18:33.49 | Secretive | Manx: Yes, 1.301 is in my sip.conf ;-) |
18:33.52 | Zeeek | friends good. |
18:33.56 | terrapen | Those damned blue-collar tweekers |
18:34.07 | Secretive | These are PolyCom Soundpoint IP 600's |
18:35.30 | Secretive | Is this problem perhaps because of having multiple phones behind the same Linksys router? |
18:37.10 | Tall-guy | phone lines?? |
18:37.49 | doughecka_ | try allowing reinvites |
18:38.09 | Tall-guy | sorry, mis read there.... |
18:39.26 | bjohnson | Secretive: the simple fact of having multiple phones on a LAN would not cause you trouble |
18:39.57 | Tall-guy | no group hugs, but group smaks are plentiful :) |
18:40.25 | bjohnson | Secretive: did you already do sip show peers? |
18:40.36 | *** join/#asterisk benno2 (~benno2@host40-116.pool80117.interbusiness.it) |
18:41.37 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
18:41.41 | Mw3 | is there any iax softphone for windows? |
18:41.52 | doughecka_ | ~firefly |
18:41.53 | jbot | hmm... firefly is http://virbiage.com/firefly/download/firefly-thirdparty.exe |
18:41.56 | doughecka_ | there ya go |
18:42.13 | Sedorox | is it better then x-lite? |
18:42.18 | doughecka_ | far better |
18:42.20 | buddah | anyone able to take a look at an extensions.conf entry really quick. i'm having trouble getting an entry to show up before a different one in dialplan |
18:42.29 | buddah | its set up that way on extensions.conf |
18:42.43 | Zeeek | http://www.hotsip.com/sip/tutorial/cartoon/index.html |
18:42.45 | buddah | but when i do show dialplan, its after instead of before the other line |
18:43.17 | mikegrb | buddah: the dialplan is sorted so it doesn't stay in the order you specify |
18:43.38 | mikegrb | buddah: if order is important put the two items in different contexts and then include those contexts in the right order |
18:43.41 | buddah | mikegrb: how can i get it to change the order, anyway to set priority? |
18:43.46 | buddah | hmm |
18:43.57 | shido6 | iaxcomm |
18:44.06 | shido6 | is a windows iax softphone |
18:44.06 | Tall-guy | zeek: funny |
18:44.13 | buddah | mikegrb: they are both international, but one is _0118802, and if no 8802 then its just _011 |
18:44.15 | Secretive | It appears that the problem I'm having is multiple phones being behind the same NAT router with only one public IP. |
18:44.30 | buddah | so make like internationala have the _0118802 and include international? |
18:44.43 | Zeeek | Secretive that's what I was getting at earlier |
18:44.44 | buddah | that way if no 8802 it goes to international with just _011 |
18:44.45 | buddah | ? |
18:44.54 | Nugget | NAT blows goats. |
18:45.03 | Zeeek | change the port on your clients: set client1 to 5060, client2 to 5061 etc |
18:45.07 | mikegrb | buddah: put both in thier own context |
18:45.30 | Zeeek | then forward 5060 to client1, 5061 to client 2, 5062 to client3 etc |
18:45.43 | Secretive | Zeeek: Now you have me somewhat confused. |
18:45.45 | Zeeek | It works for me in exactly that situation |
18:46.02 | *** join/#asterisk attack_ (~attack@213-213-141-51.xdsl.is) |
18:46.02 | Zeeek | The clients are behind NAT router, ya? Linksys |
18:46.03 | Secretive | How can I force my phone to connect to Asterisk using a different port? Doesn't Asterisk only listen on one port? |
18:46.14 | Secretive | Yes, a fancy little Linksys/Cisco thing |
18:46.19 | *** join/#asterisk oej (~oej@apollo.webway.se) |
18:46.21 | Tall-guy | Are these clients on the OTHER side of the NAT router that Asterisk is on???? |
18:46.21 | Zeeek | the phone will call on 5060 but receive the return on 5061 |
18:46.36 | Zeeek | believe me it works and astersik doesn't care |
18:46.40 | Zeeek | astersick |
18:46.52 | Zeeek | asterRisc (with reduced instriction set) |
18:47.08 | attack_ | the S100I "IAXy" model from Digium, can that be used to call conventional phones from a computer? |
18:47.42 | Zeeek | yes if asterisk server is hooked to the PSTN |
18:47.44 | attack_ | hmm |
18:47.44 | *** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net) |
18:47.48 | Tall-guy | nugget: yeah, I'm verbose with the shifted number keys :) |
18:47.51 | Secretive | The Asterisk server is at a remote location, on a pubic network. |
18:47.51 | Zeeek | but then you won't need a computer |
18:48.23 | Zeeek | Secretive: set client2 to 5061 - set linksys to forward 5061 to the ip of the clioent. Thats' all |
18:48.24 | attack_ | what i'm looking for is a solution that allows me to call, say, someone in sweden over the internet at the local rate there |
18:48.27 | Tall-guy | secretive: ah, ok, then listen to Zeeek, he's on the right track |
18:48.27 | Secretive | Zeeek: Is this what you're telling me to change: reg.1.server.1.port="" |
18:48.42 | Zeeek | I dunno what that is |
18:48.50 | Zeeek | attacke read this pls |
18:48.51 | Zeeek | Starter tutorial: |
18:48.51 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
18:48.51 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
18:48.51 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
18:48.51 | Zeeek | THE reference of the moment: |
18:48.52 | Zeeek | http://www.asteriskdocs.org |
18:48.58 | attack_ | like, placing a gateway or something that transfers calls from the internet to conventional lines |
18:49.01 | attack_ | yes |
18:49.11 | attack_ | i've been reading the voip-wiki for like 2 hours now |
18:49.18 | Zeeek | attack_ the first and second articles above have ALL the answers to your next ten questions |
18:49.27 | attack_ | aight... i'll look at it |
18:49.28 | attack_ | thanks |
18:49.30 | harryvv | onlamp is pretty good. |
18:49.31 | Zeeek | forget the wiki these are articles |
18:49.43 | attack_ | mkay |
18:49.58 | Zeeek | wiki is grat once you know 90% of everything :) |
18:49.58 | harryvv | I goto it once in a while :) |
18:50.22 | harryvv | zeek, you have deb and a zap running?> |
18:50.23 | Zeeek | Secretive I need the points for helping solve at least one problem today |
18:50.43 | Zeeek | Secretive - help me out - try what I say - it has been working here for a year now |
18:51.02 | Zeeek | with Linksys WRT54g, GT BT102, X-Lite etc |
18:51.06 | ariel_ | afternoon all |
18:51.14 | Zeeek | evening |
18:51.44 | ariel_ | Zeeek, yes your right for your area. How is the France these days? |
18:51.46 | ManxPower | Secretive, Asterisk may think peer/friend/users with a . in the name is a host. |
18:51.55 | Zeeek | wonderful but cold |
18:52.03 | ManxPower | Secretive, If you do a "sip show peers" do you see 1.301? |
18:52.20 | Secretive | Manx, Yes. |
18:52.28 | Zeeek | he has multiple SIP clients behind NAT all on 5060 - no way |
18:52.59 | harryvv | btw, I do not have a sip phone or ata install. Wife still uses the phone hooked to the same land line. When my machine goes into ivr she hears it and it does not stop. I am asuming that when a ata or sipphone is picked up * will detect this and stop the ivr? |
18:53.07 | ManxPower | Zeeek, Any DECENT nat router will handle that JUST FINE. |
18:53.25 | ManxPower | Zeeek, Also the error message he is getting does not indicate a NAT issue. |
18:53.28 | Zeeek | that depends on what you call a decent NAT router |
18:53.39 | harryvv | IVM |
18:53.47 | ManxPower | Zeeek, Linksys is "decent" for this discussion I think. |
18:53.48 | *** join/#asterisk HitTop (~Jack@host6614613596.biz.tor.fcibroadband.com) |
18:53.59 | harryvv | i have a d-link its been good. |
18:54.01 | Zeeek | I agree I like mine a lot |
18:54.18 | harryvv | What soho routers should be avoided? |
18:54.20 | Zeeek | but I'm glad you limit the scope to this discussion |
18:54.46 | Zeeek | otherwise I'd have to call you in later discussions in case you trashed the same router :) |
18:55.30 | *** join/#asterisk Gwemo (~gen@nsc66.147.62-163.newsouth.net) |
18:56.49 | Tall-guy | Pix 501 makes a neat device.....has SIP fixup protocl/proxy thing too....but then again, its 500 bucks. |
18:57.41 | vaewyn | wrt54g rock |
18:57.43 | PatrickDK | heh, we use the hell out of pix devices |
18:57.44 | Zeeek | I guess the reason I'm so hot on forwarding has to do with the fact that my SIP hardphone doesn't register. |
18:58.26 | ManxPower | Zeeek, That would do it. |
18:58.27 | Zeeek | so sitting behind the router, I was assuming that asterisk would call it to qualify |
18:58.51 | Zeeek | which it does. Hence, the 5061 for X-Lite (which does reg) |
18:58.56 | ManxPower | Zeeek, You can't ever do manual port forwarding with multiple phones behind the same NAT router all using port 5061 |
18:58.56 | Tall-guy | patrickdk: have you tried "eyebeam" from xten with the pix'es? |
18:59.08 | ManxPower | You need to rely on the router's port number translations |
18:59.27 | Zeeek | no I use 5060, 5061, 5062 |
18:59.57 | HitTop | hi, i want to know how to disable logging for asterisk |
19:00.14 | Zeeek | unfortunately you missed me doing one of the alltime stupidest network manoeuvres ever earlier today |
19:00.28 | Tall-guy | zeek: 220V into 10 baseT? :) |
19:00.32 | harryvv | zeek what was that |
19:00.56 | HitTop | because right now. i have this error msg: "Logger Warning: Unable to open log file '/var/log/asterisk/messages': No such file or directory" |
19:01.04 | Zeeek | I set the gateway to 192.168.2.1 to have a second LAN card on a different subnet. Then I wonderd why I couldn't ping the new gateway |
19:01.04 | zoa | touch that file |
19:01.19 | harryvv | Last fortune 500 company I worked at there sprinker guys tested it in the data center during the week and it shut down the data centers power to all 100 servers. |
19:01.49 | Zeeek | Having set the gateway to a new ip and leaving the network interface on the old subnet was making it a big challenge to communicate |
19:01.51 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
19:02.00 | harryvv | 1 million dollars in transactions were lost for that day. |
19:02.01 | harryvv | :) |
19:02.26 | Zeeek | so who am I to be telling people to forward ports? |
19:02.49 | greg_work | harryvv: why are there sprinklers in the data center?? |
19:03.10 | Zeeek | to make the girls blouses transparent? |
19:03.25 | vaewyn | halon kills rabbits |
19:03.27 | vaewyn | :P |
19:03.28 | greg_work | you're supposed to use that gas stuff that doesn't hurt electronics (just kills people.. ;) ) |
19:03.39 | Nivex | Halon |
19:03.48 | mikegrb | halon is illegal |
19:04.02 | mikegrb | there is a new improved enviromentally friendly replacement |
19:04.11 | mikegrb | but still kills humans |
19:04.14 | mikegrb | it has a number |
19:04.16 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
19:04.32 | mikegrb | I walked past the cylendars 50 times a day for 2 years but I don't remember the numbers |
19:04.36 | mikegrb | was like 4 digits though |
19:04.38 | greg_work | i actually saw a quick spot on discovery channel or something the other day about some new one that puts fires out instantly but doesn't hurt people |
19:04.58 | riksta | wtf is halon |
19:05.09 | greg_work | it was just a promo though (like 4 seconds long), i didnt actually see the segment on it |
19:05.19 | vaewyn | riksta: liquid/gas that puts out fire by eating all the oxygen in the room |
19:05.21 | Nugget | riksta: we are not a google proxy. |
19:05.34 | riksta | like i can be arsed to search for that? |
19:05.46 | riksta | vaewyn: cool |
19:05.57 | moonwick | I don't know how a chemical can starve a fire of oxygen without starving a human. |
19:06.00 | Nugget | riksta: *shrug* it's not our job to ensure that you aren't ignorant. |
19:06.02 | harryvv | Actually I think thay were testing the firealarm system . I dont know how it all worked but it cut power to the data center and the UPS were to kick in still providing power and it did not. Gradually each server started powering down and everyones pagers started to go off in IT. Everyone was running to the server room |
19:06.08 | RoyK | ~lart riksta |
19:06.15 | riksta | can't you just fill the room with CO2? |
19:06.26 | RoyK | CO is more effective |
19:06.31 | harryvv | There is halon in the data center |
19:06.39 | RoyK | nice |
19:06.39 | moonwick | riksta: and kill any humans that don't get out fast enough? |
19:06.50 | harryvv | RoyK I have worked with halon in the millitary it replaced co2 |
19:06.50 | riksta | moonwick: well how can halon not have the same effect |
19:06.51 | mikegrb | harryvv: I doubt it |
19:06.52 | vaewyn | moonwick: no worse than halon |
19:07.03 | mikegrb | harryvv: halon isn't in use any more |
19:07.05 | moonwick | vaewyn: exactly. |
19:07.08 | harryvv | yea halon can kill it replaces or eats oxygen |
19:07.13 | riksta | a few humans aren't as important as my data :D :D |
19:07.18 | vaewyn | riksta: is the same effect... halon is just easier to maintain the volume needed |
19:07.30 | mikegrb | my last employer I was there when we had to retrofit the old halon system |
19:07.33 | RoyK | only halon isn't adopted as oxygen by the lungs |
19:07.35 | harryvv | halon is still use for aircraft in the millitary. |
19:07.43 | RoyK | nice |
19:07.55 | RoyK | aircraft fire goes out, all people die |
19:07.55 | mikegrb | harryvv: but you said it was in your datacenter |
19:08.00 | Tall-guy | (and over the frier in mcdonalds) |
19:08.01 | harryvv | huge 50 gallon unit I had to wheel around. |
19:08.02 | RoyK | aircraft falls down |
19:08.02 | mikegrb | harryvv: also, military doesn't use it anymore either |
19:08.06 | greg_work | harryvv: finding out your UPS's don't work when you actually need them sucks.. you need to test every once in a while ;) |
19:08.20 | harryvv | mike, yes as of 4 years ago when I worked there thay had a halon system |
19:08.28 | mikegrb | december 2001 |
19:08.32 | mikegrb | halon became illegal |
19:08.36 | harryvv | greg, well thay are tested on sundays. |
19:08.57 | mikegrb | I worked on retrofitting a military halon system to the replacement gas |
19:09.02 | harryvv | mike, you mean new sales of halon? what about brandfauthered systems? |
19:09.07 | Nugget | fm200 is the replacement. |
19:09.13 | riksta | "While the production of Halon ceased on January 1, 1994 under the Clean Air Act, it is still legal to purchase and use recycled Halon and Halon fire extinguishers. In fact, the FAA requires all commercial aircraft to exclusively use halon." |
19:09.36 | mikegrb | harryvv: grandfathered is not alowed either |
19:09.39 | Sedorox | hmmm |
19:09.40 | harryvv | okay |
19:09.43 | harryvv | that sucks |
19:09.50 | mikegrb | Nugget: yes! |
19:09.55 | harryvv | what does halon do eat oxygen? |
19:09.55 | mikegrb | Nugget: that's the ticket |
19:09.58 | Delvar | nn all |
19:10.05 | vaewyn | harryvv: yep... big time |
19:10.08 | harryvv | okay |
19:10.24 | mikegrb | december 2001 is when it was outlawed in most of europe |
19:10.30 | mikegrb | jan 94 is usa |
19:10.32 | ariel_ | military uses 3 systems depending on if it was replaced or not. 1) ClorobromoMethane, 2) Halon 3) FM200/b All of which will kill you so pilots and crew ware o2 masks. |
19:10.35 | harryvv | A chemical agent that perhaps binds with oxygen rendering it uless as a oxident. |
19:10.41 | mikegrb | er new halong |
19:10.52 | *** join/#asterisk _mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
19:10.53 | mikegrb | dunno when all halon was outlawed in us |
19:10.58 | mikegrb | I''d guess around 200 |
19:11.01 | mikegrb | 2000 even |
19:11.06 | *** part/#asterisk _mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
19:11.22 | riksta | 1994 |
19:11.28 | ariel_ | halon is outlawed in new setups but if it's inplace it's not required to take out unless it's damaged. or used. |
19:11.45 | *** join/#asterisk _mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
19:11.51 | harryvv | Such thing as foam fireextinguishers? |
19:11.51 | mikegrb | ariel_: yes, as we had to take out the halon system to avoid huge fines |
19:12.02 | mikegrb | ariel_: and it hadn't been used or damaged |
19:12.05 | mikegrb | harryvv: AFFF |
19:12.18 | mikegrb | harryvv: it is water with a foam additive |
19:12.24 | harryvv | I see |
19:12.30 | mikegrb | that is what is used at airports |
19:12.34 | Nugget | '94 sounds about right. I was installing a fire-supression system at tobias.com around then and I recall we chose fm200 because it was clear that halon wouldn't be a long-term solution. |
19:12.59 | Nugget | I think we could still buy halon then but were warned that we might never be able to recharge it |
19:13.13 | mikegrb | ariel_: thay may have been in the past but around 2000/2001 it was outlawed completely including existing installations |
19:13.20 | ariel_ | Nugget, yes your right. just like Freon 12. |
19:13.30 | harryvv | yup |
19:13.39 | harryvv | r13 replaced it |
19:13.51 | ariel_ | mikegrb, just last year we still had in the building for the FAA halon system. |
19:14.10 | ariel_ | harryvv, R1344 i think it was called to replace 12 |
19:14.11 | mikegrb | doubtful |
19:14.19 | harryvv | I need to goto a live fire extinguishing course some time :) did it in the millitary. |
19:14.32 | ManxPower | What is so good about Haylon .vs. just plain nitrogen? |
19:14.34 | mikegrb | harryvv: me too |
19:14.36 | harryvv | arial, probebly called r13 for short |
19:14.43 | mikegrb | harryvv: we got to put burning planes out and such |
19:14.45 | ManxPower | Isn't the goal to just push all the oxygen our of the room? |
19:14.46 | mikegrb | was quite fun |
19:14.49 | mikegrb | and hot |
19:14.58 | mikegrb | ManxPower: yes |
19:15.21 | mikegrb | ManxPower: halon and fm200 chemically "eat" the O2 as well though |
19:15.36 | mikegrb | ManxPower: which is what makes them more effective then CO2 |
19:16.00 | ManxPower | mikegrb, Ah! OK. |
19:16.11 | harryvv | manx, no the goal is to seperate the fuel fumes from the source and seperate the air from the fumes. putting a barrier between them puts the fire out. People say gasoline is flamable and I say no its not its the hydrocabons that leave it thats flamable. Thay look at me strange :) |
19:16.26 | mikegrb | indeed |
19:16.31 | mikegrb | gas is very much not flammable |
19:16.40 | ariel_ | mikegrb, sorry look at this site:http://www.reliablefire.com/halon/halon.html |
19:16.47 | ManxPower | harryvv, Like when I tell people "the web" is not "the internet"? 8-) |
19:16.51 | vaewyn | ever tried to light gas when it is 20 below :} |
19:16.53 | harryvv | heheh |
19:16.54 | ariel_ | Halon is still in use today. |
19:16.58 | mikegrb | harryvv: that's what the AFFF does, sits on top of oil/gas/etc |
19:17.37 | dsmouse | so |
19:17.41 | ariel_ | ok lets get back to Asterisk |
19:17.51 | dsmouse | is there supposed to be a conf... cst... nm |
19:18.08 | RoyK | ariel_: ass-per-risk |
19:18.44 | Mw3 | when will the conf start ? |
19:18.49 | Mw3 | in 40 minutes ? |
19:18.55 | harryvv | Having been a aircraft tech was pretty interesting. Actually the millitary service. One day I was refuling my aircraft a heavylift helicopter and heard this shhhhhh and fuel supervisor said SHUT THE Fuel off! I said what? looked down and massive stream of fuel was pooring out from underneath. Fifty Gallons spilled out of it in less then one min. |
19:18.56 | dsmouse | Mw3: looks like it |
19:18.56 | ariel_ | RoyK, yes but like I said it's still in use. I have seen it and I know of an FAA building with it still on halon 1301 |
19:19.16 | harryvv | yea |
19:19.18 | harryvv | :) |
19:19.27 | ariel_ | harryvv, what service where you in? |
19:19.30 | Mw3 | ok, i'm in CET so if 40 minutes left i've managed to convert thist cst time to cet |
19:19.31 | harryvv | usaf |
19:19.33 | Mw3 | :) |
19:19.37 | ariel_ | same here |
19:19.51 | harryvv | OFSCOM heavy lift crew chief |
19:19.51 | dsmouse | what's cet? |
19:19.56 | dsmouse | or where? |
19:19.59 | ariel_ | I was a flight Mech/Crew chief on C-130e/h |
19:20.01 | Mw3 | central europen time |
19:20.03 | harryvv | ohhyea |
19:20.07 | Mw3 | hungary |
19:20.11 | dsmouse | ah, yea |
19:20.15 | dsmouse | so like gmt+2? |
19:20.23 | harryvv | arial cool ever work on the spectere's ? |
19:20.25 | Mw3 | +1 imho :) |
19:20.39 | dsmouse | +1, ok |
19:21.02 | ariel_ | harryvv, I was mainly in the Grey ghost program working EC130e/h special ops. |
19:21.09 | harryvv | ahh yea |
19:21.15 | harryvv | spookey :) |
19:21.33 | ariel_ | I was in the PI and Kadenia for almost 8 years |
19:21.47 | harryvv | PI |
19:21.49 | harryvv | :) |
19:22.34 | *** join/#asterisk Moc____ (~mochouina@64.235.210.66) |
19:22.42 | zoa | i was in kindergarten |
19:22.44 | zoa | nice to meet you |
19:22.44 | Moc____ | hi all |
19:23.31 | benno2 | stupid question: if I have a zaphfc card. does I need to eg use Dial(Zap/g1) in extensions.conf ? I used an i4l card where it was dial(Modem/g1..) |
19:24.07 | RoyK | stupid question, yes |
19:24.20 | RoyK | chan_zap is Dial(Zap/something |
19:24.32 | RoyK | chan_modem is Dial(Modem/something |
19:25.29 | benno2 | RoyK: thks :) |
19:25.36 | *** part/#asterisk santiago (~santiago@201.245.167.88) |
19:28.48 | RoyK | chan_h323 is Dial(H323/asdf |
19:28.48 | RoyK | etc |
19:28.48 | RoyK | ad infinitum |
19:28.48 | jero_SFLphone | can anyone help me in guessing why I'm getting one of 10 caller IDs on a TDM400 ? |
19:28.48 | Zeeek | I never could ad infinitum |
19:28.48 | RoyK | because of quantum |
19:28.49 | RoyK | Zeeek: no? |
19:28.49 | Zeeek | I can easily ad nauseum though |
19:28.49 | RoyK | :) |
19:28.49 | RoyK | bibo ergo sum |
19:28.49 | Zeeek | wanna see? |
19:28.49 | Zeeek | caveat emptor (esp. non-Digium clone cards) |
19:28.49 | vaewyn | Zeeek: I always tell people that tell me that to go ahead... I'll give them a head start :P |
19:28.49 | RoyK | stupido ergo americano est |
19:28.49 | vaewyn | carpe carp |
19:28.49 | Zeeek | projectile vomiting has become one of those extreme sports they have magazines about |
19:28.49 | RoyK | carpe crap |
19:28.49 | RoyK | .... |
19:28.52 | zoa | carpe canem |
19:28.53 | RoyK | carpe bibum |
19:28.53 | vaewyn | carpe smeg |
19:28.53 | vaewyn | :} |
19:28.53 | zoa | no way |
19:28.53 | Zeeek | bibendous pendulum |
19:28.53 | zoa | veni vidi party bibi dormi bissi trissi parti |
19:28.53 | RoyK | bibendus? |
19:28.57 | RoyK | wtf is that? |
19:28.57 | zoa | bibere = drinking |
19:29.01 | zoa | so maybe a drinker |
19:29.08 | RoyK | bibo == I drink |
19:29.09 | Zeeek | bibendum.com |
19:29.19 | Zeeek | ahem Roy |
19:29.28 | RoyK | ahem? |
19:29.37 | Zeeek | sorry I was ad nauseuming |
19:29.41 | HitTop | i wonder if you can limit the log file size for asterisk? |
19:29.46 | vaewyn | add nausea |
19:29.59 | Zeeek | no there's enough there already |
19:30.09 | Zeeek | shaken, not stirred |
19:30.15 | RoyK | HitTop: there is. 2 gigs |
19:30.34 | RoyK | HitTop: it doesn't, or didn't, open the log file with O_LARGEFILE |
19:30.38 | HitTop | RoyK: thx ^^ do you know where to adjust it? |
19:30.45 | RoyK | I've seen asterisk servers crash because of that |
19:30.45 | Zeeek | HitTop but I'd recommend using a cron job to check it and copy it to be compressed |
19:30.47 | RoyK | rtfs |
19:31.10 | RoyK | Zeeek: you'll need to reload asterisk, perhaps restart, to close the file |
19:31.17 | Zeeek | funny I never do |
19:31.18 | RoyK | unless the file stays open and grows on |
19:31.27 | tzafrir_home | HitTop: isn't the standard logrotte enough? |
19:31.34 | Zeeek | I cp it then cp /dev/null to it |
19:31.52 | RoyK | setup asterisk to log to /dev/null |
19:32.00 | tzafrir_home | It only checks daily, but it can check by size, if you want |
19:32.05 | vaewyn | mv blah.log blah.log.0; cat /dev/null > blah.log :} |
19:32.06 | Secretive | Zeeek, the router we have is a 10/100 8 port VPN router |
19:32.21 | HitTop | tzafrir_home: i actually wants to limit the size to be smaller.. i just got a linksys router to work with asterisk.. but the log saves in ram, it will crazy as time goes by |
19:32.40 | Zeeek | Secretive but now I've been shot down by Manx who pointed out (rightly so) that your problem doesn't appear to be with NAT |
19:32.55 | tzafrir_home | HitTop: log to syslog and use a syslog that has size limitations |
19:33.11 | tzafrir_home | IIRC the syslog that comes with busybox has such an option |
19:33.19 | vaewyn | tzafrir_home: or to another machine you don't care about :P |
19:34.03 | vaewyn | ok... WIP5000 review up... --now with pictures!--- http://www.wwwrogue.com/voip/WIP5000.html |
19:35.14 | Zeeek | for $320 there should be free drinks too |
19:35.22 | Zeeek | just to look at the pictures |
19:35.29 | vaewyn | bwahaha... it really is a kickbutt phone |
19:35.52 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
19:36.18 | HitTop | tzafrir_home: syslog is an application? |
19:36.21 | Zeeek | good think they left the chinese on the buttons - i'd be lost |
19:36.30 | vaewyn | I got 3.75 hours talk yesterday on 1 charge... and it have run flawless except for a glitch after I used the web interface |
19:36.38 | HitTop | tzafrir_home: sorry. im pretty new to linux |
19:36.47 | vaewyn | Zeeek: umm... isn't that japanese? |
19:36.51 | miller7 | anyone around that can help me set up pppd server with asterisk? I have installed zapras and pppd but I'm doing something wrong with the IPs and routing so caller cannot see anything |
19:37.00 | Zeeek | you see how rusty I am? |
19:37.04 | vaewyn | hehehe |
19:37.13 | benno2 | vaewyn: thanks alot for the review ! I was wondering if using non-vlan switches if roaming could be made faster. I tried two SOHO APs and win xp with Xten softphone and it works quite well, max 1sec call outage when roaming |
19:37.19 | HitTop | tzafrir_home: anyways, thx. i'd try syslog |
19:37.55 | vaewyn | benno2: I am guessing I can turn down the MAC cacheing on the switches and get the same speedup... just havn't done it yet |
19:38.04 | Zeeek | so you guys really get off walking around between AP talking on the phone? :) |
19:38.13 | benno2 | vaewyn: or perhaps you can tune the values "start roaming when signal goes below x dB", please make more experiments and keep your page update. I think its helpful for many when considering purchasing the phone |
19:38.22 | bjohnson | can you here me now |
19:38.27 | Zeeek | nope |
19:38.27 | bjohnson | hear |
19:38.35 | vaewyn | Zeeek: when I work in a building with 6 aps to provide full building coverage I kindof have to do that |
19:39.05 | vaewyn | benno2: i will be playing quite a bit so... more to come |
19:39.33 | benno2 | vaewyn: they ripped off linphone ? :) time to let them GPL the phone code :) |
19:39.38 | Zeeek | onlmy because today it has become unacceptable to be "offline" for more than 20 secods |
19:39.50 | bjohnson | vaewyn: <- calls the cute girl in accounting. Uses "testing" as an excuse |
19:39.57 | Zeeek | precisely |
19:40.07 | *** join/#asterisk inspired (mikael@host-81-191-119-90.bluecom.no) |
19:40.14 | vaewyn | Zeeek: ohh don't worry... I ignore calls longer than that :P |
19:40.18 | Zeeek | well, at least it's built in the 3rd world by people happy to be making $100 a year |
19:40.29 | Luke-Jr | How can I determine why my PAP2's registration is being Forbidden? |
19:40.37 | bjohnson | try to change it |
19:40.43 | vaewyn | benno2: not sure where linphone got that chunk of code from yet so... checking on it |
19:40.51 | Zeeek | and then sold to gull^h^h^h efficient office people |
19:40.52 | bjohnson | is it an unlocked PAP2? |
19:41.15 | benno2 | vaewyn: btw are you using open APs , WEP or WPA 802.1x ? |
19:41.15 | Luke-Jr | bjohnson: yes |
19:41.18 | _Raptor_ | can anyone help me finding the problem with my codecs: i have set up asterisk with meetme and oh323 but when i say codec=speex in the oh323.conf it tells me Unknown codec 'speex'. what's going wrong there? |
19:41.22 | TrevorSHarrison | quick question: whats the Vonage sip gateway hostname? |
19:41.28 | Luke-Jr | bjohnson: * is returning Forbidden to its reg tho |
19:41.29 | bjohnson | Luke-Jr: on same lan as *? |
19:41.35 | vaewyn | Zeeek: heh... we only purchased it now because we know in a year+ it will be in the 200$ range |
19:41.37 | __Sparks_ | Helo (again!) I have a PSTN number provided by Sipgate pointing to my Asterisk box, is there a way to make the ringing tone the caller hears like a UK tone rather then a US one? |
19:41.40 | vaewyn | benno2: wide open |
19:41.58 | Zeeek | vaewyn - right! like everything I've ever bought in my life. |
19:42.10 | bjohnson | TrevorSHarrison: lucifer |
19:42.20 | Zeeek | My first laser printer cost $2500 |
19:42.26 | bjohnson | TrevorSHarrison: just kidding |
19:42.32 | TrevorSHarrison | :) |
19:42.36 | vaewyn | Zeeek: hehehe... well we are evaluating for a VERY large bulk buy of phones so... 300$ now versus $$$$$ later is a good trade |
19:42.55 | Zeeek | beter yet the MX80 Epson was $600 and came without a cable! |
19:43.06 | Zeeek | course that was in 1870 |
19:43.10 | benno2 | vaewyn: thks. and stability issues went away with the current firmware ? |
19:43.57 | Zeeek | stability... muhahaha |
19:43.58 | vaewyn | benno2: havn't had a change to load the current yet... stability did come back after I reloaded factory defaults and reset up the phone.... that web interface is sketchy in my book... it really screwed it up |
19:44.10 | *** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
19:44.17 | bjohnson | Luke-Jr: I hear they are similar to SPA 2000 but I don't know what might cause that other than deny and permit entries in sip.conf |
19:44.24 | vaewyn | Zeeek: This thing has my grandstream and snom beat 100/1 on stability |
19:44.43 | vaewyn | which isn't saying much with grandstream :P |
19:44.44 | Zeeek | hey the GS is rock stable. As long as you NEVER update the FW!!! |
19:45.12 | eKo1 | Zeeek: You can throw it like a rock... |
19:45.26 | vaewyn | the WIP5k is on par with my polycom IP500 |
19:45.29 | Zeeek | nah, it's break upon impact |
19:45.43 | Zeeek | it'd |
19:45.55 | eKo1 | I never commented about the impact. |
19:45.58 | vaewyn | heheh... mine would make 1000s of pieces... I put lead sheeting in it so it would quit slidding on the counter |
19:46.12 | Zeeek | ~eKo1cancelwhenbridged=yes |
19:46.29 | eKo1 | heh |
19:46.45 | Zeeek | well folks, have fun I'm off until the morning shift |
19:47.09 | vaewyn | Zeeek: also... you gotta think... 100$ of the price is just for the pretty blue LEDs so... ;P |
19:47.13 | vaewyn | cya |
19:47.19 | Zeeek | I love blue LEDs |
19:47.46 | bjohnson | PTG123: let me know when you're in |
19:47.57 | SafT | hmm, i just put an offer in on 23 SP12+'s |
19:48.09 | SafT | ~8us each, wonder if they will accept |
19:48.10 | bjohnson | SP12? |
19:48.18 | benno2 | do I need to assign a MSN to the zaphfc card ? with the i4l card I had the "msn=12345678" string in modem.conf . can(must) I put it in zaptel.conf (or zapata) sorry , new to zap* stuff :) |
19:48.19 | SafT | cisco/selsius thing |
19:51.05 | Secretive | Okay guys -- so I want something a bit technical. |
19:51.27 | RoyK | celcius? |
19:52.17 | Secretive | I want to be able to dial into my Asterisk through a VOIP DID ... the call will be answered, I will be prompted to enter a passcode. Once the code is entered, if valid, i want to be able to enter a number to dial out to. |
19:52.29 | bjohnson | Secretive: ooooo |
19:52.58 | bjohnson | Secretive: go the the user authentication wiki page in the tips and tricks section |
19:53.26 | bjohnson | you will red about, voicemail, auth by CID, authenticate, and DISA |
19:53.57 | bjohnson | long live the wiki |
19:54.11 | bjohnson | seriously .. anybody doing backups of it? |
19:54.20 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
19:54.30 | dca[laptop] | hi all |
19:54.55 | dca[laptop] | anyone ever notice a delay when doing a 'reload' on asterisk? |
19:55.27 | eKo1 | eh, yeah. |
19:56.03 | dca[laptop] | it's a long enough delay to make you think you need to do another reload but then, wham!, it reloads |
19:56.56 | dca[laptop] | quite around here, is everyone getting on that conf call? |
19:58.33 | Secretive | I get no MoH and this error: Feb 17 14:03:08 NOTICE[27169]: res_musiconhold.c:472 monmp3thread: Request to schedule in the past?!?! |
19:58.40 | ariel_ | dca[laptop], yes it happens allot. But if you do the reload from the manager api it's almost instant. |
19:59.38 | zoa | when is that conference call ? |
19:59.45 | zoa | is it 2pm there already? |
20:00.01 | CMike | hi all |
20:00.11 | *** join/#asterisk HitTop (~Jack@host6614613596.biz.tor.fcibroadband.com) |
20:01.40 | HitTop | HI~^_^ |
20:01.57 | Secretive | Can someone please help me with my Music on HOld issue: Feb 17 14:03:08 NOTICE[27169]: res_musiconhold.c:472 monmp3thread: Request to schedule in the past?!?! |
20:02.16 | junky[work] | i think its cause ya dont have enuf cpu. |
20:02.42 | HitTop | Secretive: do u have mpg321 instaleld? |
20:02.46 | junky[work] | im getting it on an old 200MMX |
20:02.48 | bjohnson | zoa: yes |
20:02.54 | junky[work] | its mpg123 |
20:03.05 | HitTop | ops^^ hehe |
20:03.12 | Meznev | Secretive: I don't know anything about it, but you might want to make sure that all the devices have the correct time set. |
20:03.31 | jero_SFLphone | hi junky |
20:03.37 | Meznev | Those type of errors are usually because either the server's time is set ahead or the client's is set behind |
20:04.24 | benno2 | HitTop: I got the same moh error with mpg321, is this because it has an mpg123 emulation script which does not exactly match the cmdline args and therefore * calls the fake mpg123 with the wrong args resulting in mpg321 exiting ? |
20:04.29 | Secretive | Interesting. |
20:04.33 | harryvv | Anyone here that run debian and have modules installed? Did you get this error when doing /usr/src/asterisk make config ? look at http://pastebin.ca/6036 and tell me what you did to resolv it. |
20:04.35 | Secretive | My mpg123 was renamed to mpg123.old |
20:04.38 | Secretive | So it was mpg123 |
20:04.41 | Secretive | I renamed it and it works now. |
20:04.50 | Secretive | Very choppy though |
20:05.04 | HitTop | wow~ |
20:05.14 | zoa | omf this doesnt sound good :) |
20:05.15 | Secretive | THe CPU is 100% choppy though. |
20:05.21 | RoyK | ~lart zoa for fun |
20:05.36 | zoa | what is causing this ? |
20:05.46 | zoa | BRIAN |
20:05.46 | Secretive | The CPU is 100% idle, but it's very choppy |
20:05.53 | zoa | i cant hear you guys |
20:06.11 | HitTop | Secretive: is ur cpu too slow? (sorry if im wrong) |
20:06.40 | JerJer | do you have the PROPER version of mpg123 ? |
20:06.57 | Secretive | P4 2.8ghz with 512k Cache and 512mb Ram |
20:07.07 | JerJer | do you have a zaptel timing device? |
20:07.09 | HitTop | wow..@@ |
20:07.16 | Secretive | JerJer: Me, no. |
20:07.26 | RoyK | JerJer: you didn't answer that nice e-mail on the -dev list, did you? |
20:07.27 | HitTop | right.. u need to have ztdummy at least |
20:07.45 | JerJer | RoyK: ? |
20:08.16 | *** join/#asterisk kiran (~kiran@203.212.254.27) |
20:08.32 | RoyK | :) |
20:08.33 | HitTop | is it possible to disable logging in asterisk? if so, where do i set it? |
20:08.44 | bjohnson | logger.conf |
20:09.06 | Secretive | JerJer: What is the proper version?' |
20:09.08 | *** join/#asterisk DsrtZrzmra (~DsrtZrzmr@dsl-200-67-75-232.prod-empresarial.com.mx) |
20:09.28 | RoyK | <quote> |
20:09.29 | RoyK | Don't worry Michael. |
20:09.30 | RoyK | Just ignore Jeremy. It's clear that he doesn't understand the difference |
20:09.30 | RoyK | between a codec and a transport protocol. (Which may go some way towards |
20:09.30 | RoyK | explaining some of the problems chan_h323 has ;-) |
20:09.30 | HitTop | bjohnson: so if i don want logging, i write messages => ? (leaving the right side blank? |
20:09.34 | RoyK | </quote> |
20:10.30 | Secretive | [root@localhost asterisk]# mpg123 -v |
20:10.30 | Secretive | High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. |
20:10.30 | Secretive | Version 0.59r (1999/Jun/15). |
20:10.36 | DsrtZrzmra | Can anybody help me in a simple problem in asterisk setup? I just cant "register" using kphone, but i can make calls |
20:11.07 | *** join/#asterisk jayden (~ircatjerr@65.170.43.34) |
20:11.13 | DsrtZrzmra | i get: SIP/2.0 403 Forbidden |
20:11.14 | kiran | hi can any one help me in configuring asterisk CDrs to Mysql Cdrs |
20:11.22 | bjohnson | HitTop: don't know .. try it |
20:11.33 | HitTop | bjohnson: ok . i'll try then tell u^^ |
20:12.01 | Secretive | How to I turn on good debugging for asterisk cli |
20:13.23 | Meznev | As in not ultra vague sortaerrors? :P |
20:13.42 | Secretive | Meznev: As in I want to see when agents log in/out etc |
20:13.48 | Secretive | And is there anyway to see which agents are logged in right now? |
20:13.51 | bjohnson | DsrtZrzmra: sounds like same problem Luke-Jr has |
20:14.38 | bjohnson | show agents ? |
20:15.30 | yashax | simple linux question: ifconfig eth0 192.168.0.10 netmask 255.255.255.0 up (where do I type the gateway?) |
20:15.49 | RoyK | ~lart JerJer |
20:15.56 | *** join/#asterisk ZeroXeal (~zeroxeal@ool-44c166d7.dyn.optonline.net) |
20:16.06 | MEPHiST0 | yashax: route add default gw <gw> |
20:16.14 | PatrickDK | yashax, you don't |
20:16.19 | PatrickDK | interfaces don't have gateways |
20:16.42 | MEPHiST0 | indeed |
20:16.42 | Secretive | Is there anyway to logout agents from the cil |
20:16.48 | yashax | MEPHiST0: default gw is parameter? |
20:16.55 | yashax | or default |
20:16.55 | bjohnson | no |
20:16.57 | bjohnson | text |
20:17.05 | bjohnson | <gw> is the IP |
20:17.07 | MEPHiST0 | yashax: you have to set the def-gw in the default routing table |
20:17.21 | MEPHiST0 | that can be set with route add default gw <ip> |
20:17.26 | MEPHiST0 | simple thing |
20:17.57 | *** join/#asterisk numBone (~numBone@c-24-129-204-233.se.client2.attbi.com) |
20:18.49 | Secretive | I really dont like these all over the place: Feb 17 14:15:02 WARNING[27169]: chan_sip.c:4873 check_auth: Secret is #Phone1.304: |
20:19.09 | DsrtZrzmra | Its easy, y installed asterisk in debian, added an extension (8004), then im using kphone to "register" but it get SIP/2.0 403 Forbidden. I have host=dynamic. |
20:19.12 | yashax | MEPHiST0: Awesome... thanks.... hwo do you set the DNS? |
20:19.35 | MEPHiST0 | in the file /etc/resolv.conf |
20:19.48 | Weezey | how do I dial a pause? |
20:19.55 | yashax | GOT IT!!!!!! Thanks... |
20:20.12 | MEPHiST0 | weezey: atdt nnn,nnn |
20:20.15 | Weezey | thanks |
20:20.18 | MEPHiST0 | where , causes 500ms of delay |
20:21.50 | SexyKen | How do I get rid of all of those messages like: Feb 17 14:25:57 WARNING[27169]: chan_sip.c:1168 find_peer: Looking for SIP: 1.303 |
20:21.50 | HitTop | bjohnson: yes. its right, but log file must exist.. could its not a big deal |
20:22.05 | TrevorSHarrison | does anyone have an example vonage sip config they could paste? |
20:22.07 | MEPHiST0 | touch rules |
20:25.30 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
20:26.17 | kiran | hi can any one support on installing asterisk addons? |
20:27.10 | *** join/#asterisk buddah (~hnic@67.110.253.129) |
20:27.22 | buddah | how do i turn on progress tones for inbound? |
20:27.39 | buddah | not getting tones when we call any of our phones |
20:27.40 | jero_SFLphone | using ringing ? |
20:27.48 | buddah | yeah, ringing |
20:27.56 | buddah | need it, but dont have it |
20:28.16 | jero_SFLphone | what happens if your extensions script calls Ringing() at a given time ? |
20:28.20 | jero_SFLphone | you should get ringtones |
20:28.27 | buddah | ? |
20:28.43 | buddah | dont have Ringing() in anywhere |
20:29.07 | jero_SFLphone | and you want to hear ringtones telling the user that the call is being established ? |
20:29.09 | Weezey | all of my calls appear from ${CALLERIDNAME}|asterisk instead of ${CALLERIDNAME}|${CALLERIDNUM} to my SPA-841. Any thoughts? |
20:29.11 | buddah | yes |
20:29.25 | buddah | well its when someone from the outside calls in |
20:29.25 | jero_SFLphone | buddah, then try to add Ringing(). What kind of channels is it ? |
20:29.27 | buddah | they get no ringtones |
20:29.30 | jero_SFLphone | oh |
20:29.53 | buddah | so if i call from my cell to the phone behind sip, i hear no ringing |
20:30.02 | buddah | its just quiet then either they pick up or it does voicemail |
20:30.06 | jero_SFLphone | buddah: at what time do they not get any ringtone? Before the line is answered, or after |
20:30.14 | buddah | before its answered |
20:30.18 | jero_SFLphone | brb |
20:30.31 | *** part/#asterisk jayden (~ircatjerr@65.170.43.34) |
20:32.51 | _Raptor_ | [1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323 v1.12.2, PWlib v1.5.2 |
20:32.51 | _Raptor_ | Feb 17 21:31:14 ERROR[21843]: chan_oh323.c:4653 load_module: Failed to insert capability 22. |
20:33.04 | _Raptor_ | can anyone tell me what this means? |
20:33.21 | _Raptor_ | codec=SPEEXN8K in oh323.conf |
20:33.32 | bjohnson | buddah: do you answer and/or do ivr and then dial the internal extension |
20:33.58 | buddah | i have no idea what ivr is |
20:34.03 | buddah | and no |
20:34.08 | buddah | i just dial the number |
20:34.09 | buddah | thats it |
20:34.11 | bjohnson | looks like it specifying the speex codec .. but I don't know what N8K is |
20:34.53 | bjohnson | buddah: so your extensions.conf has an extension s,1,dial(<the internal phone to be called>) ? |
20:35.13 | ^Fenris | is there a way to set up asterisk to send faxes via a print queue? |
20:35.31 | bjohnson | ^Fenris: I believe so |
20:35.35 | _Raptor_ | bjohnson: yes i want speex, but with codec=speex it says unknown codec |
20:35.41 | bjohnson | ^Fenris: the easiest answer is don't use * for faxes |
20:35.53 | bjohnson | _Raptor_: no idea .. I don't use h323 |
20:35.54 | ^Fenris | bjohnson: heh |
20:35.57 | _Raptor_ | so i looked into the src and found SPEEXN8K |
20:36.07 | ^Fenris | bjohnson: i have it working pretty well now for recieving them |
20:36.07 | _Raptor_ | but now this happens |
20:36.33 | ^Fenris | bjohnson: if I setup a different machine to send them there could be line collisions (one picks up the phone while the other is sending) |
20:37.17 | ^Fenris | reading about Asterisk spandsp, maybe that does what I want... |
20:38.43 | jero_SFLphone | buddah |
20:39.29 | bjohnson | ^Fenris: you can buy a $20 hardware peice at radioshack to prevent a line in use from being interupted |
20:40.47 | ^Fenris | oh |
20:41.30 | greg_work | ${EXTEN:1} chops of the first character, right? (read: the first character is 0, not 1, ?) |
20:41.34 | buddah | yes |
20:41.43 | greg_work | ok |
20:44.11 | jero_SFLphone | anyone tried SFLphone ? |
20:45.44 | *** join/#asterisk r0d3nt|m (ctxwvp@perverseengineering.org) |
20:46.25 | greg_work | | doesn't have any meaning in dial patterns, does it? |
20:47.47 | *** join/#asterisk adjacent (~scott@office.bftwave.com) |
20:48.05 | *** join/#asterisk yurpls (~yurplsl@65.114.15.70) |
20:48.49 | *** part/#asterisk didz_ (didz_@200.218.192.52) |
20:49.01 | Luke-Jr | bjohnson: I can register KPhone fine... just not PAP2 |
20:49.22 | brettnem | hey all, I once saw a high capacilty ISDN PRI / VoIP gateway.. carrier grade hardened box.. anyone know who it may have ben? |
20:49.32 | brettnem | +e |
20:49.45 | harryvv | anyone here by chance running debian and zap? have a make config issue with asterisk that needs resolving |
20:51.04 | *** join/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
20:52.21 | *** join/#asterisk guugmember (~nachoramo@168.234.226.39) |
20:52.35 | DsrtZrzmra | im running asterisk on debian |
20:52.46 | Darwin35 | I am also |
20:52.54 | Darwin35 | new build first time |
20:52.59 | guugmember | hello guys I am making an presentation of how Asterisk can work in an Avaya environment, any pictures or suggestions? |
20:53.26 | guugmember | I have a reunion tomorrow with the Avaya Distributor of Central America |
20:53.53 | harryvv | DsrtZrzmra and Darwin35 did both of you encounter problems when doing a /usr/src/asterisk make config and if so what did you do to resolve it. ? |
20:54.35 | Darwin35 | I had no problems but I have aasterisk 1.0.5 |
20:54.48 | *** join/#asterisk Jeroen (~jeroen@084-246-048-082.PN.nl) |
20:55.27 | *** join/#asterisk Banter (Banter@209.119.214.81) |
20:55.34 | *** part/#asterisk Jeroen (~jeroen@084-246-048-082.PN.nl) |
20:55.51 | terrapen | hey, someone with FWD |
20:55.53 | terrapen | call 616306 |
20:56.03 | brettnem | hmm WARNINGs and sometimes bridging calls gives me squeals in phone calls.. any ideas?? :-/ |
20:56.09 | terrapen | for an automated test |
20:56.54 | DsrtZrzmra | no, i used apt-get install asterisk |
20:57.05 | brettnem | any ideas??? |
20:57.12 | brettnem | SIP to SIP calls |
20:57.20 | harryvv | DS, I see I did the CVS way mabey that was the problem. |
20:57.51 | *** part/#asterisk Uther_P (~uther_p@66.180.120.83) |
20:58.03 | harryvv | DsrtZrzmra and it loads the zap modiles on a system reboot? |
20:58.41 | Darwin35 | I did not I built from src |
20:59.01 | Darwin35 | and ztdummy is not building |
20:59.04 | Darwin35 | grrr |
21:00.04 | Secretive | How do I reload extensions without restarting asterisk |
21:00.12 | Darwin35 | reload |
21:00.32 | Darwin35 | but it will not reload till all the channels are clear |
21:00.33 | junky[work] | that makes sense huh? :) |
21:00.53 | vaewyn | Secretive: extensions reload |
21:00.53 | junky[work] | no, it will reload even if channels are actives. |
21:01.01 | yurpls | Anyone have a TDM400P with FXO? |
21:01.02 | Darwin35 | they changed it |
21:01.07 | klicTel | extensions reload |
21:01.29 | Secretive | What is verbose mean |
21:01.30 | johnnyb | I'm having trouble getting my transfer to work w/ Grandstream phones. I've got asterisk transfers working using the "tT" option and "#" on the phone, but I wanted to do attended transfers with the phone. |
21:01.51 | johnnyb | However, every time I try transferring it merely hangs up the line on transfer. |
21:01.54 | *** join/#asterisk rvhi (~rv@mail.o-matrix.org) |
21:02.13 | guugmember | where can I find Asterisk pictures to make a ppt presentation |
21:02.25 | *** join/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com) |
21:02.28 | yurpls | Pics of what of *? |
21:03.00 | *** join/#asterisk fearnor (~alex@66.250.55.66) |
21:03.28 | fearnor | !summon atacomm |
21:03.42 | loud | why dont you google:// asterisk filetype:ppt ? |
21:04.23 | fearnor | atacomm has very nice onhold music |
21:04.29 | vaewyn | hehehe... I always just take screenshots of code and use that as the background :} |
21:04.34 | fearnor | like 30 minutes and didn't repeat itself |
21:04.42 | fearnor | speaking of |
21:04.56 | Darwin35 | there are no pics |
21:04.57 | fearnor | anyone else has a102u cards? ;) |
21:05.10 | yurpls | whats a a102u card? |
21:05.18 | fearnor | sangoma a102. |
21:05.22 | yurpls | T1? |
21:05.26 | fearnor | yes. |
21:05.53 | yurpls | I have one of there wan cards. currently can't get the new digium card T1/E1 to work. |
21:06.05 | fearnor | you probably have digiumme |
21:06.16 | fearnor | i want to try sangoma |
21:06.32 | yurpls | I have a sangoma 2 port WAN card. Worked for years. |
21:06.49 | yurpls | Reason you are trying sangoma for *? |
21:06.58 | eKo1 | 2 port WAN? What carrier system? |
21:07.06 | Banter | I'm trying to install and i get a c compiler error?? |
21:07.18 | fearnor | cause digium cards are a pain in the freaking neck |
21:07.27 | fearnor | its 2005, about time cards have something more than 32 byte buffers. |
21:07.35 | fearnor | too interrupt sensitive. |
21:07.42 | yurpls | fearnor: Thanks for telling me after I just bought two of them and I now have problems. |
21:07.57 | fearnor | you problems are probably not the problems i'm having. |
21:08.00 | fearnor | tehy are fine |
21:08.21 | CoaxD | fearnor: C'mon, 16 byte FIFOs should be ALL YOU NEED! *rotfl* |
21:08.34 | yurpls | The 4 port T1s work great. TDM400P and the new E1/T1 is killing me. |
21:08.35 | fearnor | coaxd: 16450 for LIFE |
21:08.42 | CoaxD | fearnor: I wrote the first driver for the Exar 17C158 octal PCIset.. |
21:08.46 | fearnor | coax: unless you are a fan of 8250 |
21:08.50 | CoaxD | fearnor: Wasn't THAT a blast.. |
21:09.11 | vaewyn | outtolunc: check groklaw... is a 9 day ultimatum to get their docs up to date |
21:09.23 | ^Fenris | Ok, I'm convinced not to use * for outgoing faxes, so my next question is, can Asterisk co-exist with Hylafax? both will be using the same modem |
21:09.29 | CoaxD | fearnor: The main coolness about that chip is that you could do 32 bit fifo reads/writes.. So, you got 4x as much work done in 1 read or write as you could with a 16550 |
21:09.56 | outtolunc | aww |
21:09.57 | fearnor | fuck this noise |
21:10.05 | CoaxD | fearnor: heh. sux |
21:11.11 | *** join/#asterisk sneak (~sneak@64.220.234.21.ptr.us.xo.net) |
21:11.21 | yurpls | fearnor: What probs u having with sangoma? |
21:11.37 | fearnor | well, first, i gotta find someone who can guarantee they can ship it to me TODAY |
21:11.46 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
21:12.07 | Banter | will asterisk run on mandrake? |
21:12.12 | CoaxD | Banter: Um, yes. |
21:12.17 | CoaxD | Banter: If it is linux, it'll run |
21:12.18 | Banter | cool thank you |
21:12.24 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
21:12.35 | CoaxD | Welcome :) |
21:12.48 | Darwin35 | unical is for uniden sip phones right |
21:13.00 | Darwin35 | ? |
21:13.08 | Darwin35 | not seen it before |
21:13.16 | yurpls | I think sangoma sell direct. Thats where I get WAN cards. |
21:13.45 | Darwin35 | asterisk runs on fbsd also |
21:13.47 | vaewyn | holy cow do polycoms have a ton of crud in their XML... wowzers... you can config them to do about anything |
21:14.09 | harryvv | well, so far now one has a answer to my question concerning why /usr/src/asterisk make config generates a missing init.asterik error for debian. |
21:14.13 | ^Fenris | Darwin35: but there are no drivers for zaptel hardware for fbsd |
21:14.27 | Darwin35 | yes there is |
21:14.39 | ^Fenris | what's the name of the driver? |
21:14.41 | Darwin35 | I have 4 tdm40 cards |
21:14.46 | Darwin35 | zaptel |
21:14.55 | vaewyn | but bsd ain't supported so good luck |
21:14.56 | Darwin35 | in /usr/ports/misc/zaptel |
21:15.16 | Darwin35 | once its up its up |
21:15.18 | vaewyn | (officially supported I should say) |
21:15.25 | Darwin35 | and there is a support mail list |
21:15.36 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
21:15.45 | ^Fenris | Darwin35: hrmmm, guess I need to upgrade my fbsd box, what version are you using? |
21:15.45 | fearnor | ok well, atacom isn't getting this order |
21:15.53 | Darwin35 | and libpri is also there |
21:16.06 | vaewyn | fearnor: whatcha buying? |
21:16.13 | Darwin35 | and asterisk is in /usr/ports/net/asterisk |
21:16.20 | Darwin35 | its ver 1.0.3 |
21:16.30 | Darwin35 | the port should be updated soon |
21:16.32 | ^Fenris | of Freebsd |
21:16.47 | Darwin35 | it works on fbsd 4.10 and 5.x |
21:17.14 | ^Fenris | I'm running freebsd 5.0-Release, so maybe I just need to update my ports |
21:17.58 | ^Fenris | I have Asterisk configured with a modem for receiving faxes, can I use the same modem to send faxes? or does Asterisk have a lock on it |
21:18.14 | Darwin35 | yes |
21:18.27 | Darwin35 | you can use asterisk to fax |
21:18.27 | Mavvie | Darwin35: if you sent patches for the 1.0.5, port, sobomax doesn't need to do it on his own. |
21:18.49 | ^Fenris | Darwin35: so * doesn't lock the modem, so to speak |
21:18.57 | Darwin35 | he needs to update the port in the ports tree |
21:19.12 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
21:19.14 | Mavvie | Darwin35: if you send patches to him for the port, he doesn't need to do all the dirty work himself. |
21:19.16 | ^Fenris | was thinking of using Hylafax to send faxes |
21:19.41 | Darwin35 | I have sent him patches and nothing has happened and got no responce from him |
21:20.12 | Mavvie | did you use send-pr ? |
21:20.14 | Darwin35 | I use hylafax and a ext 56 k modem |
21:20.31 | ^Fenris | and the modem is also used by Asterisk? |
21:20.42 | Darwin35 | I use spandsp and the faxx add on from opencall.com |
21:20.47 | Darwin35 | no |
21:20.52 | Mavvie | Darwin35: did you use send-pr ? |
21:20.55 | Darwin35 | I can use it to dial my office |
21:21.17 | Darwin35 | Mavvie i believe so |
21:21.26 | Mavvie | Darwin35: what are the PR numbers? |
21:21.34 | Darwin35 | but I also emailed him directly about another issue |
21:21.46 | Darwin35 | I will have to search |
21:21.49 | *** join/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca) |
21:21.54 | Darwin35 | I am on a diff system right npw |
21:22.00 | Darwin35 | now |
21:22.07 | Darwin35 | not on the dev system |
21:22.32 | Weezey | i don't understand zapata.conf. What does it do? I have my stuff going out my SIP ATA, so do I need a zapata.conf? |
21:22.46 | vaewyn | Weezey: nope |
21:22.49 | wizhippo | has anyone here tried * and Sphinx? what version of sphinx should i use to try the eagi test script? |
21:22.49 | eKo1 | Weezey: No. |
21:22.55 | Darwin35 | zapata is used by the digium cards |
21:23.01 | JerJer | you only need zapata.conf if you have a Zaptel device |
21:23.03 | Mavvie | Darwin35: what is the name you would have used in your email address? |
21:23.07 | Darwin35 | for configuration |
21:23.09 | Weezey | gotcha, thanks. |
21:23.14 | Darwin35 | bsdtech |
21:23.21 | guugmember | can I call ENUM the DNS for SIP? |
21:23.24 | Weezey | (which also means, it can't fix my problems) |
21:23.26 | Mavvie | guugmember: yes |
21:23.42 | eKo1 | guugmember: of course |
21:24.00 | Mavvie | Darwin35: can't find any PRs from that person. |
21:24.20 | Darwin35 | ok i got festival 1.95 working with asterisk |
21:24.37 | Mavvie | guugmember: within limits of course, only 1 NAPTR record per person (unless you got my patch) |
21:25.09 | Mavvie | Darwin35: any difficult patches or just one new Lisp command. |
21:25.32 | moonwick | yaay zaptel |
21:25.50 | Darwin35 | needs to be updated from ver 1.0.3 to 1.0.5 and I will have to find all the changes int the existing patches that have to be pulled out |
21:26.10 | Darwin35 | and we need to get zaptel updated soon also |
21:26.19 | Darwin35 | I use the current cvs |
21:26.38 | Darwin35 | from the fbsd zaptel project |
21:26.55 | rvhi | hi, in voicemail, can i customize the email header for each context? |
21:27.56 | Weezey | aha! How come my calls are coming from "asterisk" or an unknown number when ${CALLERIDNUM} is defined before the call gets passed to my SPA-841? http://www.mail-archive.com/asterisk-users@lists.digium.com/msg75788.html |
21:28.32 | greg_work | is it bad to do exten=>_123,1,Macro(...) as opposed to exten=>123,1,Macro(...) ? |
21:28.48 | Darwin35 | I will work on it again tonight and get a list of patches |
21:28.58 | Darwin35 | and try to pr them again |
21:29.37 | *** part/#asterisk Banter (Banter@209.119.214.81) |
21:30.03 | benno2 | I installed a hfc-s with zaphfc on * stable (fedora core 3, kernel 2.6.10). inbound calls work perfectly but outbound calls (from a budgetone via SIP account) give me a 503 error |
21:30.36 | Weezey | I removed my SETCIDNUM command and the caller ID comes in perfectly to my set. |
21:31.21 | benno2 | any idea ? |
21:32.20 | *** part/#asterisk afrosheen (afrosheen@txprotoa8.august.net) |
21:32.25 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || Dev Conf 2PM CST FEB 24th -> IAX2/guest@66.250.68.194/996 |
21:33.20 | guugmember | what is the true value of ENUM? just not to remmeber extensions? and just remember emials for example? |
21:35.47 | eKo1 | guugmember: ENUM == DNS for VoIP devices. |
21:36.17 | harryvv | eko1 are you involved with the development of * |
21:37.15 | johnnyb | For incoming calls, we are using land lines. However, for outgoing calls we wanted to use a VoIP provider. What do you all recommend? |
21:37.28 | eKo1 | Well, I discover bugs if that is considered helping. |
21:37.43 | eKo1 | Haven't really bothered to get into the code. |
21:37.53 | eKo1 | Although I really should. |
21:40.01 | harryvv | eKo1 okay I do have a issue and it has to deal with /usr/src/asterisk make config. It cannot find a file and think that make config was setup for redhat not debian. My issue is to get the modules loaded with make config so debian can load the moduled automaticly apon a reboot then asterisk can start. |
21:40.12 | harryvv | I have to load them manually to make it work. |
21:41.12 | eKo1 | eh, just add them to modules.conf |
21:42.06 | eKo1 | Oh wait, I see what you mean. |
21:42.13 | harryvv | http://pastebin.ca/6039 |
21:42.24 | eKo1 | You're talking about zap modules? |
21:42.27 | harryvv | yes |
21:42.34 | eKo1 | Or wcfxo or whatever they're called. |
21:43.25 | guugmember | eKo1, so what is DUNDI all about? |
21:43.44 | harryvv | wcfxs |
21:43.49 | harryvv | err |
21:43.50 | harryvv | yea |
21:43.54 | bjohnson | greg_work: I don't think there is really a difference unless you're dealing with LARGE call volumes |
21:44.22 | harryvv | ek01, do you think my issue should be brought up in #asterisk_dev? |
21:46.19 | eKo1 | harryvv: http://pastebin.ca/6040 <--- That's what I do. |
21:46.22 | *** part/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca) |
21:46.54 | eKo1 | guugmember: With DUNDI, there is no central server incharge of lookups. |
21:47.45 | harryvv | ek, which distro are you running? |
21:48.06 | eKo1 | FC2. |
21:48.20 | harryvv | k. people in debian say /etc/modules.conf |
21:48.38 | harryvv | But then..the info is there and uncommented and it should be loading |
21:48.56 | harryvv | ek, let me get a copy of it and send the pastbin to you |
21:49.05 | eKo1 | Well, you could do it that way too. Whatever tickles your pickle. |
21:49.26 | *** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net) |
21:50.21 | harryvv | http://pastebin.ca/6041 |
21:50.27 | harryvv | take a look at that |
21:50.49 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
21:51.34 | eKo1 | Is there an /etc/modprobe.conf in Debian? |
21:52.16 | harryvv | yes |
21:52.19 | harryvv | err |
21:52.22 | harryvv | yes |
21:53.16 | eKo1 | Do you see anything with wcfxs and ztcfg? |
21:54.24 | harryvv | yes |
21:54.29 | harryvv | but thay are uncommented |
21:55.02 | *** join/#asterisk Sesq (~Sesq@gate.us.cyberscience.com) |
21:55.35 | eKo1 | OK, get rid of the stuff in modules.conf and uncomment the stuff in modprobe.conf. |
21:56.10 | vaewyn | ok... wth if it is all using ulaw can I hear my polycom on SIP/IAX clients... but not on the zaptel? |
21:56.25 | tzanger | why are we using ulaw anyway? |
21:56.33 | tzanger | is it ot save the server? |
21:56.36 | vaewyn | um... why not? |
21:56.41 | *** join/#asterisk __Sparks_ (ringding@bb-195-172-52-15.ukonline.co.uk) |
21:56.42 | harryvv | okay |
21:56.42 | tzanger | because it's bandwidth-wasteful |
21:56.47 | tzanger | gsm is light on proc and bandwidth |
21:56.49 | Sesq | quick question... anyone know if the voicepulse connect service is allowing more than 4 simultaneous lines yet? |
21:56.59 | eKo1 | harryvv: umm...did you uncomment an alias line in your modules.conf |
21:57.00 | vaewyn | tzanger: phones don't all support gsm... |
21:57.04 | vaewyn | or ilbc... or... |
21:57.05 | tzanger | vaewyn: true enough |
21:57.06 | bjohnson | tzanger: what g standard is gsm? g723? |
21:57.09 | __Sparks_ | Hi, I am getting wuite a few "app_dial.c:749 dial_exec: Unable to create channel of type 'SIP" messages from Asterisk - is this bad!? |
21:57.10 | tzanger | but that's what your server's for :-) |
21:57.30 | vaewyn | tzanger: 2 items... gigabit ethernet... and great call clarity :P |
21:57.43 | tzanger | vaewyn: yeah well I don't have a gigabit link to the internet |
21:57.44 | tzanger | :-) |
21:57.48 | tzanger | bjohnson: no idea |
21:57.52 | tzanger | oh |
21:58.02 | tzanger | wtf am I doing in here, I was supposed to be in asterisk-dev |
21:58.03 | vaewyn | tzanger: this is internal to campus... off campus gets ilbc/gsm/g729 |
21:58.03 | tzanger | heh |
21:58.08 | vaewyn | hahaha |
21:58.11 | junky[work] | is there any reason, when im generating .call, i cant go more then 143 active channel(s) ? |
21:59.03 | terrapen | what do y'all think of Speex? |
21:59.12 | tzanger | haven't used it |
21:59.27 | vaewyn | terrapen: ok.. but not supported enough... and slightly crunchy on the CPU |
21:59.29 | terrapen | Keep it in zer ghetto or train to Auschwitz? |
21:59.31 | bjohnson | is there a way to upgrade a user's groups in linux without logging out and back in again? |
21:59.34 | tzanger | crunchy |
21:59.34 | tzanger | heh |
21:59.40 | tzanger | crunchier than ilbc? |
21:59.47 | terrapen | (excuse the Ali G reference) |
21:59.49 | vaewyn | yep |
22:00.24 | terrapen | so it is out of zer balloon. |
22:00.50 | *** join/#asterisk SuperMMan (~graphic@d209-89-191-155.abhsia.telus.net) |
22:00.55 | vaewyn | If someone can get a hardware encoder out there it will be a mut point |
22:01.01 | vaewyn | err... moot |
22:01.03 | SuperMMan | anyone know where i can get a copy of the psql cdr table? |
22:01.16 | terrapen | i want to find the best all-around codec. |
22:01.39 | harryvv | eko1 well found it in /etc/modprobe.d/zaptel its been uncommented by default. |
22:01.47 | eKo1 | SuperMMan: wiki |
22:01.52 | _Raptor_ | <PROTECTED> |
22:02.25 | SuperMMan | eKo1, ok thanx |
22:02.34 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
22:03.11 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
22:04.49 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-142166197096.nb.aliant.net) |
22:04.59 | ManxPower | ~docs |
22:05.01 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
22:05.17 | ManxPower | _Raptor_, You need a [default] in extensions.conf |
22:05.36 | __Sparks_ | Please help me here!! - I keep getting "app_dial.c:749 dial_exec: Unable to create channel of type 'SIP" " And I don't know why! - I have a Grandstream BudgeTone if thet helps! |
22:06.14 | _Raptor_ | ManxPower: i have too |
22:06.18 | _Raptor_ | [default] |
22:06.18 | _Raptor_ | exten => 1,1,Meetme(100) |
22:06.53 | *** join/#asterisk Capouch (501@12.176.248.4) |
22:07.16 | eKo1 | __Sparks_: That doesn't help. |
22:07.26 | _Raptor_ | terrapen: ? |
22:07.34 | terrapen | the wiki? |
22:07.36 | terrapen | ~wiki |
22:07.38 | DsrtZrzmra | I have Forbidden problems using kphone, i've tryed using password and passwordless account, kphone wonk register |
22:07.41 | _Raptor_ | yes |
22:07.46 | _Raptor_ | terrapen: you mean me? |
22:07.55 | terrapen | no, Santa Claus :) |
22:07.59 | _Raptor_ | :-/ |
22:08.01 | _Raptor_ | sry |
22:08.01 | DsrtZrzmra | anybody have the same error? |
22:08.06 | terrapen | yes, you. :) |
22:08.14 | __Sparks_ | eK01: What would! |
22:08.22 | jero_SFLphone | did you try sflphone ? |
22:08.45 | harryvv | eko1 alias wcfxo zaptel ztcfg ? |
22:08.56 | _Raptor_ | ok, let me ask one simple question: what is the best way to realize h323 conferencing with speex codec? |
22:09.30 | harryvv | well that did not work |
22:10.17 | JerJer | _Raptor_: realize the fact that you no longer need H.323 |
22:10.28 | terrapen | hhahaha |
22:10.29 | __Sparks_ | eKo1: It seem to happen when I call the Sipgate PSTN number, it reports it as soon as the SIP phone rings |
22:10.40 | _Raptor_ | JerJer: why not? |
22:10.46 | _Raptor_ | JerJer: sip? |
22:11.16 | *** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net) |
22:11.27 | *** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com) |
22:12.06 | Darwin35 | zaptel head sucks |
22:12.40 | tzanger | Darwin35: what's wrong with it |
22:12.42 | Trionnis | can't say I've ever gotten head from a zaptel.... wouldn't know |
22:12.45 | Trionnis | ;) |
22:12.53 | tzanger | Trionnis: heh |
22:13.09 | Darwin35 | it wont compile on debian |
22:13.14 | jero_SFLphone | Im having lots of troubles with zaptel cards |
22:13.34 | tzanger | Trionnis: put it this way -- you need a big woman for decent zaptel head because there's just not a lot of brains in the hardware |
22:13.42 | tzanger | jero_SFLphone: like what |
22:13.45 | Trionnis | lol |
22:13.51 | Trionnis | well played :) |
22:14.09 | harryvv | Darwin35 what is your zaptel alias setup as in /etc/modprobe.d/zaptel |
22:14.15 | jero_SFLphone | tzanger, like echo (not the most important) and callerid |
22:14.24 | tzanger | jero_SFLphone: where are yo ulocated |
22:14.30 | jero_SFLphone | in montreal |
22:14.50 | tzanger | north american callerid works just fine |
22:14.58 | *** join/#asterisk menger (~menger@static-88.243.240.220.dsl.comindico.com.au) |
22:15.00 | jero_SFLphone | yes, in 5% of cases for me |
22:15.02 | tzanger | remember that with POTS you need to wait AT LEAST two rings to get it since it is sent between the 1st and 2nd ring |
22:15.12 | jero_SFLphone | ah |
22:15.31 | harryvv | tzanger get what? i have post on this end |
22:15.34 | Trionnis | never understood that |
22:15.39 | harryvv | wait two rings |
22:15.45 | ManxPower | __Sparks_, If the IP address of the phone does not show when you do a "sip show peers" then the phone is not registering with Asterisk and you cannot call it. |
22:15.46 | Darwin35 | thereis no modprobe.d on debian |
22:15.46 | Trionnis | using subband anyway, why not just send it prior to the first |
22:15.49 | tzanger | and echo -- well there's the fxotune if you're using wctdm and zapmonitor (I think) to help you with that too |
22:15.58 | jero_SFLphone | 2 rings is how much delay ? |
22:16.00 | tzanger | harryvv: that's just how it's sent here in Canada |
22:16.02 | jero_SFLphone | 8 secs ? |
22:16.09 | tzanger | jero_SFLphone: well no, the system should do it for you |
22:16.16 | tzanger | you can try waiting 4 seconds |
22:16.23 | tzanger | 2 seconds on, 4 seconds off is the ring cadence |
22:16.32 | tzanger | so 2 second ring, that 4 second pause is when the callerid is sent |
22:16.36 | ManxPower | DsrtZrzmra, I don't see your paste of the error message you are getting. |
22:16.43 | jero_SFLphone | i most often get checksum errors |
22:16.58 | tzanger | Trionnis: I didn't write the telco docs :-) |
22:17.07 | Trionnis | nor did I ;) |
22:17.10 | tzanger | jero_SFLphone: use ztmontior and get your gains set up right first :-) |
22:17.17 | harryvv | tzan, sorry I was not following the both of you. Mabey that has to do with some kind of issue where my ivm picks up the pots line at two rings or four rings. |
22:17.19 | Trionnis | just muttering :) |
22:17.22 | jero_SFLphone | tzanger, my gains are well tuned |
22:17.22 | tzanger | it could be that the audio is just to quiet or loud to reliably decode the FSK datastream |
22:17.24 | harryvv | Darwin35 take a closer look its there. |
22:17.38 | __Sparks_ | ManxPower, It is showing, and I can make & recieve calls fine, it just irritates me thowing an error when it shouldn't! |
22:17.40 | jero_SFLphone | tzanger, I tuned my gains with a milliwatt at the other end |
22:18.10 | tzanger | jero_SFLphone: that's great for your txgain, but how did you tune your rxgain (which is what the dsp is using to hear the far end) |
22:18.22 | _Raptor_ | ok, one more simple question: what is the best way for conferencing with speex (with linux and windows clients) |
22:18.33 | jero_SFLphone | tzanger, I called a milliwatt test number |
22:18.43 | tzanger | ahh okay |
22:18.43 | jero_SFLphone | tzanger, with my asterisk |
22:18.53 | tzanger | it should be pretty good then |
22:19.00 | tzanger | I am not sure what to tell you |
22:19.05 | jero_SFLphone | I think so, echo is not a problem anymore |
22:19.10 | Trionnis | anyone point me toward info about the output audio bitrate for ulaw? |
22:19.15 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
22:19.31 | Trionnis | trying to use it with FWD and icecast streaming, but it's hosing up the voice audio |
22:19.32 | tzanger | I thought output bitrate was 64kbps |
22:19.35 | Trionnis | moh is fine |
22:19.41 | Trionnis | I don't know :( |
22:19.42 | *** join/#asterisk search_learn2005 (~Miranda@209.68.139.150) |
22:19.54 | Trionnis | and can't seem to find it in the docs or the wiki |
22:19.57 | jero_SFLphone | tzanger, I even patched chan_zap to have a separate gain setting for the CallerID stuff |
22:20.04 | Trionnis | yes, I'm lame... flog me at will |
22:20.09 | Trionnis | :) |
22:20.19 | *** join/#asterisk DaLion (~Miranda@Toronto-HSE-ppp3884470.sympatico.ca) |
22:20.23 | tzanger | jero_SFLphone: you certainly sound like you know what you're doing -- you might want to post something in -dev and see what kicks up |
22:20.27 | tzanger | or join #asterisk-dev and ask there |
22:20.40 | tzanger | both places seem very strange for responses... sometimes rihgt awy, other times as if you were talking to a brick wall :) |
22:20.55 | *** join/#asterisk peted20 (~chatzilla@24-113-67-25.wavecable.com) |
22:20.57 | outtolunc | huh? |
22:21.02 | *** join/#asterisk Luke-Jr (~luke-jr@207.192.219.246) |
22:21.03 | jero_SFLphone | :) I'll try |
22:21.08 | tzanger | anyway |
22:21.09 | tzanger | gotta get kids |
22:21.11 | tzanger | later all |
22:21.16 | jero_SFLphone | bye |
22:21.17 | RoyK | <PROTECTED> |
22:21.34 | search_learn2005 | still trying to decide to use the analog way or to buy budgetstream phones: 7 fxos and ~50 fxs, existing 10/100 network, existing analog phones |
22:21.50 | search_learn2005 | any suggesstions |
22:22.37 | *** join/#asterisk genie (~test@gate.us.cyberscience.com) |
22:23.25 | *** join/#asterisk didz_ (~omg@200.218.193.30) |
22:23.53 | *** join/#asterisk riksta (~rick@81-178-248-194.dsl.pipex.com) |
22:24.38 | bjohnson | search_learn2005: existing cat5 to all proposed phone locations? |
22:24.56 | DaLion | anyone can tell me if DB odbc version of iaxfriends and sipfriends can hold like g279;ulaw;;gsm << - not i have a blank entry.. will that use #1 else #2 else #4 ? or they need to be in order |
22:25.24 | *** join/#asterisk welby (~welby@solas.plus.com) |
22:25.27 | *** join/#asterisk anti (russ@anti.developer.gentoo) |
22:25.41 | bjohnson | search_learn2005: I suggest some of both. voip phones where cat5 wiring is already run and fxs/analog phones where 2 pair is already run. Service new locations with whatever seems best for that location |
22:25.48 | anti | hmm is there anyway to get rid of the like 2 ring delay from when I call in a zap line and have it dial a SIP channel right away? |
22:26.13 | bjohnson | anti: do you have a big wait()? |
22:26.29 | DaLion | lol |
22:26.36 | anti | bjohnson: nope no wait, just immediate Dial(SIP... |
22:26.38 | *** part/#asterisk Sesq (~Sesq@gate.us.cyberscience.com) |
22:26.49 | search_learn2005 | bjhnson: every room has CAT5 avilable, what would you say for the echo issue? Which alternative has less echo trouble? |
22:27.14 | greg_work | search_learn2005: might also want to consider sipura 841's |
22:27.15 | anti | bjohnson: I see "Starting simple switch" immediately, during the very first ring, but then it sits there, doing nothing then finally Executes the Dial.. |
22:29.39 | *** part/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com) |
22:30.50 | johnnyb | What do you all think of the iLBC codec? I'm trying to decide on a codec to use for my new budgetones, and iLBC seems to be a good choice for a low-bitrate codec. However, G728 looked good, too. What do you all think? |
22:30.57 | johnnyb | I thought the G711 sounded good, but using 64kbit/s sounded awfully hoggish. |
22:31.12 | jero_SFLphone | anti: do you use callerid ? |
22:31.24 | johnnyb | I didn't notice any degredation w/ iLBC, but thought someone with more experience could offer some advice. |
22:31.53 | bjohnson | search_learn2005: they potentially all have echo problems to be solved. I doubt that is a decision maker for you. |
22:32.35 | bjohnson | search_learn2005: even if cat 5 in all rooms, it is suitable for additional bandwidth required for multiple phones? Is it available in suitable locations within the rooms? Maximize what you've already got |
22:33.19 | guugmember | alguien que hable español aca? |
22:33.24 | bjohnson | anti: I don't know .. haven't seen that myself with Sipuras or X100P that I have |
22:33.26 | ionix | anyone has a way for asterisk to pickup the name of the caller from the ANI ? Like from a RBOC database or such ? |
22:33.36 | search_learn2005 | bjohnson: if I get the VOIP phones for every room then I don' t have to get adtrans 750t which makes installation easier. Then I will only need a card with 7 fxos and that's it. Am I right? |
22:34.15 | bjohnson | ionix: I am told you can maintain your own db and go lookups based on CIDnum or subscribe to dbs provided by others |
22:35.08 | DsrtZrzmra | yo hablo español |
22:35.18 | DsrtZrzmra | pero no se mucho, apenas estoy empezando |
22:35.41 | bjohnson | search_learn2005: yes and no. Yes you would only need the fxo's for the lines. No it probably won't make the installation any easier .. just different. Also, have you looked for partial PRIs instead of 7 individual lines? |
22:35.59 | nestAr | PRI > POTS |
22:36.06 | nestAr | i <3 my pri's |
22:38.33 | bjohnson | search_learn2005: a switch from POTS to PRI would take some planning and organization. Although the monthly cost would be similar (if a partial PRI is even available), there are features available to PRIs that makes managing them and adopting future changes much easier |
22:38.43 | __Sparks_ | Another question! - If i make calls via Sipgate tp a PSTN number, the quality is very good, no echo - if however I route the call via my x100p card, theere is a lot of echo on the line for a while (it clear up eventually, but to start with it's really bad - what can be done about this!? |
22:39.10 | bjohnson | play with the x100p settings |
22:39.39 | nestAr | we pay like $425 for pri's.. one pots line at business rates is ~$90 a month.. |
22:39.57 | dsmouse | gah |
22:40.03 | harryvv | bjohnson who has a x100 |
22:40.05 | dsmouse | I can get vonage to work, but only with ulaw |
22:40.09 | dsmouse | which sucks |
22:40.18 | search_learn2005 | bjohnson: I wouldn't have a say at the 7 fxos at least at the moment. the school already has the 7 fxos and they had them for years. So, if I want to go with the adtrans method, how many 750s wil I need, and will I need the most basic model of 750, because there are 3 versions. Are there any configurations on the adtrans or will everything still be done on the asterisk server. That's why I am a li |
22:40.18 | search_learn2005 | ttle bit scared with the adtrans method, there is just not do many newbie friendly documents out there about it. And, it is an expensive device. |
22:40.22 | __Sparks_ | bjohnson, I would if I know what to play with! - can you give me a clue :-) |
22:41.24 | bjohnson | harryvv: __Sparks_ does |
22:41.37 | __Sparks_ | is that a bad thing to have then!?! |
22:41.42 | *** join/#asterisk denon (denon@synapse.subneural.net) |
22:41.42 | *** mode/#asterisk [+o denon] by ChanServ |
22:41.43 | harryvv | __Sparks_ you also run debian? |
22:41.48 | __Sparks_ | yea |
22:41.58 | harryvv | good! |
22:42.03 | bjohnson | __Sparks_: no idea. I fighting an echo with SPA fxo myself that might be related to gain or line impedence |
22:42.14 | harryvv | does your load the modules apon reboot and start *? |
22:42.53 | harryvv | bj, more then likly line impedence |
22:42.58 | Jlau515 | in a extensions.conf, a exten => s,1,anwser, would that anwser all incoming calls from a zaptel channel |
22:43.00 | DaLion | bkw you alone on conf now ;) |
22:43.01 | bjohnson | search_learn2005: check ebay .. used adit600 with 24 fxs often go for < $300 I'm told |
22:43.22 | harryvv | bj, good example is when my wife picks up the analog phone on the same line that goes into the asterisk box it upsets the impedence and echo goes up. |
22:43.28 | bjohnson | search_learn2005: tzanger is usually helpful with chan bank questions |
22:44.12 | bjohnson | search_learn2005: generally, a chan bank system could lower your up front cost to about $30 USD per phone compared to about $80 each for cheap new ones |
22:44.40 | Jlau515 | i'm trying to play a main menu prompt |
22:45.18 | bjohnson | search_learn2005: also some chan banks can provide fxo ports .. so 3 chan banks and a quad T1 card from digium would allow you to do anything you want |
22:46.22 | search_learn2005 | bjohnson: but how do I hook up a channel bank to a pci card no idea, and I am a visual learner. So any website where I see a channel bank hooked up to a card via T1? |
22:46.24 | bjohnson | greg_work: talk to tzanger .. I think they are usually about $800 CDN / mo for 23 channels |
22:47.06 | bjohnson | search_learn2005: probably .. but I don't know where. Try the usual suspects |
22:47.12 | bjohnson | ~usual_suspects |
22:47.15 | bjohnson | damn |
22:47.24 | bjohnson | wiki, mailing list archives, google |
22:47.26 | *** join/#asterisk Luke-Jr (~luke-jr@207.192.219.246) |
22:47.33 | bjohnson | time for me to go |
22:48.36 | trym | I have installed spandsp to have asterisk receive faxes. When a fax call is made to asterisk, asterisk starts whining about RFC3389. I also notice that the volume spandsp/asterisk is communicating with varies.. which is not normal for a fax session. Any suggestions? |
22:50.06 | Darwin35 | fricking cvs |
22:50.25 | *** part/#asterisk search_learn2005 (~Miranda@209.68.139.150) |
22:50.41 | Darwin35 | i need zaptel for timing |
22:50.49 | Darwin35 | and for when the new card gets here |
22:51.23 | *** join/#asterisk Nukemizer (~Nuke@65.103.231.133) |
22:53.19 | Nugget | I was never able to get meetme working in freebsd. |
22:53.25 | Nugget | I finally just gave up |
22:55.26 | *** join/#asterisk anthm (anthm@208.254.19.131) |
22:55.26 | *** mode/#asterisk [+o anthm] by ChanServ |
22:58.16 | *** join/#asterisk kippi (fc@cpc4-hatf3-6-0-cust243.lutn.cable.ntl.com) |
23:00.00 | Jlau515 | can someone help me with an ivr issue? |
23:00.12 | kippi | hey |
23:01.17 | *** join/#asterisk Legend (~legend@24.244.142.133) |
23:01.46 | Jlau515 | i have this in my extensions.conf |
23:01.52 | Jlau515 | exten => s,2,Answer ; Answer the line |
23:01.53 | Jlau515 | exten => s,5,BackGround(demo-congrats) ; Play a congratulatory message |
23:02.29 | Jlau515 | i want to know should that be playing if an incoming call came from my zaptel channel |
23:04.39 | ManxPower | Jlau515, Not with those priorities it won't. |
23:04.43 | ManxPower | Hell, it won't work at all. |
23:04.57 | ManxPower | priorities must be consecutive (normally) |
23:05.06 | *** join/#asterisk jarnaud (~jarnaud@65.217.47.11) |
23:05.08 | jsolares | where's s,1 ; s,3 ; s,4? |
23:05.11 | ManxPower | starting at 1 |
23:05.30 | jarnaud | Someone has played with auto dialout? |
23:09.01 | Jlau515 | i actually have prioritys for 1 2 3 4 5 |
23:09.19 | Jlau515 | was wondering if using 's' means that will answer all incoming calls |
23:09.54 | Jlau515 | i didnt want to spamm the channel my include all my other s,1 ; s,2 ; ... |
23:10.18 | Nukemizer | Diguim support is having me update my cvs to get my T1 card to work - i did not get instructions on what I need to do so I was hoping someone could verify what I have found "asterisk-update.sh update" |
23:10.20 | Jlau515 | dialing out works for me, but dialing in does not so far |
23:12.37 | jsolares | Jlau515: it all depends if that's the context you have defined for all your incoming channels |
23:13.18 | Jlau515 | its in my [from-sip] content |
23:13.24 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
23:14.25 | jsolares | Jlau515: what exactly do you want to do? |
23:14.31 | *** join/#asterisk ennuyeux72 (~ennuyeux7@62.53.79.208) |
23:14.55 | Jlau515 | just play the demo prompt when a call comes in from my zapata channel |
23:15.21 | jsolares | what context did yo put in your zapata.conf? |
23:15.35 | Nix | hey guys |
23:15.47 | Jlau515 | i just looked i had default context |
23:15.54 | Jlau515 | it needs to be from-sip right? |
23:16.06 | tzanger | do not use the default context |
23:16.10 | *** join/#asterisk micrisc2 (~micrisc@mail.techhelpresources.com) |
23:16.19 | *** join/#asterisk doctor_za_ljubav (bkwyg@195.252.86.177) |
23:16.19 | tzanger | I'd put zap in something like 'from-zap', from sip in 'from-sip' etc until you understand more |
23:16.20 | jsolares | mhnoyes: or you should make an inbound one in extensions with the promp including what you want |
23:16.28 | jsolares | *could |
23:16.36 | Jlau515 | k |
23:16.36 | jsolares | and prompt |
23:16.37 | jsolares | bleh |
23:16.39 | Jlau515 | i'll try that |
23:16.42 | doctor_za_ljubav | hello |
23:16.52 | jsolares | it'll work ;) |
23:17.02 | jsolares | hi |
23:17.06 | ManxPower | 6 hours on cpnference calls today. *sob* |
23:17.12 | *** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
23:17.16 | doctor_za_ljubav | can someone tell me where can i download a digium T400P linux driver..... |
23:17.28 | Jlau515 | hmm still dont work |
23:17.37 | Jlau515 | i see the external number in zap show channel |
23:17.40 | micrisc2 | anyone using spandsp fax addon? i am having problems compileing, /usr/include/tiffiop.h:38:18: port.h: No such file or directory i get this but the file is there |
23:17.57 | *** join/#asterisk yurpls (~yurplsl@65.114.15.70) |
23:18.02 | Jlau515 | but the demo does not answer |
23:18.28 | Jlau515 | i tested the demo by asigning a extension for demo playback and that plays |
23:18.55 | *** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com) |
23:19.05 | ManxPower | Jlau515, Did you fix your priorities? |
23:19.24 | jsolares | he had priorities, just did not want to spam the channel with all of them |
23:20.07 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
23:21.22 | *** join/#asterisk lilneon (~tj_r3@cuscon4828.tstt.net.tt) |
23:21.29 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
23:21.33 | lilneon | hi guys an dgood night everyone |
23:21.47 | Nugget | "linux is great for supporting things like $7 ebay tape drives. This is of critical importance for those users who have only $7 worth of data." <-- heh |
23:22.08 | ennuyeux72 | anyone know if asterisk deals proprely with a pending response to an reinvite request |
23:22.34 | mikegrb | Nugget: :D |
23:25.06 | *** join/#asterisk ProAtWork (~procyan@host157.reign.radel.com) |
23:25.19 | ProAtWork | hi, anyone know of a source for cisco 7960 ringtones? |
23:25.43 | ProAtWork | also, if you have problems converting one of these things to SIP I just spent a bit of time figuring it out |
23:25.49 | RoyK | wtf is a 9760? |
23:26.09 | ProAtWork | 7960 is the big cisco phone with the display |
23:29.37 | Jlau515 | i am getting the follow error from the cli |
23:29.40 | Jlau515 | <PROTECTED> |
23:29.45 | anti | So I have a SIP phone and a zap channel, someone calls in the zap channel, rings my sip phone, I pick up. I'm trying to come up with a way I can enable recording then, something where I can press #[something] and have it announce on the line this call is about to be recorded (comply with laws) and then run Record() .. but sadly, I can't figure out how. |
23:29.53 | Jlau515 | 4152484088 is the number i am dialing from |
23:30.01 | Jlau515 | anybody know how to fix this issue |
23:31.10 | Nugget | ProAtWork: I've got a bundle of them. the same bundle that I presume everyone in here has. |
23:31.31 | yurpls | Anyone help with a TDM400P card? |
23:31.38 | Nugget | there's that one guy's config linked from the wiki that we all seem to find and use |
23:33.52 | ProAtWork | Nugget hmmm... the voip-info wiki? |
23:34.56 | Nugget | of course |
23:35.07 | ProAtWork | I don't find any links to ringtones on that page |
23:35.14 | ProAtWork | I found a link to a bunch of .gsm files |
23:35.28 | ProAtWork | these won't work on a 7960 according to the docs |
23:35.36 | Nugget | it's not ringtones specifically, just that guy's config. |
23:35.43 | *** join/#asterisk Ayano (~erik_leee@209.143.187.254) |
23:35.48 | ProAtWork | oh.. I can configure the phone.. not a problem |
23:35.57 | ProAtWork | just looking for .pcm files |
23:36.00 | Nugget | jesus. |
23:36.01 | Nugget | yes. |
23:36.02 | Nugget | I know. |
23:36.13 | Nugget | that sample config includes a whole shitload of ringtones. |
23:36.28 | Nugget | google for ringlist.dat |
23:36.50 | ProAtWork | ahhh.. in the tar file |
23:36.57 | ProAtWork | ok I'm catching on now |
23:36.58 | Nugget | right-o |
23:37.03 | Nugget | sorry if I wasn't clear. |
23:37.18 | ProAtWork | thanks for pounding me in the head enough times with the clue-by-4 |
23:37.45 | rvhi | hi, for voicemail, is it possible to customize the header of notification email for each context |
23:37.46 | rvhi | ? |
23:38.13 | _Raptor_ | cya |
23:38.30 | *** join/#asterisk nextime (~nextime@ns0.nexlab.net) |
23:40.44 | Ayano | What are the best choice phone for asterisk, I've worked with budgetone, and some lower ciscos, what do you guys recomend? |
23:41.44 | jsolares | i like the avaya 4602 IP |
23:42.01 | Ayano | Do you have to upgrade firmware to use it? |
23:42.14 | d-tech | uniden looks like a good phone for the price |
23:42.19 | jsolares | if it has h323 firmware then yes |
23:42.28 | jsolares | if it has sip firmware then no |
23:42.32 | *** join/#asterisk eye69 (magnus@ipv6.upcore.net) |
23:42.34 | kippi | hey, where is the best place to read up on how to create SIP users on asterisk (on a freebsd system) |
23:42.47 | jsolares | the wiki |
23:42.48 | lilneon | hey guys.. anyone know where or can help me with a script to open or forward he relevant ports to allow voip(SIP or IAX) on a proxy with Squids behind that proxy? |
23:43.17 | jsolares | lilneon: does it have iptables for firewall? |
23:43.22 | Nugget | I'm fond of my cisco 7960, but I dunno if they're worth the cost. |
23:43.27 | Ayano | Kippi: The asterisk wiki has the easy context to set up sip users |
23:43.40 | Nugget | I can say that you absolutely do not want that zyxel/pulver wireless thing. |
23:43.42 | lilneon | jsolares: yes.. |
23:44.03 | Ayano | I've used them, sometimes the firmware is a pain. |
23:44.08 | lilneon | jsolares: need a script to by pass or open/forward the relevant ports for voip to work |
23:44.30 | jsolares | iptables -t nat -A PREROUTING -s 0/0 -d <public ip> -d udp -port 5060 -j DNAT --to <internal ip> |
23:44.38 | jsolares | or something like that, let me check |
23:45.05 | kippi | ok, i'll have a read of it and try and get it working |
23:45.09 | jsolares | -p udp |
23:45.17 | jsolares | not -d udp |
23:45.25 | Ayano | kippi: hold on, I'll give you a link. |
23:45.27 | jsolares | port 5060 is for SIP |
23:45.32 | doctor_za_ljubav | anyone to help me install t400p card? |
23:45.43 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
23:46.13 | Ayano | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
23:46.41 | kippi | Ayano: thanks |
23:46.48 | lilneon | jsolares: would it matter if they have a dynamic ip? |
23:47.27 | jsolares | the internal or the external? |
23:47.37 | jsolares | and yes it does matter, whenever they change the rule is worthless |
23:47.38 | __Sparks_ | I have a problem calling X-Lite extentions from my BudgeTone - as soon as I talk, X-Lite crashes - if I call the other way (X-Lite to BudgeTone) there isn't a problem - any ideas why!? |
23:47.57 | lilneon | jsolarres: ok cool.. thnx.. |
23:48.19 | jsolares | read up on iptables and portforwarding, google is your friend ;) |
23:48.36 | lilneon | jsolares:way ahead of u |
23:49.05 | jsolares | i'm glad i have my * server with a public ip, no hassle pita configuration for that |
23:50.10 | Ayano | jsolares: I set it up using this once.... http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions |
23:50.37 | jsolares | yeah, that's a good read :) |
23:50.56 | Ayano | It causes an echo if you don't have a good router. |
23:51.17 | jsolares | so that's where the echo comes from :o |
23:52.09 | Ayano | If the router doesn't translate fast enough it makes it a pain to get rid of the echo. |
23:52.38 | jsolares | oh well, atleast the callee doesnt hear it |
23:53.11 | jsolares | and since i'm the caller it matters not |
23:54.03 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
23:54.13 | Ayano | True, I found that it is sparatic. Sometimes good, sometimes bad. Sometimes all it takes is to hang up and call again and it goes away...... Strange. I took out the router and had no problems. |
23:54.42 | Ayano | Thats why I figured it was a cheap nat router problem. |
23:54.46 | jsolares | too bad i cant take out the router |
23:54.48 | *** join/#asterisk alerios (~alerios@201.245.167.88) |
23:54.55 | jsolares | maybe i should upgrade the firmware on my wrt54g |
23:55.35 | Ayano | Read the spec on the firmware upgrade and it will tell you what it changes. |
23:55.46 | Ayano | Be carefull, sometimes it wipes your settings. |
23:55.59 | Ayano | Make a copy of your configs on the router first. |
23:56.16 | lilneon | jsolares:i have to run this on their server right? |
23:56.21 | jsolares | i dont really have setting that i could wipe out, i do like sveasoft's firmwares, the alchemy one has qos and support for l7 |
23:56.27 | jsolares | lilneon: yep |
23:56.35 | Nukemizer | can somebody confirm for me that to update my cvs for zaptel - lipri and asterisk i would do a "cvs checkout zaptel" " cvs checkout libpri" " cvs checkout asterisk" ? |
23:56.45 | lilneon | ok coo,l |
23:57.11 | didz_ | cvs checkout asterisk zaptel libpri |
23:57.16 | Ayano | lilneon: did you look at this? http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions |
23:57.40 | lilneon | ayano:yeah.. believe it or not i do take tghe advice on here seriously |
23:57.46 | Nukemizer | lilneon, thanks |
23:57.50 | jsolares | you would be case 3 and 6 |
23:57.57 | Nukemizer | didz_ thanks :) |
23:58.04 | lilneon | also got this from dca[laptop]:http://www.voip-info.org/tiki-index.php?page=Asterisk%20firewall%20rules |
23:58.15 | lilneon | thnx to teliax support.. :) |
23:59.27 | jsolares | that one is for openning up a firewall, but not for portforwarding |