| 00:00.01 | Qwell | $3 :D |
| 00:00.05 | Corydon76-home | Qwell: I never want to hear you call me sick again. :-P |
| 00:00.51 | Strom_C | Corydon76-home: are you going to be in huntsville this week? |
| 00:01.03 | Corydon76-home | Strom_C: wasn't planning on it |
| 00:01.09 | Strom_C | alright |
| 00:01.16 | Corydon76-home | Why, what's in HSV? |
| 00:01.20 | Strom_C | i'm catching a redeye tonight |
| 00:01.25 | Corydon76-home | Ouch |
| 00:01.35 | Strom_C | well it was either that or catch a 7 am flight |
| 00:01.51 | Corydon76-home | Wednesday is payday anyway. I need to be in the office for that |
| 00:02.07 | Corydon76-home | and I'm a little short until then |
| 00:02.16 | Strom_C | and the chances of me waking up in time for the 7 am flight are close to zero :) |
| 00:02.30 | Corydon76-home | Strom_C: oh, just stay up until then. ;-) |
| 00:02.43 | Strom_C | heh |
| 00:03.18 | Corydon76-home | Wait until you get to be an old fart like me. You'll start thinking 5 am is a good time to be up |
| 00:03.29 | Qwell | Strom_C: when do you get in? |
| 00:03.39 | Strom_C | 9amish |
| 00:03.45 | Qwell | hah |
| 00:03.51 | Qwell | I leave tomorrow at 9ish :p |
| 00:03.59 | Strom_C | maybe i'll see you at the airport |
| 00:04.04 | Qwell | likely |
| 00:04.15 | Qwell | still got my cell #? |
| 00:06.15 | Supaplex | how do I get the callerid in 1.0.7? (yea, it's old, but I'll upgrade when debian etch goes stable late this year) |
| 00:06.18 | Strom_C | phone list for the win |
| 00:06.37 | Strom_C | Supaplex: um, why not just download the latest release version of asterisk and compile? |
| 00:06.54 | Supaplex | may as well *sigh* |
| 00:06.55 | Qwell | touche |
| 00:07.05 | *** join/#asterisk JunK-Y (n=junky@modemcable198.14-83-70.mc.videotron.ca) |
| 00:09.07 | *** join/#asterisk enzo (n=enzo@143-023.proxy.saarctextile.com) |
| 00:09.09 | enzo | hi |
| 00:09.38 | enzo | i try to re-install asterisk on another server (the first is dead) |
| 00:10.08 | enzo | i want to have wcfxo, zaptel, wcfxs drivers, they're not in zaptel package in debian ? |
| 00:10.19 | Strom_C | wcfxs? |
| 00:10.22 | Strom_C | 1.0.7? |
| 00:10.29 | Strom_C | is it 2004 again or am I imagining things? |
| 00:11.30 | enzo | yes i use 1.0.7 |
| 00:11.38 | Supaplex | enzo: testing++ |
| 00:11.45 | enzo | i use debian sarge |
| 00:11.54 | JunK-Y | wctdm *blink* *blink* |
| 00:11.56 | Supaplex | well, keep it or move on. |
| 00:12.41 | Strom_C | oh christ |
| 00:12.44 | Strom_C | www.asterisk.org |
| 00:13.01 | Strom_C | download, compile, install, ?????, profit |
| 00:13.31 | enzo | i don't understand, i get zaptel-source or zaptel from asterisk.org, and when i do make, i get a lot of errors |
| 00:13.39 | enzo | any clue? |
| 00:13.51 | Strom_C | do you have the kernel headers installed? |
| 00:13.56 | enzo | yes |
| 00:14.23 | Supaplex | mind read some vague error? what? |
| 00:14.43 | Strom_C | enzo: pastebin your error |
| 00:15.13 | enzo | http://pastebin.ca/246226 |
| 00:15.13 | enzo | beginning of the errors |
| 00:15.42 | Strom_C | I don't know french |
| 00:15.47 | enzo | i try to compile zaptel 1.2.11 |
| 00:16.42 | JunK-Y | enzo: its like ur kernel headers isnt correct. |
| 00:17.18 | enzo | i've installed 2.4.27 kernel headers, not correcT? |
| 00:17.51 | JunK-Y | uname -a ? |
| 00:18.06 | enzo | 2.4.26-1-386 |
| 00:18.17 | JunK-Y | u need these kernel headers, not 2.4.27 |
| 00:18.27 | enzo | i don't have it in debian |
| 00:18.27 | JunK-Y | cause rror: erreur de syntaxe before "ssize_t" is pretty clear :) |
| 00:18.47 | JunK-Y | use a kernel that you have the exact header too. |
| 00:19.06 | enzo | by the way, even if i have asterisk 1.0, i can install the last stable zaptel right ? |
| 00:19.53 | JunK-Y | take kernel-image-2.4.27-2 |
| 00:20.00 | JunK-Y | u can have headers for that image. |
| 00:20.29 | JunK-Y | # apt-cache search 2.4.27-2 |
| 00:20.47 | JunK-Y | u should go for 2.6, in my opinion. |
| 00:21.27 | enzo | i can also |
| 00:22.30 | Supaplex | Asterisk 1.2.10 is good enuf :) |
| 00:22.47 | enzo | yes but backward compatible? |
| 00:23.19 | JunK-Y | enzo: try installing the kernel i mentionned, u should be able to install latest zaptel with that. |
| 00:23.32 | enzo | the 2.4.27 you mean JunK-Y ? |
| 00:23.40 | Supaplex | enzo: is what backward compatible to what? |
| 00:23.55 | JunK-Y | 2.4.27-2 ya |
| 00:24.01 | enzo | i have agi scripts with asterisk1, i don"t want to tune it to support asterisk1.2 |
| 00:24.21 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 00:24.34 | enzo | i've installed 2.4-27-3 JunK-Y |
| 00:25.21 | Supaplex | enzo: looking back, if you're just looking to reinstall to replace what you had running on sarge, consider using apt-file to your advantage. |
| 00:25.37 | enzo | i reboot, bbl |
| 00:25.43 | JunK-Y | enzo: both image and header 2.4.27-3 ?? |
| 00:25.52 | Supaplex | lol |
| 00:26.06 | JunK-Y | blah |
| 00:26.13 | Supaplex | his issue :) |
| 00:27.07 | Supaplex | the upgrade works. nothing like a 10 line dialplan to test... |
| 00:28.26 | *** join/#asterisk |Serge| (n=tyutyu@cpe-72-178-201-233.satx.res.rr.com) |
| 00:28.44 | |Serge| | hi. i just want to ask if VMware wont support my x100p |
| 00:29.11 | |Serge| | i have a windows xp |
| 00:30.08 | Supaplex | that's a question to ask vmware. I doubt they abstract it. |
| 00:31.26 | *** join/#asterisk enzo (n=enzo@143-023.proxy.saarctextile.com) |
| 00:31.28 | enzo | re |
| 00:31.33 | JunK-Y | enzo: both image and header 2.4.27-3 ?? |
| 00:31.42 | enzo | i've rebooted my computer instead of the serveur, so tired... |
| 00:31.45 | enzo | yes JunK-Y |
| 00:31.50 | JunK-Y | so you should be fine. |
| 00:31.56 | enzo | athena:~# uname -a |
| 00:31.56 | enzo | Linux athena 2.4.27-3-386 #1 Thu Sep 14 08:44:58 UTC 2006 i686 GNU/Linux |
| 00:32.20 | enzo | rhaaa lot of errors again... |
| 00:32.46 | JunK-Y | same ones? |
| 00:33.40 | enzo | i don't know how to check the exact version of the headers |
| 00:33.50 | enzo | in aptitude, it seems to be in fact 2.4.27-10s... |
| 00:34.09 | JunK-Y | enzo: delete ur deb package and make sure you just have the -3? |
| 00:34.18 | enzo | athena:/usr/src# dpkg -l |grep kernel-source |
| 00:34.18 | enzo | ii kernel-source- 2.4.27-10sarge Linux kernel source for version 2.4.27 with |
| 00:34.36 | enzo | pff impossible to have the exact number |
| 00:34.59 | enzo | i've had the same problem on my laptop while compiling nvidia driver, so i've upgraded to the very last 2.6, and it was ok |
| 00:35.02 | JunK-Y | the headers. |
| 00:38.00 | *** join/#asterisk viLeR (i=1000@200.26.142.90) |
| 00:39.22 | *** join/#asterisk adamowitz (n=adamowit@ip68-14-27-224.ri.ri.cox.net) |
| 00:40.40 | bobloblian | hi, I am looking for some opinons on hardware... there are 7 networked computers and 7 rj11 jacks |
| 00:40.47 | enzo | asterisk 1.2 is more stable than asterisk 1.0 ? |
| 00:41.14 | bobloblian | what are the advantages to just using analog phones instead of redoing all the cabling for ip phones? |
| 00:41.27 | bobloblian | or disadvantages, as the case may be? |
| 00:49.33 | enzo | JunK-Y: i've installed kernel2.6 |
| 00:49.58 | JunK-Y | with correct headers? |
| 00:50.01 | enzo | but when i compile i get errors, the first being: grep: /usr/src/linux/include/linux/autoconf.h: unknown file |
| 00:50.09 | enzo | yes correct headers JunK-Y |
| 00:50.19 | JunK-Y | do u have autoconf installed? |
| 00:50.47 | enzo | but after, files are missing again |
| 00:51.09 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-92.sd.sd.cox.net) |
| 00:51.10 | enzo | well, gonna sleep, and i'll install the last kernel from tgz, will be better than debian package... |
| 00:51.25 | enzo | that's the way i've done the first time when i've installed asterisk1.0 |
| 00:51.46 | JunK-Y | i did few times, without problems. |
| 00:51.54 | JunK-Y | (and with packages) |
| 00:52.17 | enzo | JunK-Y: do you know where i can read the features between asterisk1.0 ans 1.2 ? |
| 00:52.19 | Supaplex | I love it when you try to fix more than what's broken. |
| 00:52.45 | JunK-Y | see when the release of 1.2 occured. |
| 00:52.49 | JunK-Y | a long time ago :) |
| 00:55.48 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 00:55.52 | enzo | changes are tremendous between asterisk 1.0 and 1.4 ? |
| 00:56.08 | JunK-Y | what means tremendous? :) |
| 00:56.17 | enzo | big new features? |
| 00:56.24 | enzo | more stability ? |
| 00:56.25 | JunK-Y | ohh yeah baby :) |
| 00:56.34 | JunK-Y | not about stability at this point. |
| 00:56.37 | enzo | changelog somewhere? |
| 00:56.37 | topping | hi gang can someone provide a pointer on setting CID for an fxs line? |
| 00:56.51 | Supaplex | diff -urN old/ new/ | wc -l = lots |
| 00:56.55 | JunK-Y | but like Supaplex said, u work A LOT more then ya should just for ur porlbme. |
| 00:57.28 | Supaplex | topping: show functions |
| 00:58.01 | enzo | good night |
| 00:58.16 | enzo | gonna install from scratch tomorrow (hope it'll work) |
| 00:58.22 | JunK-Y | enzo: see ya, good luch. |
| 00:58.30 | enzo | i need it :) |
| 00:58.32 | *** part/#asterisk enzo (n=enzo@143-023.proxy.saarctextile.com) |
| 00:58.33 | JunK-Y | s/luch/luck/ |
| 01:01.24 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 01:01.24 | linagee | anyone here use voicepulse? :(I |
| 01:01.24 | Supaplex | any piticular reason some of the leading for Say would get clipped? Is playing 1-2s of nothing a standard work around after answering? |
| 01:02.02 | JunK-Y | Supaplex: just use Answer(300); where 300 is the number of ms of blank after answering. |
| 01:02.24 | *** join/#asterisk adamowitz (n=adamowit@ip68-14-27-224.ri.ri.cox.net) |
| 01:03.37 | linagee | is voicepulse inbound DIDs not working for anybody? :( |
| 01:05.19 | *** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net) |
| 01:07.17 | topping | Supaplex: isn't it a zapata.conf thing? |
| 01:08.39 | Supaplex | I'm not sure. My calls come in on the AST? I forget. it's from another asterisk box. |
| 01:09.00 | topping | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf.sample |
| 01:09.30 | topping | I'm trying to get my fxs set up so it has the correct callerid so ${CALLERIDNUM} can be used elsewhere in the dialplan |
| 01:10.03 | Supaplex | ohhh I thought you were answering my question :) |
| 01:10.09 | Supaplex | what release of * you got? |
| 01:10.29 | topping | hehe 1.2.8 |
| 01:10.57 | Supaplex | you can use CDR or CALLERID. For whatever reason CALLERID always returns blank for me. |
| 01:12.53 | topping | hmm, so if i want to have the outbound dialplan always use one of those, how do I get the fxs ports configured to push the correct CID? |
| 01:13.47 | Strom_C | topping: set it up in zapata.conf |
| 01:13.55 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-10.mtco.com) |
| 01:13.56 | topping | with callerid= ? |
| 01:14.04 | topping | did that. no luck |
| 01:14.11 | Strom_C | also, in 1.2.x and greater, you should be using ${CALLERID(num)} instead of ${CALLERIDNUM}) |
| 01:14.18 | *** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net) |
| 01:14.22 | Strom_C | pastebin your zapata.conf |
| 01:14.27 | topping | ah cool maybe that's an issue |
| 01:15.45 | topping | hmm, guess not |
| 01:15.48 | Strom_C | pastebin your zapata.conf |
| 01:17.03 | topping | http://rafb.net/paste/results/gD8xSR76.html |
| 01:17.41 | Strom_C | your callerid should read "Name"<3115552368> |
| 01:17.49 | Strom_C | no spaces, no parentheses, no dashes |
| 01:17.58 | Strom_C | except for spaces in the name, of course |
| 01:18.07 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 01:18.14 | Strom_C | also, fxs ports use fxo signaling |
| 01:18.42 | Strom_C | so unless you have your signaling backwards, you're assigning your caller ID to your POTS lines at this point |
| 01:18.50 | topping | do i have them backwards? haha |
| 01:19.04 | Strom_C | well, are ports 1-4 your stations or are they your trunks |
| 01:19.22 | topping | yah i think i have the cid in the wrong place |
| 01:20.00 | topping | bingo |
| 01:20.02 | topping | thanks! |
| 01:22.20 | Supaplex | :) |
| 01:23.28 | Strom_C | you're welcome |
| 01:24.07 | topping | i'm getting this app pretty well dialed in. i manage an apartment building that sits almost on top of an eight-lane interstate |
| 01:24.27 | topping | there's a huge banner on the freeway, people are constantly calling the number for information about leasing |
| 01:25.07 | topping | i managed to connect the door boxes through asterisk using ARA and postgres |
| 01:25.26 | topping | so then connected that number on the freeway to people that wanted to get out of their leases early |
| 01:25.33 | *** join/#asterisk remmo (n=chatzill@smack.isp.net.au) |
| 01:32.59 | *** join/#asterisk neoalex (n=neoalex@user-12ldt1n.cable.mindspring.com) |
| 01:33.14 | neoalex | hi, I have a 1700 cisco router with two fxs ports |
| 01:33.21 | neoalex | how can I make it connect to my asterisk |
| 01:33.26 | neoalex | through SIP |
| 01:35.14 | topping | neoalex: it's probably similar to the cisco ata setup |
| 01:35.46 | topping | you need to configure sip endpoints in sip.conf on the asterisk end and configure the cisco to connect to them |
| 01:37.06 | neoalex | It's the first time I'm touching a cisco router, or anything cisco for that matter :) |
| 01:37.23 | topping | it's an adventure! :) |
| 01:37.31 | neoalex | the asterisk side, I can handle, the cisco side gets me worried |
| 01:37.36 | neoalex | yeah, I can tell :P |
| 01:37.40 | topping | nah don't fret it |
| 01:37.58 | topping | treat the cisco like an ata-500 in your mind |
| 01:38.04 | topping | it's just an ata |
| 01:38.06 | neoalex | I don |
| 01:38.22 | neoalex | checking my mind, don't have any reference to ata-500 :) |
| 01:38.39 | Supaplex | hummm |
| 01:38.41 | topping | it's the cisco ata |
| 01:38.52 | topping | old skool stuff vonage used to ship out |
| 01:39.07 | Qwell | It's alive! buahahahaha |
| 01:39.08 | neoalex | told you, first time I've ever touched anything cisco |
| 01:39.51 | neoalex | not to mention the guys on cisco don |
| 01:40.04 | neoalex | don't seem to be very helpful :) |
| 01:40.18 | Strom_C | Qwell: it works? :) |
| 01:40.21 | Supaplex | they want you to pony up for cisco call manager |
| 01:40.39 | neoalex | of course :))) |
| 01:41.15 | Qwell | indeed |
| 01:41.28 | Qwell | ringer sucks...but that's tweakable |
| 01:41.30 | neoalex | I got that thing too, I'll give it a try, I have a felling I'm going to be disappointed though |
| 01:41.37 | Qwell | it's a "dull" ring |
| 01:43.03 | *** join/#asterisk |Serge| (n=tyutyu@cpe-72-178-201-233.satx.res.rr.com) |
| 01:45.02 | *** join/#asterisk adamowitz (n=adamowit@ip68-14-27-224.ri.ri.cox.net) |
| 01:45.35 | Supaplex | these lime flavor tootsie rools are ... odd |
| 01:46.52 | aydiosmio | yeah |
| 01:46.59 | aydiosmio | I found whole bags of the vanilla ones |
| 01:47.00 | aydiosmio | I was in heaven |
| 01:47.14 | Supaplex | ouu now that's a tempting idea |
| 01:47.25 | aydiosmio | "limited edition" -- haven't seen them since before halloween |
| 01:47.28 | Supaplex | the strawberry tootsie pops are a welcome sight :) |
| 01:47.35 | Supaplex | pity |
| 01:47.48 | aydiosmio | http://www.candydirect.com/bulk/Tootsie-Rolls-Vanilla.html |
| 01:47.54 | Supaplex | like the orange creeme kit kats ouuu hooo ooo |
| 01:47.56 | aydiosmio | 4.5lbs for $25 |
| 01:54.42 | *** join/#asterisk QbY (n=Kelvin@016-032-051.area7.spcsdns.net) |
| 01:55.18 | QbY | Ok.. Probably old news.. Is it safe to assume that VoicePulse is down again? If so, do they know? |
| 01:58.24 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 02:04.04 | linagee | QbY: ohhh! :( |
| 02:04.06 | linagee | QbY: here too. |
| 02:04.07 | linagee | damn |
| 02:04.10 | linagee | damn damn damn |
| 02:04.14 | linagee | this is the second time |
| 02:04.18 | linagee | (for me anyway) |
| 02:04.22 | QbY | third for me |
| 02:04.26 | linagee | ew |
| 02:04.33 | linagee | QbY: third strike is a charm? |
| 02:04.33 | QbY | they were so rock solid for so long. |
| 02:04.36 | QbY | yep.. |
| 02:04.56 | linagee | QbY: i noticed voicepulse also increased their rates for international |
| 02:05.06 | linagee | QbY: and for some reason i am being charged double... ??? |
| 02:05.08 | linagee | (afaik) |
| 02:05.13 | QbY | ouch |
| 02:05.15 | linagee | was going to call on monday |
| 02:05.17 | linagee | QbY: indeed |
| 02:05.21 | QbY | beginning of the end |
| 02:05.28 | linagee | QbY: three DIDs should be like $33/mo, right? |
| 02:05.34 | QbY | yep. |
| 02:05.47 | linagee | QbY: somehow i am being charged like $88/mo, or misunderstanding their billing or balance screen |
| 02:06.11 | QbY | that balance screen is for the life of the account |
| 02:06.23 | linagee | QbY: "Service Charge" = ? |
| 02:06.31 | QbY | those are your total service charges |
| 02:06.33 | linagee | er, let me make sure i have the right term |
| 02:06.55 | QbY | mine is like 5,000 |
| 02:07.03 | QbY | and i know that's not what we pay a month to them |
| 02:07.18 | linagee | QbY: My Payments, My Service Charges, My Usage |
| 02:07.22 | QbY | yeah |
| 02:07.27 | linagee | can you explain each? |
| 02:07.33 | linagee | my payments = lifetime charges? |
| 02:07.40 | linagee | my service charges = monthly charges, right? |
| 02:07.45 | QbY | My Payments == Total money you've sent them (for the life of the account) |
| 02:07.45 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:07.48 | linagee | my usage = per minute like for international? |
| 02:07.54 | linagee | ok |
| 02:07.54 | QbY | My service charges == prices of your DIDs |
| 02:08.03 | linagee | $88 for My Service Charges? |
| 02:08.11 | QbY | My service charges == Total service charges you've paid them, including the prices of your DIDs |
| 02:08.17 | linagee | i click on "Numbers" and it has three DIDs...? |
| 02:08.30 | linagee | i have two 858 area code DIDs and one toll free one |
| 02:08.44 | linagee | "$11/number activation fee |
| 02:08.44 | linagee | $11/number per month (subtracted from credit on 1st of every month)" |
| 02:09.10 | linagee | and no, this is not the first month i've had voicepulse, so it's not like my service charge is higher because of the activation |
| 02:09.14 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 02:09.29 | QbY | k. |
| 02:09.29 | linagee | QbY: something is fishy |
| 02:09.40 | QbY | hehe.. |
| 02:09.47 | linagee | QbY: oh! |
| 02:09.59 | linagee | QbY: i click on "Calls" to generate a call log report |
| 02:10.06 | linagee | QbY: (trying to figure out the charges) |
| 02:10.19 | linagee | i click generate, it makes a .zip, but it's a blank .xls!!! |
| 02:10.31 | linagee | QbY: the only .zip files that have anything are from like three months ago |
| 02:10.44 | QbY | you need to generate a new one.. |
| 02:10.50 | linagee | QbY: sounds like something is seriously corrupt with their system (either literally or intentionally) |
| 02:10.58 | linagee | QbY: i click generate. it's blank. |
| 02:11.15 | linagee | QbY: i look through all the zips i have on the screen. the only info i have is from 09-31-2006 |
| 02:11.16 | linagee | one day |
| 02:11.19 | QbY | it takes a while to generate them |
| 02:11.23 | linagee | QbY: no |
| 02:11.27 | linagee | it comes right back for me |
| 02:11.30 | linagee | and then it's blank...?? |
| 02:11.48 | linagee | (there is a new zip on the screen. comes right back. i realize it should take a while...) |
| 02:12.08 | linagee | QbY: like i said, it's like their system is corrupt, or ??? |
| 02:12.46 | linagee | QbY: and to think! i was *about* to print business cards w/voicepulse numbers. :-/ |
| 02:13.22 | QbY | linagee.. it could be because their server is down |
| 02:13.34 | QbY | who knows, but i'm gonna sign LOAs tomorrow and try to port my numbers away. |
| 02:13.39 | linagee | QbY: that strangeness happened when it was up and working. was trying to figure out the weird charges. |
| 02:13.52 | linagee | QbY: is there a per month to port a number away? |
| 02:14.05 | linagee | QbY: does it actually tie up channels with voicepulse when the number is forwarded? |
| 02:14.28 | linagee | QbY: in other words, if voicepulse is down, will the forwarding be affected? |
| 02:14.47 | linagee | IANAPG |
| 02:15.56 | QbY | linagee.. Porting doesn't work that way.. Once the number is ported, voicepulse wouldn't see any more traffic on it.. |
| 02:16.20 | linagee | QbY: so porting a number does something in the "big database in the sky that determines what phone switch your call goes to" |
| 02:16.23 | linagee | SS7 and such |
| 02:16.26 | linagee | or something |
| 02:16.27 | QbY | yes. |
| 02:17.01 | linagee | QbY: big database in the sky that determines what phone switch your call goes to = ? :) |
| 02:17.17 | QbY | i guess so.. i'm not 100% sure of how porting works.. |
| 02:17.32 | linagee | QbY: maybe another chan phone guru can explain |
| 02:19.46 | linagee | "LNP is made possible by the Location Routing Number (LRN" |
| 02:19.59 | linagee | "LNPs and LRNs are supervised by the Number Portability Administration Center operated by NeuStar, Inc. under the appointment of the Federal Communications Commission (FCC)." |
| 02:20.09 | linagee | interesting. it seems all the eggs are in one basket/one company. heh |
| 02:20.50 | linagee | "Each local exchange and long distance carrier needs to know what that new LRN is so when someone in an another area dials the number being ported, the carrier knows what LRN to route to. This is accomplished through Local Service Management System (LSMS) databases distributed among the exchange carriers." |
| 02:21.02 | linagee | hrm |
| 02:21.49 | linagee | "LNPs and LRNs are supervised by the Number Portability Administration Center, operated by Lockheed Martin under the appointment of the Federal Communications Commission (FCC)." |
| 02:22.09 | linagee | interesting. one says NeuStar Inc, one says Lockheed |
| 02:24.25 | linagee | "n November 1999, Lockheed-Martin Information Management Services became NeuStar, Inc. NeuStar continues to be the LNP Administrator managing the NPAC." |
| 02:24.56 | *** join/#asterisk AlexGC (n=hpod@189.157.181.180) |
| 02:25.08 | linagee | interesting... |
| 02:25.11 | linagee | "Location Routing Number A 10 digit number used to uniquely identify a switch that has ported numbers. LRN utilizes AIN triggers, SS7 signaling, and unique 10 digit code for switch identification." |
| 02:25.29 | linagee | there should be a dash after Routing Number |
| 02:27.53 | linagee | _VoicePulse: you're down |
| 02:32.24 | AlexGC | good evening/night |
| 02:33.08 | AlexGC | I'm follwing this instrucions that tell me to type this line: svn co http://svn.digium.com/svn/asterisk-sounds/trunk asterisk-sounds |
| 02:33.23 | AlexGC | but I'm getting an Could not open resquestes SVN filesystem |
| 02:33.31 | AlexGC | all other worked fine. |
| 02:33.41 | AlexGC | any pointers to the correct one? |
| 02:35.13 | Qwell | asterisk-sounds is no more |
| 02:35.27 | Qwell | ignore that step, and move on |
| 02:35.37 | linagee | QbY: still here? |
| 02:35.39 | linagee | wtf? |
| 02:35.49 | AlexGC | thanks qwell :) |
| 02:35.51 | Qwell | just when you get to compiling asterisk, do a `make menuselect`, and enable the extra sounds, if you want them |
| 02:36.28 | linagee | what does it mean if my NPA-NXX for voicepulse does not appear in the nationalpooling.com list? |
| 02:36.31 | linagee | huh??? |
| 02:36.41 | linagee | 858-605 |
| 02:36.47 | linagee | (-XXXX) |
| 02:37.38 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-139-149.dyn.embarqhsd.net) |
| 02:39.14 | linagee | oh that's weird... hrm |
| 02:40.08 | linagee | even my cell phone and home phone are not on the list |
| 02:42.33 | *** join/#asterisk icyfire0573 (n=icyfire@u1016342.ul.warwick.net) |
| 02:43.52 | AlexGC | Qwell.. I'm following the instructions to instal asterisk on edubuntu. Are this any good? or just the part of the sounds is outdated? |
| 02:43.54 | icyfire0573 | this is a dumb question. How do I do a conference call w/ asterisk? I can do conference calling with my phone (SPA941) but I don't know how to make asterisk make the calls and join them together. |
| 02:46.22 | QbY | Voicepulse should truly have an after-hours number, or pager.. |
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| 03:12.15 | variable_office | can asterisk output voice ds1? or only input? |
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| 03:32.30 | *** join/#asterisk tim27dr (n=tim27@97-70.dr.cgocable.ca) |
| 03:33.56 | tim27dr | any know where i can download the french sounds prompts, this link dont seem to work svn co http://svn.digium.com/svn/asterisk/sounds/fr/trunk |
| 03:40.35 | *** join/#asterisk threat2 (i=threat@60-240-43-214.static.tpgi.com.au) |
| 03:42.12 | tim27dr | any can help me ??? |
| 03:52.59 | *** join/#asterisk bmg505 (n=leon@c1-76-3.rndf.isadsl.co.za) |
| 03:53.38 | threat2 | hmmm, tim was only in here for 5 minutes :/ |
| 03:56.15 | threat2 | What type of support did he except? |
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| 04:15.54 | *** join/#asterisk packetman (n=entworks@d57-225-189.home.cgocable.net) |
| 04:16.38 | packetman | Can anyone support and current Dell servers that would support asterisk and a Digium TDM400 card well? |
| 04:17.06 | packetman | opps I mean Can anyone suggest a current Dell servers that would support asterisk and a Digium TDM400 card well? |
| 04:20.36 | nick125 | most dells should work |
| 04:21.24 | JT | maybe you also mean "server" unless you need multiple |
| 04:21.30 | JT | yeah, they "should" work |
| 04:21.47 | JT | poor zap timing accuracy will be your biggest issue |
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| 04:38.26 | packetman | hmm |
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| 04:38.44 | packetman | Im looking at new 1U rack servers from Dell |
| 04:39.00 | packetman | you know of any that will support PCI card for the TDM400? |
| 04:54.39 | JT | if the card fots |
| 04:54.44 | JT | fits |
| 04:54.57 | JT | it's full height pci |
| 04:57.49 | Qwell | ha |
| 04:58.02 | Qwell | I just iaxyified my phone :D |
| 04:58.11 | Qwell | It's now officially a "voip phone" |
| 04:58.16 | JT | what sort of phone? |
| 04:58.19 | Qwell | http://cgi.ebay.com/VINTAGE-ORANGE-ROTARY-DIAL-WALL-PHONE-telephone_W0QQitemZ140030993301QQcmdZViewItem |
| 04:58.35 | Qwell | with a builtin iaxy :P |
| 04:59.34 | Qwell | cool side effect, is that the LEDs from the iaxy shine through the hole at the hookswitch...so, when it's offhook, it shines orange, when it's onhook, it's hidden, except from an extreme angle |
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| 05:08.10 | file | Qwell: geek. |
| 05:08.28 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 05:09.37 | bobloblian | Hi, I am finding myself a little unclear about using analog phones, I should be able to use a tdm400p, but are there limits to which phones I can use? for example, can I use an old meridian phone? |
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| 05:10.11 | Qwell | file: You have to see it someday |
| 05:11.21 | file | it's probably a brain washing phone as well |
| 05:11.39 | Qwell | it might be |
| 05:15.33 | bobloblian | or even better, is there an analog phone anyone would recommend to connect to the tdm400p? |
| 05:18.28 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 05:19.05 | *** join/#asterisk sysreq (n=sysreq@142-217-128-85.telebecinternet.net) |
| 05:19.08 | Qwell | bobloblian: doesn't really matter |
| 05:19.14 | Qwell | as long as it's analog |
| 05:19.33 | Qwell | bobloblian: see the link above - even one of those would work with a tdm400p |
| 05:21.07 | bobloblian | ok, but would it support putting one line on hold and answering another and that kind of thing? |
| 05:22.08 | bobloblian | or are there analog phones that would? |
| 05:22.36 | Qwell | "on hold".. you'd flash over with call waiting/three way calling |
| 05:22.48 | Qwell | same as with any old analog phone on a traditional pstn line |
| 05:22.59 | Qwell | just hit the hookswitch for a second |
| 05:23.33 | *** join/#asterisk rumba (n=info@cpe-68-201-149-21.sw.res.rr.com) |
| 05:25.42 | bobloblian | ok, so basically any phone will work, but won't do as many things as an ip phone will do, correct? |
| 05:26.11 | Qwell | pretty much |
| 05:28.17 | bobloblian | ok, ty |
| 05:29.18 | Qwell | any *analog* phone |
| 05:34.18 | *** join/#asterisk ltd (n=z@202-161-7-241.dyn.iinet.net.au) |
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| 05:47.54 | *** join/#asterisk ProActive (n=ProActiv@bl7-140-137.dsl.telepac.pt) |
| 05:47.58 | ProActive | Hi all |
| 05:53.00 | *** join/#asterisk [pyro] (n=pyro@tor/regular/bracketed-pyro) |
| 05:59.57 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 06:01.44 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 06:01.48 | Sargun | how is chan_bluetooth |
| 06:03.11 | Sargun | or not |
| 06:09.53 | sjobeck | hi all |
| 06:10.00 | sjobeck | how is everyone tonight? |
| 06:10.15 | Sargun | Great! |
| 06:10.26 | sjobeck | wondering if any one out there is or knows someone who is a hardhitting t1 expert for a project I have mapping incoming channel to outgoing channel |
| 06:10.53 | sjobeck | pretty low level protocol stuff going on in the project |
| 06:10.56 | sjobeck | beyond me |
| 06:11.07 | sjobeck | any and all ideas welcomed. thanks so much |
| 06:11.15 | Sargun | Anyone here selling CDMA motorola devices? |
| 06:12.54 | Sargun | sjobeck, Where is the NDA? ;-) |
| 06:13.15 | sjobeck | sargun: pls explain |
| 06:13.30 | Sargun | Non-disclosure agreement to find out more about the project |
| 06:14.28 | sjobeck | sargun: ah, yes, right, thx ............ well, if I dont find someone who has lots of experience at the low levels of channel & frames & protocols & so on, no point in going further, dont you agree |
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| 06:33.44 | parag_ast | Can anybody let me know meaning of this " Requested indication -1 on channel Zap/2-1 " |
| 06:36.06 | Sargun | That means something bad happened |
| 06:36.21 | Sargun | with your channel Zap/2-1 |
| 06:36.38 | Sargun | btw: you need to post a full log with verbosity at at least 10 |
| 06:40.40 | icyfire0573 | this is a dumb question. How do I do a conference call w/ asterisk? I can do conference calling with my phone (SPA941) but I don't know how to make asterisk make the calls and join them together. |
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| 06:57.28 | trelane | icyfire0573, google meetme() |
| 06:59.24 | parag_ast | Sargun, do u want the logs of /var/log/asterisk/full ?? |
| 07:01.21 | parag_ast | :/topic #asterisk |
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| 07:15.07 | neoalex | hi, is there a softphone for linux supporting g729 |
| 07:18.52 | Sargun | maybe |
| 07:19.55 | neoalex | ummm... that´s helpfull |
| 07:20.39 | hads | You wont find a free one |
| 07:21.36 | sbingner | hey how do I get two * servers to synch the voicemail waiting indicator? |
| 07:22.02 | neoalex | just like there´s no free implementation of g729 for asterisk or really won´t find one |
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| 07:24.03 | bigreddog | hi all, I hope someone can help a newbie. I have a smc-ultra card. I'd like to use it with A@H but do not know how to get the driver compiled? anyone know? |
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| 07:30.58 | bigreddog | hi all, I hope someone can help a newbie. I have a smc-ultra card. I'd like to use it with A@H but do not know how to get the driver compiled? anyone know? |
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| 07:50.45 | *** join/#asterisk bigreddog (n=me@vc-196-207-41-253.3g.vodacom.co.za) |
| 07:51.55 | bigreddog | hi anyone know how to compile a network device driver? |
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| 08:49.40 | roxy_ | when a sip user identified it does it against a password in sip.conf. Is there a way to do it against a LDAP ? |
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| 09:40.14 | Sargun | roxy_, hello |
| 09:40.41 | Sargun | roxy_, by investing in a programmer, I believe there is a way to do that |
| 09:48.53 | roxy_ | Sargun: you mean, atm there isn't ? |
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| 11:33.09 | redozqld | Anyone got some time to answer some questions about multiple x100p cards in a single box using tb 1.2.3?? (Complete newbie but there is hope for me!) |
| 11:35.19 | *** part/#asterisk cybergyp1y (n=cybergyp@APoitiers-157-1-21-46.w83-193.abo.wanadoo.fr) |
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| 11:45.49 | redozqld | I've got 4 cards in a trixbox 1.2.3 machine, when any of the cards is connected (any of the 4) to a phone line, and is rung, it always goes to the same end point (be it a core point or whatever) is there any reading or any way to allow each card to ring and go to a set point??? any config files to change?? any ideas ;) thanks... |
| 11:46.16 | wasim | redozqld: extensions.conf |
| 11:46.43 | redozqld | any good reading for this?? |
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| 11:54.52 | JT | ~thebook |
| 11:55.00 | jbot | i guess thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
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| 12:01.12 | redozqld | Thanks I'll start reading... it cant be that hard |
| 12:06.29 | pipipi | wow, nice book, available FREE, awesome! |
| 12:08.31 | tzafrir | redozqld, this should be possible, IIRC, with FreePBX (which is the interface you see in trixbox) |
| 12:08.32 | tzafrir | ask in #freepbx |
| 12:09.22 | EmleyMoor | I now have the zaptel modules on - in the absence of my card, I guess the only thing I can do is set the country? |
| 12:12.21 | tzafrir | EmleyMoor, if you need zaptel for timing and have no card, use ztdummy |
| 12:12.32 | EmleyMoor | I have a card in transit |
| 12:12.48 | EmleyMoor | I do not have any need for zaptel other than that |
| 12:15.33 | tzafrir | Zaptel is used as a timing source in various places |
| 12:15.39 | tzafrir | (in Asterisk) |
| 12:16.15 | EmleyMoor | Yes - I have no purpose for them until the arrival of the card as it happens |
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| 13:05.19 | *** join/#asterisk enzo (n=enzo@143-023.proxy.saarctextile.com) |
| 13:05.25 | enzo | hi |
| 13:05.35 | enzo | do you know if zaptel drivers are in linux kernel ? |
| 13:13.41 | xheliox | <PROTECTED> |
| 13:17.18 | enzo | why ? |
| 13:17.29 | enzo | should be interesting to add drivers to linux trunk |
| 13:17.36 | *** join/#asterisk eYeLez (i=clkio@dxb-as31747.alshamil.net.ae) |
| 13:17.51 | Qwell | enzo: What kernel are you running right now? |
| 13:18.02 | enzo | i compile the last 2.6.18.2 |
| 13:18.10 | Qwell | okay, bad person to ask :p |
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| 13:18.27 | enzo | (cause i can't install zaptel on my kernel2.4.27...) |
| 13:18.32 | Qwell | personally, I run 2.6.15 on the box that runs asterisk |
| 13:18.34 | eYeLez | hi, i would like to get into asterisk, what is the cheapest way to get up and running? i noticed the cost of the cards on digium are way up there $400 ouch!, im in learning stage, whats a cheap easy to find card to goof around from? |
| 13:18.43 | Qwell | I don't want to upgrade my kernel every time there is a zaptel fix |
| 13:19.01 | enzo | i don't want either Qwell |
| 13:19.03 | Qwell | eYeLez: $400 is with like 4 modules |
| 13:19.13 | enzo | but i need to install zaptel :) |
| 13:19.22 | Qwell | enzo: indeed, and it's only like 1 command |
| 13:19.24 | eYeLez | will a standard analog modem do? |
| 13:19.28 | Qwell | eYeLez: no |
| 13:19.35 | Qwell | an analog modem does not an fxs make |
| 13:19.38 | Qwell | fxo* |
| 13:20.06 | eYeLez | dont they have two ports one to phone and one to line? hence fxs + fxo? |
| 13:20.12 | Qwell | no... |
| 13:20.20 | eYeLez | or im just confused :) |
| 13:20.21 | enzo | do you know if digium will make asterisk gui stable enough for the 1.4 release? |
| 13:20.28 | Qwell | the "phone" port is just passthrough |
| 13:20.45 | eYeLez | true that! |
| 13:20.54 | Qwell | enzo: it isn't part of 1.4, so that's fairly irrelevant |
| 13:21.06 | enzo | k |
| 13:21.15 | enzo | i hope sooner asterisk will have a gui |
| 13:21.24 | Qwell | if there are specific issues you're seeing, open a bug |
| 13:21.35 | xheliox | I don't, but I'm selfish. :) |
| 13:22.01 | Qwell | bbl, airport :D |
| 13:22.14 | eYeLez | Qwell so which piece of hardware you recommend? single fxs + fxo.. this is too much: http://www.digium.com/en/products/hardware/analogcards.php |
| 13:22.27 | Qwell | eYeLez: for 2 ports, it's about $200 |
| 13:22.36 | Qwell | and yes, that's what I would recommend |
| 13:23.22 | eYeLez | ok thats for hardware |
| 13:23.35 | xheliox | eYeLez: I'm not making a recommendation one way or another, but Digium isn't the only company who makes Zaptel compatible hardware. |
| 13:24.38 | eYeLez | now is this possible, i want to eventually set up a voip/pstn gateway for calling card companies in north america dump their traffic destined to my country to my server, is that possible with asterisk ? in other words am i on the right track? |
| 13:30.32 | Rhizome | eYeLez: yup no problem ;) |
| 13:31.01 | *** join/#asterisk Arnar_ (n=arnarb@landi.oddi.is) |
| 13:34.34 | eYeLez | thanks rhizo |
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| 15:18.19 | gmustafa | i want to use Quintum as FXo gateway with asterisk |
| 15:18.24 | gmustafa | any howto? |
| 15:18.30 | gmustafa | or thoughts ? |
| 15:24.17 | EmleyMoor | Is there a howto on handling different ring cadences/ |
| 15:24.18 | EmleyMoor | ? |
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| 16:14.14 | hwt | hi. in asterisk realtime, can i simply change an extension on the fly and it will work, or do i have to reload? |
| 16:15.05 | InfraRed | you need to reload |
| 16:15.13 | InfraRed | it reads the config files at the start |
| 16:16.41 | hwt | InfraRed: i'm talkin realtime her. mysql database. |
| 16:18.32 | InfraRed | havent used * realtime |
| 16:18.51 | InfraRed | but if it matches the extension as a query everytime an incoming call is in |
| 16:18.56 | InfraRed | you shouldnt need to realod |
| 16:19.05 | InfraRed | try and see? |
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| 16:37.05 | EmleyMoor | Am I right in thinking I can tell different ring cadences apart using an FXO module in asterisk? |
| 16:40.06 | EmleyMoor | (so that I could have it ignore "ring-pause", answer straight to IVR on "ring-ring-pause", and call all phones on "ring-ring-ring-pause") |
| 16:41.58 | *** join/#asterisk saftsack (n=saftsack@pD9E07CD4.dip.t-dialin.net) |
| 16:42.42 | [TK]D-Fender | EmleyMoor : There is a measure of "distinctive ring support for Zaptel FXo channels. Look up here : http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels |
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| 16:44.48 | Weezey | anything new and exciting in the world of *? |
| 16:46.11 | EmleyMoor | Hmmm... it seems pulse dial is supported too! |
| 16:48.56 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 16:50.35 | EmleyMoor | Both my Yeomen need new bells - will have to order some |
| 16:53.39 | awannabe | is there a way to have each voicemail box have differnet options as far as having some boxes get a attacemtns, and others not. can that be done? |
| 16:54.15 | Dovid | yes |
| 16:54.44 | Dovid | awannabe: the settings for if they should recieve an email or not is in voicemail.conf |
| 16:54.51 | [TK]D-Fender | awannabe : Go read the WIKI page on voicemail.conf |
| 16:54.54 | Dovid | if u are using real time you can set it there |
| 16:55.58 | awannabe | [TK]D-Fender, yeah it says for the whole context, ill go look again |
| 16:56.11 | awannabe | maybe this confilg file is old somehow, hrmm |
| 16:56.38 | EmleyMoor | OK - further question: Anyone used distinctive ring (CallSign) on BT with asterisk? |
| 16:57.09 | awannabe | i see it now, im just blind!! lol |
| 16:57.12 | MoutaPT | If i want my TDM board channels to be port=1,2,3,4 and my TE110P board to be the next channels on my system, my span definition for TE110P must be : |
| 16:57.12 | MoutaPT | span=2,1,0,esf,b8zs |
| 16:57.12 | MoutaPT | bchan=5-28 |
| 16:57.12 | MoutaPT | dchan=29 |
| 16:57.12 | MoutaPT | fxsks=1,2 |
| 16:57.14 | MoutaPT | fxoks=3,4 |
| 16:57.16 | MoutaPT | is this right? |
| 16:57.21 | Dovid | MoutaPT: |
| 16:57.23 | Dovid | ~pb |
| 16:57.24 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 16:58.52 | file | [TK]D-Fender: rain is eeeeevil |
| 17:00.42 | MoutaPT | you are right, bust because this wasn't a logging or something that kind i didn't use paste bin |
| 17:00.44 | MoutaPT | Sorry guys |
| 17:00.50 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 17:02.22 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 17:04.01 | [TK]D-Fender | awannabe : * e user_option(s) field can be used to override default settings defined in the general section, or set a specific time zone for this user. Specifically, there are 9 setting=value pairs which can be specified in the user_option(s) field. There can be multiple setting=value pairs defined in the user_option(s) field. Each setting=value pair after the first must be delimited with a vertical bar (|). The nine settings which may be used are: a |
| 17:04.01 | [TK]D-Fender | [general] section. The tz setting is used to override the default time zone and it must be set to a custom time zone defined in the [zonemessages] section. |
| 17:04.19 | [TK]D-Fender | Hmmm, there really WAS a line-break there... |
| 17:05.11 | [TK]D-Fender | file : Rain is infinitely preferrable to SNOW. |
| 17:05.24 | file | true |
| 17:06.31 | awannabe | [TK]D-Fender, yeah i got it now :) i guess im being a moron is all, heh, thanks! |
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| 17:08.34 | UnixBehr | good morning |
| 17:09.56 | *** part/#asterisk UnixBehr (n=UnixBehr@ppp-71-128-113-209.dsl.irvnca.pacbell.net) |
| 17:10.19 | cosmic_hippo | I have a tdm400p with 2 fxs modules. Zaptel.conf(fxoks=1-2). ztcfg shows 2 channels configured. However, zap show channels just shows the pseudo chan |
| 17:11.04 | cosmic_hippo | So asterisk doesn't seem to see the chans. Any idea? |
| 17:13.04 | Weezey | cosmic_hippo: you added them to /etc/asterisk/zapata.conf ? |
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| 17:14.30 | UnixBehr | anyone here use phonecall ? |
| 17:14.39 | UnixBehr | the gui |
| 17:15.59 | UnixBehr | good morning btw |
| 17:17.04 | cosmic_hippo | Weezey: Hrrrm. good question. I have a group=1 in there but no specific channel info |
| 17:21.00 | [TK]D-Fender | cosmic_hippo : That would do it.... |
| 17:21.21 | cosmic_hippo | haha, awesome. let see...man zapata.conf :) |
| 17:21.37 | cosmic_hippo | thanks! |
| 17:25.47 | tzafrir | cosmic_hippo, right. Those files lack man pages |
| 17:25.52 | *** join/#asterisk bkw__ (n=ASSERTKI@m3710fa48.tmodns.net) |
| 17:26.26 | tzafrir | Now I wonder if anybody considers a nice script to convert the sample config file to man pages... |
| 17:26.38 | cosmic_hippo | haha, I should be able to catch some good info online |
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| 17:27.59 | hwt | hm, in realtime i can't get it to ring on multiple phones. |
| 17:28.52 | tzafrir | cosmic_hippo, the sample config files, which are also part of the "api documentation" (huge, but in html format) generated by doxygen |
| 17:29.02 | tzafrir | and the wiki: |
| 17:29.06 | *** part/#asterisk UnixBehr (n=UnixBehr@ppp-71-128-113-209.dsl.irvnca.pacbell.net) |
| 17:29.07 | tzafrir | ~wiki |
| 17:29.22 | cosmic_hippo | awesome -- thanks tzafrir! |
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| 17:29.36 | hwt | is it poor security to include the "local users context" in the inbound context? |
| 17:30.14 | quidpro | Hmm, is there a good channel out there for cellphones? I'm trying to find a recommendaiton on a cheap GSM/CDMA multi-mode. :) |
| 17:30.19 | Bladerunner05 | hi all, I'm looking for a good configuration for bt100 and asterisk |
| 17:31.13 | [TK]D-Fender | hwt : Depends whats in there. |
| 17:32.03 | hwt | [TK]D-Fender: well, that's the problem. the local context has outbound included. :) |
| 17:32.14 | hwt | maybe a goto is safer then. |
| 17:32.18 | [TK]D-Fender | hwt : I think you can answer your own question then :) |
| 17:32.35 | hwt | [TK]D-Fender: yup. so you would also go with a goto? :) |
| 17:33.00 | [TK]D-Fender | hwt : Maybe you should create 1 context for internal extens, one for external, and then creat one for your suers the INCLUDEs both, and use the internal one for your inbound calls. |
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| 17:49.41 | gmustafa | hi |
| 17:49.55 | gmustafa | i want to use Quintum with asterisk as FXO gateway |
| 17:49.59 | gmustafa | any howto |
| 17:53.19 | cosmic_hippo | Back to the zapata.conf. My extension specific settings are in zapata_additional.conf which is included. But I still don't have any channels other than the pseudo channel for zap show channels |
| 17:54.12 | MoutaPT | did u run genzaptelconf? |
| 17:54.36 | cosmic_hippo | no -- question on that though. will that overwrite my curent /etc/zaptel.conf ? |
| 17:54.50 | MoutaPT | yes |
| 17:55.03 | Corydon76-home | cosmic_hippo: pastebin, please |
| 17:55.07 | MoutaPT | what's in your zaptel.conf |
| 17:55.09 | MoutaPT | ? |
| 17:55.15 | MoutaPT | paste bin it |
| 17:55.15 | cosmic_hippo | aight -- stby |
| 17:56.49 | cosmic_hippo | http://pastebin.com/822652 |
| 17:57.08 | cosmic_hippo | Now I added the fxoks=1-2. Now it looks like I've done this backwards |
| 17:57.16 | cosmic_hippo | should I use genzaptel.conf instead? |
| 17:57.26 | MoutaPT | do you have 2 modules FXS ? |
| 17:57.28 | MoutaPT | is that? |
| 17:57.48 | cosmic_hippo | yea --tdm400 with 2 fxs modules |
| 17:58.01 | Corydon76-home | What does your zapata.conf (and any included files) look like? |
| 17:58.02 | MoutaPT | make ztcfg -vvvv |
| 17:58.44 | cosmic_hippo | ztcfg -vvvv: http://pastebin.com/822655 |
| 17:59.26 | MoutaPT | every thing sounds ok |
| 17:59.35 | MoutaPT | what's in your zapata.conf |
| 17:59.43 | MoutaPT | as well as zapata_additional.conf |
| 17:59.45 | MoutaPT | pbin |
| 18:00.20 | cosmic_hippo | zapata.conf: http://pastebin.com/822657 |
| 18:00.25 | cosmic_hippo | zapata_additional.conf: http://pastebin.com/822659 |
| 18:01.29 | Corydon76-home | cosmic_hippo: your include statement has incorrect syntax. It should be #include "zapata_additional.conf" |
| 18:01.43 | Corydon76-home | Note the quotes and the pound sign |
| 18:02.03 | cosmic_hippo | stby...yea that was me. Ensuring that it was included. I forgot to change that back |
| 18:02.16 | MoutaPT | yeah that's the problem |
| 18:02.25 | MoutaPT | zapata_additional doesn't seem to be wrong |
| 18:02.34 | MoutaPT | reload your asterisk |
| 18:02.43 | cosmic_hippo | and *bingo* |
| 18:02.51 | cosmic_hippo | I have a channel now! |
| 18:03.03 | Corydon76-home | It also needs to be channel => 1-2 |
| 18:03.19 | cosmic_hippo | in zapata_additional? |
| 18:03.33 | Corydon76-home | Yes, if you intend to configure both channels |
| 18:04.04 | cosmic_hippo | yup. Ok |
| 18:04.35 | *** join/#asterisk dusan2 (n=dusan@209-223-47-160-static.oplink.net) |
| 18:05.40 | cosmic_hippo | awesome thanks for the help everybody! |
| 18:07.17 | *** join/#asterisk dusan2 (n=dusan@209-223-47-160-static.oplink.net) |
| 18:17.46 | *** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at) |
| 18:23.36 | cosmic_hippo | Now that I got my zaptel working...I'm trying to conquer one more issue. I'm using trixbox...and attempting to define my Inbound route. But I'm getting "No DID or CID Match" no matter what I define. |
| 18:24.34 | cosmic_hippo | so if I just use my full number I get the "number you have dialed is not in service" recording and "No DID or CID Match" on the asterisk cli |
| 18:24.53 | MoutaPT | you need to use exten=> s,1, Answer |
| 18:24.53 | cosmic_hippo | it works fine with any did/cid to an extension though |
| 18:25.03 | MoutaPT | TDM boards don't use DID |
| 18:25.20 | MoutaPT | s extension will answer you call |
| 18:25.39 | cosmic_hippo | I'm just using the tdm board for ringing my analog fax. I'm using viatalk as my sip gateway |
| 18:27.38 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 18:29.57 | MoutaPT | ok so in incoming calls context |
| 18:30.03 | MoutaPT | use "s" extension |
| 18:30.16 | MoutaPT | and rx application |
| 18:30.22 | MoutaPT | to handle your faxes |
| 18:30.50 | MoutaPT | i'm not sure if you will get CallerId with TDM |
| 18:30.58 | MoutaPT | any one here can talk about it? |
| 18:32.13 | cosmic_hippo | I'm not sure that I understand it when you say "get callerid with tdm" I'm attempting to route an incoming call from a sip provider (vialtalk) to an fxs extension. That routing would be based on DID information from viatalk. So at that point the TDM card is a destination. I'm I completely lost on this? |
| 18:33.16 | MoutaPT | now i got what you want |
| 18:33.17 | MoutaPT | :) |
| 18:33.22 | cosmic_hippo | heh, sorry |
| 18:33.38 | MoutaPT | which calls you want to route? |
| 18:33.39 | MoutaPT | all? |
| 18:33.42 | MoutaPT | to fax? |
| 18:34.03 | cosmic_hippo | nah, I've got all working. But I'm going to get another line through viatalk to use as a fax line. So I will want to route that did to my fxs fax |
| 18:34.09 | MoutaPT | not sure you will be sucessfull with FAx from VoiP to FXS:) |
| 18:34.18 | MoutaPT | but u can try |
| 18:34.28 | cosmic_hippo | ahhh |
| 18:34.32 | cosmic_hippo | haha |
| 18:34.34 | MoutaPT | ok so you just need to define a context |
| 18:34.41 | MoutaPT | for this peer trunk |
| 18:34.55 | MoutaPT | i mean it is a peer in your sip.conf |
| 18:35.11 | MoutaPT | but you have more accounts on this sip provider? |
| 18:35.54 | cosmic_hippo | hrrrm. I was going to try to just add onother line to my account. Having both of the numbers go to the same account and utilize * to recognize the did to route |
| 18:36.01 | cosmic_hippo | not sure if that's possible though ? |
| 18:36.04 | gmustafa | hi |
| 18:36.11 | MoutaPT | it is possible |
| 18:36.12 | gmustafa | i am new to asterisk |
| 18:36.25 | MoutaPT | but i depends of what you have with your provider |
| 18:36.33 | MoutaPT | you have DID or another account? |
| 18:36.40 | gmustafa | i have install trixbox, i have a Quitum fxo, i want to use it for calling pstn lines |
| 18:36.45 | gmustafa | any idea? |
| 18:36.46 | MoutaPT | i must tell u i'm not used with DID on Voip providers |
| 18:37.02 | MoutaPT | gmustafa quintum has SIP? |
| 18:37.06 | gmustafa | yes |
| 18:37.20 | gmustafa | Quintm Tenor AX series |
| 18:37.38 | MoutaPT | does it allow register? |
| 18:37.57 | MoutaPT | create a trunk to quintum |
| 18:38.00 | gmustafa | i dont know but here is a SIP singalling group option under VOIP setttings in quitum |
| 18:38.03 | MoutaPT | and define an outbound route |
| 18:38.18 | gmustafa | create a trunk and define outbound route in trixbox? right |
| 18:39.38 | gmustafa | ? |
| 18:41.02 | gmustafa | what do i put in as host name and user& password? |
| 18:42.16 | [TK]D-Fender | gmustafa : please read the channel topic |
| 18:42.42 | gmustafa | opss sorry |
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| 18:43.05 | MoutaPT | host is the Ip of your quintum |
| 18:44.38 | *** join/#asterisk Arnar (n=arnarb@landi.oddi.is) |
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| 18:53.25 | linolopes | hell everybody! Can anyone tell me if I can use Asterisk to operate any external devices, such as open a door, etc? |
| 18:54.41 | *** part/#asterisk cosmic_hippo (n=cosmic_h@cpe-76-185-20-33.tx.res.rr.com) |
| 18:54.55 | quidpro | linolopes: I suppose you could do anything with it, it's a matter of building the hardware interface. |
| 18:55.12 | EmleyMoor | linolopes: Do any of these devices interface as a telephone anywhere? |
| 18:56.11 | linolopes | well, currently I have a regular PBX system which has a module I use to connect the external gate of my house to it |
| 18:56.30 | quidpro | Try googling on asterisk and X10 |
| 18:57.09 | quidpro | lino: What part of BR are you from? |
| 18:57.10 | linolopes | also, I'm able to connect the intercom to this module, so I can answer the intercom and open the front gate if I want to, using a key combination in any of te PBX extensions |
| 18:57.18 | linolopes | I'm in Belo Horizonte |
| 18:57.41 | quidpro | Nice, I am thinking about visintg your country in January... have a friend who lives in SP. |
| 18:57.57 | EmleyMoor | linolopes: That should be quite easy then - as long as there's a way to interface the telephonic side of it to Asterisk easily |
| 18:58.00 | *** part/#asterisk MoutaPT (n=root@a213-22-40-63.cpe.netcabo.pt) |
| 18:58.38 | linolopes | THis is the problem. My current analog PBX system has a proprietary black box I use to bridge the intercom with the PBX |
| 18:59.00 | EmleyMoor | Is that black box effectively a standard telephone? |
| 18:59.03 | quidpro | Lino: You could use your existing PBX and black box as an extension off of Asterisk. |
| 18:59.12 | linolopes | a box such as this one should be constructed to interface the intercom with Asterisk, right? |
| 18:59.33 | saftsack | localnet=10.0.0.0/255.255.0.0 with this option all telephones in 10.0.1.XX and 10.0.2.XX are seen as local, right? |
| 18:59.37 | linolopes | what I want is to get rid of my current PBX, since it's not working as it should... |
| 18:59.50 | EmleyMoor | If the black box is effectively a telephone, you could try it on an FXS port |
| 19:00.34 | linolopes | it's not a phone... it accepts my intercom 2-wire at one side and plugs directly into a proprietary connecton in the PBX side... :( |
| 19:01.47 | EmleyMoor | Oh :-( |
| 19:02.01 | EmleyMoor | No way you can make the old PBX simply act to switch all calls through to it?> |
| 19:02.20 | linolopes | this 2-wire line comming from the intercom apparently trnasmits the voice from the intercom and also serves to send like a pulse to open the front gate |
| 19:02.28 | EmleyMoor | Failing that, you would need some kind of black box |
| 19:02.47 | [TK]D-Fender | linlinlin : You'll want an X10 CM11A controller, and the UM508 Universal module |
| 19:03.00 | [TK]D-Fender | linolopes rather. |
| 19:03.20 | [TK]D-Fender | linolopes : Most door openers run on a basic relay taht all you need to do is short. |
| 19:03.47 | [TK]D-Fender | linolopes : Pretty easy to do, and then while you're at it you can have * control all of your lights, and make coffee for you like mine used to ;) |
| 19:04.12 | linolopes | Fender: That's exactly what my door opener seems to do. It seems to open the gate using a electrical pulse.. |
| 19:04.22 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 19:04.27 | linolopes | I already use X10 to control the lights in my hme theater |
| 19:04.55 | [TK]D-Fender | linolopes : Great, then you only need the UM508 |
| 19:04.56 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 19:05.00 | linolopes | The X10 will solve the problem of openeg the door. And what about answering the intercom? |
| 19:05.01 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 19:05.26 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 19:06.24 | [TK]D-Fender | linlin : Thats trickier. thats a "push-to-talk" intercom isn't it? |
| 19:07.01 | [TK]D-Fender | Dang auto-complete |
| 19:09.54 | linolopes | fender: not really. This is how it works now: One press the intercom ring button at the front gate. The PBX rings all the selected extensions. Then yu can answer the call using any PBX extension. After answering, you can key in a specific key combination and the PBX will open the front gate via the black box |
| 19:12.03 | [TK]D-Fender | linolopes : Ok that soulds like an FXS dialer. That you should be able to integrate as-is. |
| 19:12.30 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 19:13.20 | linolopes | ok, thak you VERY MUCH for your help. I will do more research and find out the best way to do it. |
| 19:13.32 | linolopes | you guys have helped a lot, poiting my the right way |
| 19:13.35 | linolopes | THANKS A LOT! |
| 19:15.03 | *** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt) |
| 19:15.19 | wiseguy_ | hello |
| 19:15.34 | wiseguy_ | is it possible to pickup call from asterisk console? |
| 19:16.23 | [TK]D-Fender | linolopes : no |
| 19:17.34 | *** join/#asterisk asdx (n=ubuntu@200.61.236.33) |
| 19:18.08 | asdx | ok, i just installed asterisk, what should i do now? i want to use only softphones at the moment, just for try it... |
| 19:20.44 | [TK]D-Fender | asds : Go install some soft-phons then. Develop your dialplan, etc. |
| 19:21.39 | asdx | what is dialpan? |
| 19:23.21 | [TK]D-Fender | asdx : extensions.conf |
| 19:25.53 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 19:26.18 | EmleyMoor | Does anyone know how to fill in the boxes in linphone to attach it to asterisk? |
| 19:27.59 | EmleyMoor | For example, do I set it up as a remote service? |
| 19:29.39 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 19:30.02 | EmleyMoor | Ah, got that |
| 19:31.26 | EmleyMoor | How do I make calls on it without needing the @server.name? |
| 19:32.35 | EmleyMoor | Also, what are the good iax softphones these days? |
| 19:33.22 | gmustafa | <PROTECTED> |
| 19:33.22 | gmustafa | <PROTECTED> |
| 19:33.32 | gmustafa | i m using quintum |
| 19:35.09 | EmleyMoor | I'm happy to put moziax on but seek a decent softphone (SIP or IAX) |
| 19:35.12 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:35.51 | *** join/#asterisk adamowitz (n=adamowit@ip68-14-27-224.ri.ri.cox.net) |
| 19:38.46 | CunningPike | asdx: Take a look at The Book |
| 19:38.49 | CunningPike | ~thebook |
| 19:38.51 | jbot | well, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:39.21 | CunningPike | EmleyMoor: What OS? |
| 19:39.44 | EmleyMoor | Linux |
| 19:39.59 | EmleyMoor | Preferably one in stable Debian (or backports therefor) |
| 19:40.09 | [TK]D-Fender | EmleyMoor : Get Ekiga |
| 19:40.19 | EmleyMoor | Ekiga? |
| 19:40.34 | EmleyMoor | Ah... |
| 19:40.41 | [TK]D-Fender | EmleyMoor : Newfie jig |
| 19:40.48 | [TK]D-Fender | !@&? |
| 19:41.02 | [TK]D-Fender | how the heck did taht come out of a copy&paste? |
| 19:41.14 | CunningPike | [TK]D-Fender: p0wned ;) |
| 19:41.22 | [TK]D-Fender | http://www.ekiga.org/ |
| 19:41.45 | [TK]D-Fender | CunningPike : Firefox does some &^#@'d up shit. occasionally locks out my cursor keys editing the address bar too. |
| 19:43.39 | EmleyMoor | Why do softphones not have some consistency in naming? |
| 19:44.24 | Nugget | You mean like the tremendous consistency in naming that we enjoy with every other type of software? </sarcasm> |
| 19:44.56 | EmleyMoor | If they all had SIP or IAX (as appropriate) or phone in the name they'd be so much easier to find |
| 19:45.25 | *** join/#asterisk rg1_ (n=rg1@www.airlinksystems.com) |
| 19:45.33 | EmleyMoor | ekiga needs an audio plugin? |
| 19:45.38 | Nugget | why stop there? software names should also include supported platforms and licensing too! |
| 19:45.58 | Nugget | what should it be named if it supports SIP and IAX? |
| 19:46.03 | rg1_ | anyone know when/where ${DNID} gets set? |
| 19:46.11 | EmleyMoor | Any suggestions as to my audio plugin for ekiga? |
| 19:47.08 | EmleyMoor | ... well? |
| 19:47.19 | [TK]D-Fender | Nugget : And should include the contact info of the authors incluing a picture, and GPS coords so you can launch ICBM's at them for non-compliance via Google Earth! |
| 19:47.28 | Nugget | I agree |
| 19:47.45 | Nugget | but not if the picture is encumbered. only png format. |
| 19:47.54 | *** join/#asterisk af_ (n=af@ip-172-242.sn1.eutelia.it) |
| 19:48.03 | file | Nugget is... speaking? |
| 19:48.24 | [TK]D-Fender | file : on... telnet no less! ;) |
| 19:48.24 | Nugget | telnet is eeeeeeevil! |
| 19:48.43 | EmleyMoor | What audio plugin does ekiga need? (and why did Debian not install it?) |
| 19:49.00 | Nugget | because linux dependancy checking sucks. |
| 19:49.54 | EmleyMoor | Well, yes, it sucks - but what should it have sucked? |
| 19:49.57 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 19:50.16 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 19:50.28 | EmleyMoor | Um... no |
| 19:50.36 | EmleyMoor | I got recommended ekiga here |
| 19:50.54 | rg1_ | anyone out there that can help me know the variable in the dial plan that shows the phone# that was dialed to get into asterisk? |
| 19:51.00 | rg1_ | the DID# |
| 19:51.23 | [TK]D-Fender | rg1_ : ${EXTEN} ..... |
| 19:51.25 | Nugget | rg1_: ${EXTEN} |
| 19:51.47 | rg1_ | ah |
| 19:51.48 | rg1_ | thanks |
| 19:55.19 | EmleyMoor | Any better suggestions for a decent Linux softphone? |
| 19:58.10 | tzafrir | twinkle is nice |
| 19:58.18 | tzafrir | So is kaix |
| 19:58.24 | tzafrir | kiax, that is |
| 20:00.23 | rg1_ | anyone know if there is a way to "dump" all variables that are set in the dialplan? |
| 20:00.46 | EmleyMoor | Hmmm... need work to get them on here but thanks for the hints |
| 20:01.38 | tzafrir | Backports of quite a few voip-related packages could also be found at http://pkg-voip.buildserver.net/ . Though they tend to be bleeding edge |
| 20:01.51 | [TK]D-Fender | rg_ :nope. What are you trying to accomplish? |
| 20:02.08 | *** join/#asterisk jbroome (n=jbroome@unaffiliated/jbroome) |
| 20:03.10 | rg1_ | weel i did ${EXTEN} and it said the value was "s" |
| 20:03.26 | rg1_ | I'm trying to see what values are set/available to me that I might be able to use |
| 20:04.55 | [TK]D-Fender | rg1_ : You may want to actually try LOOKING on the WIKI.... http://www.voip-info.org/wiki-Asterisk+variables |
| 20:06.07 | [TK]D-Fender | rg_ : Again, try and come up with something specific you want to do and maybe we can help. And it came back as "s" becasue thats where you were when you checked. you should look earlier up in your dialplan to see where it came in on, and why. |
| 20:06.08 | xheliox | TK: Why read the documentation when you can come on IRC and guess at it? |
| 20:06.33 | [TK]D-Fender | xheliox : Shucks, why didn't *I* think of that! |
| 20:07.17 | Nugget | Catch a man a fish and you feed him for a day. Catch him a fish on IRC and he'll bitch at you for not cleaning and cooking it for him too. |
| 20:08.03 | [TK]D-Fender | Nugget : "Light a fire for a man and he's warm for a day. Light a man on fire and he'll be warm for the rest of his life" |
| 20:08.19 | rg1_ | D-Fdner - ok, i will look for the exten at the start of the script. thanks much |
| 20:08.27 | xheliox | http://catb.org/esr/faqs/smart-questions.html |
| 20:08.54 | [TK]D-Fender | xheliox : and the link to "dumb answers"? :) |
| 20:09.09 | xheliox | I can supply those. :) |
| 20:09.17 | xheliox | I have years of experience. |
| 20:11.51 | *** join/#asterisk xnon (i=xnon@200.8.30.31) |
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| 20:14.05 | gmustafa | i m getting error "503- service unavailable" in asterisk from quintum |
| 20:16.30 | [TK]D-Fender | gmustafa : If I had to venture a guess I'd say that might mean that the Quintum didn't think it had any free channels to give *. take a closer look and see what resources that connection has available to it. |
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| 20:17.40 | *** join/#asterisk GuadamuX (n=edgar@201.195.22.6) |
| 20:18.31 | gerphimum | im trying to understand the definition of a "channel"... if you have 1 fxo line, 2 fxs lines, and 3 voip phones, does that count at 6 channels ? |
| 20:19.00 | GuadamuX | that's correct |
| 20:19.47 | gerphimum | and i suppose a voip provider would count as a logical channel, as well |
| 20:20.46 | gerphimum | are there any differences in a "channel" on the asterisk side, or do you simply configure each to work with whatever hardware it is plugged in to |
| 20:21.09 | *** join/#asterisk GuadamuX (n=edgar@201.195.22.6) |
| 20:22.38 | GuadamuX | I'm looking for ways for Asterisk load balancing, I've heard about ARA and DUNDI, does somebody know about this or another asterisk clustering ways ?? |
| 20:22.58 | *** join/#asterisk aadilismail (n=aadilism@202.125.143.66) |
| 20:23.35 | aydiosmio | AIX trunk, DNS round robin |
| 20:23.42 | aydiosmio | IAX |
| 20:23.47 | aydiosmio | winnder |
| 20:27.46 | *** join/#asterisk xnon (i=xnon@200.8.30.31) |
| 20:29.06 | *** join/#asterisk xnon_ (i=xnon@200.8.30.31) |
| 20:30.00 | aadilismail | how can i dial further extensions... like if i dial through user agent .. 44208878XXXX... user agent ask to dial any extensions like 2020, 23 24 any ??? |
| 20:31.45 | aadilismail | mean after dialing the destination user agent able to dial extension?? how can i ? |
| 20:32.23 | GuadamuX | thanks, aydiosmio |
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| 20:33.01 | *** join/#asterisk xnon (i=xnon@200.8.30.31) |
| 20:33.38 | GuadamuX | aydiosmio, those ways provide redundancy? |
| 20:34.38 | *** join/#asterisk xnon_ (i=xnon@200.8.30.31) |
| 20:35.24 | [TK]D-Fender | GuadamuX : Your terminology is a little off. |
| 20:36.39 | [TK]D-Fender | gerphimum : A channel is any leg of a call regardless of what tech it uses. Each of your voip pohones could potentially support multiple calls. Each one would be a channel. |
| 20:38.13 | GuadamuX | well, I've chosen Asterisk as my Final Graduation Project, and I'm looking for the ways to build high-availability VoIP systems with Asterisk, so, I need to determine how make Asterisk redundant and load-balancing |
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| 20:40.39 | *** join/#asterisk SomethingISODD (n=dan@h109.42.63.69.cable.ottr.cablerocket.net) |
| 20:40.49 | SomethingISODD | Hello All if anyone is around can you please tell me whats wrong with this exten => s,8,GotoIfTime(09:00-16:30|mon-fri|*|*?office-open,s,9) |
| 20:41.48 | SomethingISODD | ?? |
| 20:45.37 | rg1_ | still working on trying to get the DID # that someone dialed to get into the system |
| 20:45.56 | rg1_ | I have the following 3 lines that start this off: |
| 20:45.58 | rg1_ | [from-trunk] ; just an alias since VoIP shouldn't be called PSTN |
| 20:45.58 | rg1_ | include => aa_default |
| 20:45.58 | rg1_ | include => ext-did ; defined in extensions_additional.conf, were incoming DIDs are mapped to internal extensions or auto-attendant |
| 20:45.58 | rg1_ | <PROTECTED> |
| 20:46.00 | rg1_ | include => from-pstn-reghours |
| 20:46.02 | rg1_ | ;include => from-pstn |
| 20:46.05 | hads | ~pb |
| 20:46.14 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 20:46.14 | rg1_ | sorry for the paste, but thought it might be small enough |
| 20:47.03 | rg1_ | anyway, what I want to do is capture the EXTEN (which should have the DID#) as the first thing in [fromt-trunk] |
| 20:47.17 | rg1_ | can i do that? |
| 20:47.57 | SomethingISODD | Please can anyone tell me whats wrong. |
| 20:48.12 | [TK]D-Fender | rg1_ : Sure. you should already BE int he exten (look at the pattern match). thenk just set another variable = ${EXTEN} and be on your merry way |
| 20:49.14 | rg1_ | something sodd - how about exten =>s,8,GotoIf(Time(09:00-16:30|mon-fri|*|*)?office-open,s,9) |
| 20:49.23 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-235-141.red.bezeqint.net) |
| 20:49.40 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 20:49.49 | SomethingISODD | ok let me try that it just keeps passing and going to the invailed extension |
| 20:50.19 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 20:50.24 | [TK]D-Fender | SomethingISODD : Have you checked to see if your clock is even right? |
| 20:50.30 | rg1_ | D-Fender, the zapata.conf sends the call to [from-trunk] for channels 1-23 |
| 20:50.50 | [TK]D-Fender | rg1_ : What kind of lines are those? |
| 20:50.54 | rg1_ | and [from-trunk] does those includes |
| 20:51.10 | rg1_ | D-Fender - what doyou mean? |
| 20:51.14 | SomethingISODD | [TK]D-Fender ya that was my first idea, but its the correct time and rg1_ your idea worked perfect |
| 20:51.19 | *** join/#asterisk xnon_ (i=xnon@200.8.30.31) |
| 20:51.24 | SomethingISODD | lol just need to fix the audio alittttle slow |
| 20:51.37 | SomethingISODD | sox -r 8000 filename.wav -c1 /location right?? |
| 20:51.57 | *** join/#asterisk metafore (n=john@adsl-68-127-185-145.dsl.pltn13.pacbell.net) |
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| 20:53.33 | rg1_ | D-Fender - what would my line look like in the dialplan to capture that "exten"? |
| 20:53.58 | [TK]D-Fender | rg1_ : Answer my previous question. What kind of interface are you working with? |
| 20:54.13 | rg1_ | PRI |
| 20:54.17 | rg1_ | digium |
| 20:54.31 | *** join/#asterisk xnon (i=xnon@200.8.30.31) |
| 20:54.52 | [TK]D-Fender | rg1_ : Well calls would come in under the exten in the context specified in your channel definition. |
| 20:55.20 | rg1_ | D-Fender - this is what I get when i call in from my cell phone - the "to" is the DID |
| 20:55.36 | rg1_ | -- Accepting call from '5094269931' to '5096873346' on channel 0/1, span 1 |
| 20:55.46 | rg1_ | so I'm trying to capture the 5096873346 |
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| 20:56.23 | *** part/#asterisk zgr (i=opera@213.145.38.166) |
| 20:56.24 | *** join/#asterisk xnon_ (i=xnon@200.8.30.31) |
| 20:56.33 | rg1_ | the 1st# is the callerid and the 2nd is the DID that it called - trying to get that 2nd number captured into a variable |
| 20:56.49 | [TK]D-Fender | rg1_ : pastebin your extensions.conf and your zapata.conf. You are clearly not getting the concept of ${EXTEN} |
| 20:57.04 | justinu|laptop | like fender said it's already in a variable... ${EXTEN} |
| 20:57.04 | rg1_ | ok, let me get that stuff up |
| 20:57.25 | rg1_ | ok, let me try one more thing.... |
| 20:57.31 | [TK]D-Fender | justinu|laptop : I'm sure you already know what I'm smelling right now.... |
| 20:57.53 | justinu|laptop | ;) |
| 20:59.04 | saftsack | hi |
| 20:59.15 | a2lti | hi |
| 20:59.31 | saftsack | are there any other enterprise interfacecard than sangomas and digiums? |
| 20:59.46 | *** join/#asterisk xnon (i=xnon@200.8.30.31) |
| 21:00.10 | trelane | saftsack, tons |
| 21:00.16 | [TK]D-Fender | saftsack : Rhino makes some, as well as a few others. Got a real question? |
| 21:00.19 | trelane | depends on what you're trying to interface |
| 21:00.19 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 21:01.16 | *** join/#asterisk xnon_ (i=xnon@200.8.30.31) |
| 21:01.45 | xheliox | TK: Speaking of asking questions without reading the manual, off the top of your head, is there a way to adjust the rtp port range that the Polycom's use? |
| 21:01.58 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 21:02.07 | [TK]D-Fender | xheliox : Yes |
| 21:02.19 | xheliox | Excellent. |
| 21:02.24 | xheliox | Now I'll go RTMF. |
| 21:02.46 | [TK]D-Fender | xheliox : Thats how you can get multiple phones working behind a remote NAT with some small measure of sanity |
| 21:03.02 | xheliox | TK: Yah. Exactly what I'm trying to do. |
| 21:03.25 | xheliox | TK: Still no stun support, right? You have to manually define the external IP, eh? |
| 21:03.28 | saftsack | [TK]D-Fender, yes i want to know if there are some cheap fxo and fxs cards available and if there are some minipci cards |
| 21:04.10 | a2lti | hey guys.. I'm trying dialing out via SIP. The goal is to play announcement when the user picks up.. no matter what I use Dial() invokes it after the SIP server answer,and not when 2 channels are bridged.. is there a workaround? |
| 21:04.13 | rg1_ | ok, let me ask this another way..........If i want a line to execute at the top of a context, regardless of what exten it will execute in the context normally, is there a way to do that? |
| 21:04.58 | xheliox | rg1: _.,Dial() |
| 21:05.07 | *** join/#asterisk xnon (i=xnon@200.8.30.31) |
| 21:05.08 | [TK]D-Fender | xheliox : Correct |
| 21:05.22 | hads | _. is silly |
| 21:05.36 | xheliox | hads: Is that not what he just asked for? |
| 21:05.43 | [TK]D-Fender | saftsack : Cheap FXO is only the X100, and its junk. You should already know about Digium & Sangoma's mainstream product line. |
| 21:05.56 | rg1_ | xheliox - ok, and if I put that in there it will execute and then will it also execute the more closely defined ext? |
| 21:06.14 | *** join/#asterisk xnon_ (i=xnon@200.8.30.31) |
| 21:06.20 | hads | xheliox: Yes, it is, but it's not a good solution. |
| 21:06.21 | xheliox | You could write a dialplan to do that, that won't do that by default. |
| 21:06.23 | *** join/#asterisk adorah (n=admin@84.94.146.135.cable.012.net.il) |
| 21:06.24 | saftsack | [TK]D-Fender, and whats about minipci cards? |
| 21:06.45 | xheliox | hads: I don't think he's interested in good solutions if he's willing to just let us guess at it for him. |
| 21:07.04 | hads | xheliox: Fair enough :) |
| 21:07.07 | rg1_ | xheliox/anyone - for example, if the ext=5096873340 and my dialplan has a context [ext-did] |
| 21:07.08 | *** part/#asterisk jbroome (n=jbroome@unaffiliated/jbroome) |
| 21:07.15 | rg1_ | so like this: |
| 21:07.19 | rg1_ | [ext-did] |
| 21:07.39 | rg1_ | exten => _.,1,Dial() |
| 21:07.59 | rg1_ | exten => 5096873340,1,Goto(...) |
| 21:08.10 | rg1_ | will BOTH the _. AND the 509... execute? |
| 21:08.14 | xheliox | No. |
| 21:08.42 | [TK]D-Fender | saftsack : nothing I've heard of |
| 21:08.59 | saftsack | [TK]D-Fender, do you think it is cheaper to build up a gateway with a cheap soekris board or with a buyed gateway |
| 21:09.18 | saftsack | hmm ok. strange .... because in germany i can buy a BRI minipci card |
| 21:09.49 | [TK]D-Fender | saftsack : You can come up with that answer yourself with about 5 minutes of research and Elementary school math..... |
| 21:09.58 | *** join/#asterisk xnon (i=xnon@200.8.30.31) |
| 21:10.03 | hads | There's probably not a lot of point in building a basic gateway when you can buy them off the shelf. |
| 21:10.06 | saftsack | http://www.junghanns.net/en/quadBRImini_produkt.html |
| 21:10.17 | saftsack | [TK]D-Fender, yes if i would know any prices |
| 21:11.24 | *** join/#asterisk xnon_ (i=xnon@200.8.30.31) |
| 21:12.04 | [TK]D-Fender | <PROTECTED> |
| 21:12.07 | rg1_ | so is there VAR like CALLERIDNUM that is set for the DID number that was dialed? |
| 21:12.37 | saftsack | [TK]D-Fender, what is jfgi? |
| 21:12.37 | hads | EXTEN, as has been mentioned. |
| 21:12.49 | rg1_ | yes, but that changes |
| 21:12.54 | rg1_ | as you go to a new context |
| 21:13.04 | [TK]D-Fender | rg1_ : Ok, now you've really shown the allergy to what you should have been doing for some time : |
| 21:13.05 | saftsack | [TK]D-Fender, i mean google doesnt throw out common gateways |
| 21:13.07 | [TK]D-Fender | ~rtfm |
| 21:13.11 | jbot | methinks rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM |
| 21:13.11 | [TK]D-Fender | ~book |
| 21:13.12 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:13.13 | [TK]D-Fender | ~wikis |
| 21:13.14 | jbot | i guess wikis is http://www.voip-info.org |
| 21:13.44 | [TK]D-Fender | saftsack : www.voip-supply.com , www.atacomm.com , www.telephonydepot.com , www.voxilla.com |
| 21:13.56 | saftsack | [TK]D-Fender, thx |
| 21:14.22 | [TK]D-Fender | saftsack : For crying out loud EVERYBODY knows the common VoIP equipment retailers, and they're all listed nicely on the WIKI. |
| 21:14.52 | saftsack | yes i saw the links on the wiki but i live in germany :-P |
| 21:15.03 | saftsack | didnt find any voip stuff distributor in germany :( |
| 21:16.16 | [TK]D-Fender | ~jfgi |
| 21:16.17 | jbot | somebody said jfgi was http://www.justf*ckinggoogleit.com/ |
| 21:20.07 | linagee | ~asterisk |
| 21:20.08 | jbot | well, asterisk is the best free PBX in the world |
| 21:20.18 | sevard | that's objective. |
| 21:20.33 | linagee | free PBX? *grin* |
| 21:20.34 | linagee | freePBX! :) |
| 21:22.19 | [TK]D-Fender | linagee : "well, asterisk is best in the PBX-free world" <- |
| 21:25.30 | [TK]D-Fender | (as read through chan_dyslexia.so) |
| 21:25.30 | EmleyMoor | I am glad I've found out that asterisk will accept pulse dial phones |
| 21:26.08 | saftsack | what is the better choice? a 24 port fxs gateway with a t1 card with hardware ec or a digium tdm2400p card with hardware ec? |
| 21:26.15 | *** join/#asterisk gerphimum (n=trekkie@cpe-68-206-83-62.satx.res.rr.com) |
| 21:26.45 | gerphimum | does anyone know any information regarding particular landline service providers that are offering sms capability |
| 21:27.28 | [TK]D-Fender | saftsack : What 24 port FXS gateway needs a T1 card? Sounds more like a CHANNEL BANK. |
| 21:28.12 | [TK]D-Fender | saftsack : And to help answer that maybe you could be precise as to whether you actually NEED 24 analog channels (why on earth would you?) |
| 21:28.41 | xheliox | TK: Residential deployments? |
| 21:29.43 | [TK]D-Fender | xheliox :His reason, not yours :) |
| 21:30.15 | xheliox | Oh. ;( |
| 21:30.56 | linagee | who uses pulse dial phones??? |
| 21:31.06 | gerphimum | my grandma :| |
| 21:31.37 | linagee | gerphimum: buy her a $10 walmart DTMF phone. :p |
| 21:31.49 | linagee | or better yet, something with a brain that talks SIP. :) |
| 21:32.13 | linagee | gerphimum: buy your grandma a barbietone. she will hate you for years to come. :} |
| 21:32.18 | gerphimum | the only form of excercise my grandma has is when shes moving that old school rotary phone... thingy... |
| 21:32.25 | EmleyMoor | linagee: I happen to like the Yeoman as a phone but tone dial ones are rare |
| 21:32.38 | linagee | Yeoman? |
| 21:32.42 | EmleyMoor | I have a wall DTMF version that will be going in the hall |
| 21:32.45 | [TK]D-Fender | linagee : SIP does not have a distict interface from the concept of DTMF and dial-tone. Everything else is phone specific |
| 21:33.06 | EmleyMoor | The Yeoman was the common British Post Office telephone from 1967 until 1981 |
| 21:33.07 | saftsack | [TK]D-Fender, sorry i misstyped it. i mean a channelbank, thats right |
| 21:33.08 | linagee | [TK]D-Fender: having a rotary SIP phone would just be creepy |
| 21:33.14 | gerphimum | rofl tru dat.. |
| 21:33.22 | saftsack | [TK]D-Fender, maybe if i have 24 old analog telephones? :> |
| 21:33.39 | EmleyMoor | I have a rotary Yeoman with amplifier that I might have by my bed |
| 21:33.43 | gerphimum | linagee >> that phone has fared my grandma through the ages.. the one and only thing that has.. i cant tear that away from her |
| 21:34.12 | linagee | gerphimum: scrape out the insides of the phone with an icecream scoop and put a SIP brain into it. |
| 21:34.16 | sevard | [TK]D-Fender: that'll be five dollars, please. |
| 21:34.27 | gerphimum | linagee >> ah, im much too lazy for all of that. |
| 21:34.48 | [TK]D-Fender | saftsack : in that case you definately don't want ANY PCI cared involved at all. I'd suggest a Mediatrix 1124 gateway, or an AudioCodes MP-124 |
| 21:35.02 | [TK]D-Fender | sevard : For? |
| 21:35.19 | asdx | so, asterisk is a system for incoming calls and pass them trough other phones? |
| 21:35.22 | sevard | [TK]D-Fender: my love -- you can paypal it. |
| 21:35.28 | asdx | incoming/outcoming |
| 21:35.48 | [TK]D-Fender | sevard : Don't pay for vapourware, sorry.... |
| 21:35.48 | linagee | asdx: no. asterisk is like a PBX on crack |
| 21:35.51 | sevard | asdx: asterisk is a PBX, please wiki 'pbx' |
| 21:35.55 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 21:35.57 | asdx | ok |
| 21:36.02 | linagee | sevard: asterisk is a PBX on crack. :) |
| 21:36.08 | asdx | i don't know what PBX means |
| 21:36.24 | linagee | asdx: something that only rich businesses have. :) |
| 21:36.30 | asdx | lol |
| 21:36.32 | gerphimum | not anymore! |
| 21:36.56 | sevard | asdx: that's why I said "please wiki 'pbx'" http://en.wikipedia.org/wiki/Pbx |
| 21:37.03 | asdx | sevard: ok |
| 21:37.04 | EmleyMoor | Private Branch eXchange |
| 21:37.14 | linagee | linux is windows on crack. asterisk is a PBX on crack. |
| 21:37.15 | asdx | what does private branch exchange means? lol |
| 21:37.34 | linagee | asdx: you've ever had to dial 9 to get an outside line? |
| 21:37.42 | asdx | linagee: yes |
| 21:37.45 | linagee | asdx: you're dialing 9 at a PBX. |
| 21:37.56 | asdx | linagee: i see |
| 21:38.00 | linagee | asdx: it's the box that says, "oh. you dialed a 9. i will do this and this and this." |
| 21:38.18 | linagee | asdx: and more |
| 21:38.22 | gerphimum | lots.. more. |
| 21:38.34 | linagee | gerphimum: well like i said. a PBX on crack. :) |
| 21:38.37 | asdx | linagee: and the call is passing trough the PBX? |
| 21:38.41 | gerphimum | there ya go. fitting definition. |
| 21:38.46 | sevard | asdx: if you had read that article instead of just the title of the article you would understand what 'private branch exchange' meant |
| 21:39.01 | asdx | sevard: ok, i'll read now |
| 21:39.20 | sevard | asdx: after you read that you should read the book |
| 21:39.21 | gerphimum | linagee >> do you know anything about landline sms support ? |
| 21:39.22 | sevard | ~thebook |
| 21:39.25 | jbot | rumour has it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:39.25 | asdx | sevard: i have read that article but i didn't understand very well... i'll read again |
| 21:39.32 | asdx | sevard: i'm reading the book, i'm in chapter 2 |
| 21:39.36 | linagee | gerphimum: a 1950s PBX can't look up a value in a database when you punch in numbers into the keypad and use text to speech to reach you back some response |
| 21:39.37 | linagee | :) |
| 21:39.47 | gerphimum | this is true. |
| 21:39.47 | linagee | s/reach/read/ |
| 21:40.01 | gerphimum | then again, in the 1950s, crack wasnt very popular. |
| 21:40.06 | linagee | LOL |
| 21:40.15 | gerphimum | ;) |
| 21:40.33 | asdx | all this sounds so great... |
| 21:40.43 | awannabe | it doesnt sound great, it is!!! |
| 21:40.46 | asdx | and exciting |
| 21:40.53 | linagee | asdx: imagine you can define the routes for calls to take just like an internet router defines the routes for web traffic to take. |
| 21:40.53 | asdx | xD |
| 21:41.14 | asdx | linagee: yeah.. :) |
| 21:41.15 | linagee | asdx: asterisk : phones :: router : networks |
| 21:41.42 | gerphimum | pbx : phones :: router :: computers * |
| 21:41.43 | linagee | if you can read my strange nomenclature, give yourself a gold star |
| 21:41.53 | gerphimum | nt on the analogy though |
| 21:41.54 | gerphimum | ;) |
| 21:41.56 | linagee | gerphimum: bah. :p |
| 21:42.13 | linagee | gerphimum: i'm not sure how to interpret your second :: |
| 21:42.18 | gerphimum | ... |
| 21:42.20 | gerphimum | <.< |
| 21:42.35 | gerphimum | id interpret it as a glitch in your monitor... and leave it at that |
| 21:42.41 | linagee | pbx is to phones as routers as computers? |
| 21:43.02 | gerphimum | as so definied in the movie serenity |
| 21:43.19 | gerphimum | ahhhhh yes. |
| 21:43.30 | linagee | er, standby system |
| 21:43.36 | gerphimum | so then, anyone know anything about sms support from landline carriers ? |
| 21:43.53 | linagee | gerphimum: why use a PSTN gateway when you can use an email gateway? |
| 21:44.07 | gerphimum | cuz i dont know about sms via email |
| 21:44.14 | gerphimum | is it possible ? |
| 21:44.14 | linagee | gerphimum: standby as in backup/secondary/hot spare. not as in sleeping your computer. :p |
| 21:44.25 | gerphimum | ah, ok. thats cool then |
| 21:44.34 | linagee | gerphimum: yes. you find the sms email gateway you want to send to and send it |
| 21:44.41 | linagee | usually like phonenumber@company.com |
| 21:44.55 | linagee | you will have to go to company.com for the specifics. |
| 21:45.06 | saftsack | [TK]D-Fender, why is a gateway better than a channelbank + gateway card? can you tell me the reasons? |
| 21:45.10 | linagee | if you want a general SMS gateway that will send to any cell, i believe there are pay gateways. :P |
| 21:45.23 | gerphimum | true, but what if you dont know the company... you just want to send a message to say +12345678901 |
| 21:45.33 | linagee | gerphimum: use a pay gateway |
| 21:45.41 | linagee | gerphimum: they will do the routing for you. :p |
| 21:46.21 | gerphimum | i read on the wiki that some landline carriers are making sms a feature, assuming you have the ability to send an sms signal by means of some device on your household phone network |
| 21:46.41 | gerphimum | besides, who wants to pay for stuff these days |
| 21:46.59 | [TK]D-Fender | saftsack : First you get better call control without using stupid * DTMF options. Second, they are easier to deploy and don't place the kind of load that Zaptel does. This means you can have more attached than you can have PCI cards allow for |
| 21:47.03 | linagee | gerphimum: you could hook a cellphone up to an asterisk box and send messages to it which get sent to the network. |
| 21:47.08 | linagee | gerphimum: albeit cludgy |
| 21:47.25 | [TK]D-Fender | saftsack : Better wiring options since you aren't distance limited. the list goes on and on. |
| 21:47.30 | rg1_ | new issue/new question....... |
| 21:47.35 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 21:48.07 | rg1_ | in the dialplan you can get a value like this " ${CALLERID(num)} " - how would you get that same value in an AGI script (using PHP) |
| 21:48.26 | linagee | gerphimum: you do txting, you pay. :p |
| 21:48.34 | linagee | gerphimum: unless you live in japan or something |
| 21:49.02 | linagee | some civilized country where <512 bytes of data are free |
| 21:49.24 | saftsack | [TK]D-Fender, ok thank you :> |
| 21:49.27 | gerphimum | linagee >> quite. what i was thinking is... say im having a party, and people are calling my house for directions.. my phone will instantly be on party mode so that it automatically picks up the phone and goes through regular information like when the party is and whatnot... i want asterisk to be able to send a text message to the caller including directions to my house |
| 21:49.40 | saftsack | so overall a decentralised built up is better in many ways? |
| 21:49.53 | linagee | gerphimum: why not have them called up and delivered a message instead? |
| 21:50.00 | linagee | gerphimum: why does it have to be txted? |
| 21:50.04 | gerphimum | because i suck at remembering shit |
| 21:50.10 | linagee | ?? |
| 21:50.11 | [TK]D-Fender | saftsack : you can also set up redundant call-servers so if your primary * box goes down it can forward to another,. |
| 21:50.20 | linagee | gerphimum: i'm saying why txt over voice |
| 21:50.25 | linagee | s/over/instead of/ |
| 21:50.29 | gerphimum | txt message you can look back when you forget |
| 21:50.38 | gerphimum | like i always do |
| 21:50.49 | linagee | gerphimum: why not WML then? :) |
| 21:50.53 | gerphimum | wml ? |
| 21:50.57 | [TK]D-Fender | saftsack : Also you don't need to worry about EC much with most gateways so your cost-per-port is pretty low as well. |
| 21:50.58 | saftsack | [TK]D-Fender, but now another view. if i would buy 24 fxs ata's it would cost 1200 which is cheaper than a pri fxs gateway .... :> |
| 21:51.03 | linagee | gerphimum: wireless markup. |
| 21:51.09 | linagee | gerphimum: web phone |
| 21:51.17 | saftsack | with an ata my cost per port is at about 60bucks |
| 21:51.40 | linagee | gerphimum: most cellphones these days support it |
| 21:52.13 | gerphimum | are you talking about a cure for my forgetfulness, or a way to deliver a message to those requiring it |
| 21:52.19 | linagee | gerphimum: "show people where the party is at" = "send out mass spam to lots of phones" |
| 21:52.20 | linagee | lol |
| 21:52.27 | [TK]D-Fender | saftsack : Why are we talking about PRI now? thats for PSTN, not for PHONES. And I was referring you to 2 different 24-port gateways. You COULD just get 12 2-port FXS ATA's which would even cheaper, but the wiring would be a hell of a mess. |
| 21:52.43 | gerphimum | lol ;) brb, need a snack now. all this knowledge is making me hungry |
| 21:52.44 | [TK]D-Fender | saftsack : not to mention configuring a dozen stupid little boxes. |
| 21:53.00 | linagee | gerphimum: look into gnokii. http://www.gnokii.org/ |
| 21:53.10 | linagee | gerphimum: remote control a phone. can send out txt messages too |
| 21:53.22 | [TK]D-Fender | saftsack : Maybe you should actuallt describe a REAL scenario you are looking to satify... |
| 21:53.30 | linagee | saftsack: balls? |
| 21:54.04 | *** join/#asterisk xnon (i=xnon@200.8.30.31) |
| 21:54.16 | saftsack | [TK]D-Fender, i want to find out if its cheaper if i have a voip telephone or an analog telephone + channel costs. |
| 21:54.37 | linagee | saftsack: it's cheaper to have voip telephones |
| 21:54.50 | linagee | saftsack: buying analog telephones means you will have to pay double |
| 21:55.02 | linagee | saftsack: because in the future you will have to buy voip phones |
| 21:55.10 | gerphimum | unless his house isnt wired for cat5 :| |
| 21:55.20 | linagee | gerphimum: then wire it for cat 5. :) |
| 21:55.33 | linagee | gerphimum: nothing a drill can't fix |
| 21:55.46 | gerphimum | with 24 phones, his house must be huge, and that project alone might cost more than what the voip phones would cost |
| 21:56.13 | linagee | gerphimum: i'm thinking small call center instead of a residence. |
| 21:56.14 | gerphimum | then again, cat5 is the wave of the future, rj11 is worthless these days |
| 21:56.22 | linagee | gerphimum: exactlyh. |
| 21:56.36 | gerphimum | cat5 would be a good investment in the long run |
| 21:56.36 | linagee | gerphimum: you can put video over cat5... voice over cat 5... |
| 21:56.39 | saftsack | linagee, why do i need voip phones in the future? i mean a softpbx is flexible and the telephones behind it are not interesting for reaching the pstn |
| 21:56.46 | linagee | gerphimum: directly or over ethernet packets even. :) |
| 21:57.03 | linagee | saftsack: caller ID? |
| 21:57.33 | linagee | saftsack: regular phones don't show caller ID. they are also lower quality. once you get fed up with that, you'll shell out for voip phones in the future and end up paying twice. |
| 21:58.07 | [TK]D-Fender | VoIP is a nifty idea, but mearly a means to an end. There is little need for more functionality that you can get out of an analog phone for anyone who isn't in a busy call environment. |
| 21:58.14 | linagee | saftsack: if you're going to buy 24 of something, might as well be of the best quality per dollar to last the longest. |
| 21:58.16 | saftsack | ok accepted but i had the best voice quality with an analogue phone. it is a little bit better than my snom phone |
| 21:58.48 | linagee | saftsack: maybe your snom phones just sucked. what protocol? PCM? |
| 21:58.52 | [TK]D-Fender | saftsack : Snom sucks. Should have gotten a Polycom or Cisco. |
| 21:58.56 | asdx | switching calls, etc, are all doing with a pbx system? |
| 21:59.06 | gerphimum | asdx >> yessir |
| 21:59.13 | asdx | I see |
| 21:59.13 | hads | That's a bit rough [TK]D-Fender |
| 21:59.21 | awannabe | the snoms rock |
| 21:59.28 | awannabe | more featutres, cheaper |
| 21:59.53 | saftsack | i heard that the using of snoms in europa isnt as good as in america in contrast to snoms .... |
| 22:00.12 | [TK]D-Fender | hads : You right. SNOM IS CHEAP CRAP MARKETED BY THE FACT THEY RUN LINUX. *WHOOPIE!* there... now my intonation is clearer ;) |
| 22:00.12 | asdx | and you can call from one phone to another phone easily inside the pbx, without using a external line? |
| 22:00.18 | awannabe | take up WAY WAY less space when using as a recposnist phone |
| 22:00.24 | saftsack | what is with ciscos? is the cisco sip firmware as good as the sccp firmware? |
| 22:00.31 | gerphimum | asdx >> yessir.. |
| 22:00.32 | linagee | [TK]D-Fender: what is better cheap crap? snom or gxp2000? :-D |
| 22:00.34 | awannabe | the cisco sip firmware sucks |
| 22:00.36 | asdx | gerphimum: cool |
| 22:00.43 | justinu|laptop | snom 360 is a decent phone... kinda funky tho |
| 22:00.45 | hads | I have a lot of customers that are very happy with the snoms, espcially when compared the Polycom phones at twice the price over here. |
| 22:00.45 | awannabe | they have no features on them!! snoms have backlit display! |
| 22:00.51 | *** join/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 22:00.52 | justinu|laptop | audio quality isn't up to polycom standards |
| 22:00.53 | [TK]D-Fender | saftsack : Better jitter buffers, PLC, quality mics/speakers, etc. |
| 22:00.55 | awannabe | hads, agreed, im installing 90 right now |
| 22:01.19 | [TK]D-Fender | awannabe : For those who insist on backlight I suggest the Aastra 480i |
| 22:01.27 | linagee | use a PC buzzer as a speaker and sell the product for $5. lol |
| 22:01.28 | awannabe | ive never used that one |
| 22:01.34 | linagee | that's cheap crap |
| 22:01.39 | Ryanw | where can i get a winamp plugin to play .ulaw files or whats the name of a windows sound player that will play them? |
| 22:01.46 | awannabe | im putting a group of snoms in 4 doctrs offices |
| 22:02.04 | [TK]D-Fender | hads : yeah, maybe where you are Polycom is more expensive, but you get what you pay for. In North America they are almost on par with each other though. |
| 22:02.28 | asdx | i kinda understand what is a PBX now :) |
| 22:02.28 | [TK]D-Fender | hads : Then you could always choose Linksys. They're not bad overall (for the dollar where you are) |
| 22:02.37 | gerphimum | Ryanw >> http://www.downloadjunction.com/product/software/62345/index.html google is a one stop shop for all things random |
| 22:02.47 | *** join/#asterisk BosHaus (i=nobody@pool-71-164-156-136.dllstx.fios.verizon.net) |
| 22:03.06 | hads | [TK]D-Fender: I agree about the get what you pay for. Linksys are OK, less feature rich with Asterisk at present though, good pricing though. |
| 22:03.09 | saftsack | [TK]D-Fender, is it possible to get a polycom/cisco phone for about 130bucks? |
| 22:03.17 | gerphimum | saftsack >> ebay |
| 22:03.27 | [TK]D-Fender | hads : My assessment exactly. |
| 22:03.37 | linagee | saftsack: yes. off of ebay. :) |
| 22:03.40 | [TK]D-Fender | hads : But it depends on your needs. |
| 22:03.42 | linagee | get it for $0.99! |
| 22:03.56 | [TK]D-Fender | saftsack : Polycom IP 301 = $115 USD |
| 22:03.59 | hads | The Polycom distributors here are working hard with Polycom to try and get the pricing down to a decent point here. |
| 22:04.16 | awannabe | the 301 is cheasy..not even a full duplex speaker phone! |
| 22:04.22 | hads | I think that is because they want to actually sell something :) |
| 22:04.29 | [TK]D-Fender | awannabe : Thats its only downside really. |
| 22:04.32 | linagee | hads: a distributor trying to lower the price? that's got to be a first. lol |
| 22:04.50 | [TK]D-Fender | awannabe : Thats why I typically suggest the IP 430 for PoE-only installs, or the IP 501 otherwise |
| 22:04.51 | linagee | sounds like marketing fodder |
| 22:04.59 | awannabe | thats why i cant use them, people these day LOVE speakerphone heh |
| 22:05.06 | saftsack | [TK]D-Fender, does the 301 provide the same voice quality as its big brother? |
| 22:05.32 | hads | linagee: Na, they are actually really nice. They've already come back once with some more competitive volume pricing. |
| 22:05.33 | linagee | no full duplex speakerphone? wow. that's worse than a barbietone! |
| 22:05.39 | awannabe | LOL |
| 22:06.38 | *** join/#asterisk adorah (n=admin@84.94.204.41.cable.012.net.il) |
| 22:08.14 | asdx | so if i have 1 telephone line, i can plug it to one computer and with asterisk i can multiplex that line and have like 100 telephones? |
| 22:08.28 | gerphimum | asdx >> theoretically. |
| 22:08.31 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 22:08.36 | asdx | nice |
| 22:08.38 | asdx | =] |
| 22:08.55 | [TK]D-Fender | saftsack : From what I've seen, yes. |
| 22:09.01 | gerphimum | asdx >> theres a bit more to it, though. you cant just use some random 56k modem thats lying around |
| 22:09.35 | asdx | gerphimum: i see |
| 22:09.50 | [TK]D-Fender | asdx : Something like that. You can have 1 line come into *, and have 100 PHONES as well, but only 1 of them is going to use the line at a time. |
| 22:10.08 | [TK]D-Fender | saftsack : How many phones do you need, and what kind of call volume? |
| 22:10.39 | asdx | [TK]D-Fender: i see |
| 22:11.09 | asdx | [TK]D-Fender: but i can pass the call to any of the 100 phones right? |
| 22:11.10 | gerphimum | asdx >> yes. |
| 22:11.16 | asdx | i understand |
| 22:11.17 | asdx | :) |
| 22:11.43 | gerphimum | asdx >> with your incoming phone line, you will need to plug that into whats called an FXO card. that card takes and is able to use (incoming and outgoing) the line back to the pstn (public switched telephone network) |
| 22:12.09 | BosHaus | do you have to have a phone line or t1 to get asterisk to work? or can I host my own phone system through my network? |
| 22:12.28 | BosHaus | got plenty of bandwidth.. fiber optic. |
| 22:12.30 | [TK]D-Fender | asdx : * can direct all calls arriving to it from any interface just about any way you want it to. |
| 22:12.53 | asdx | wow |
| 22:12.55 | RaYmAn-Bx | BosHaus: it is certainly possible to use asterisk without any PSTN or t1 phone lines |
| 22:12.56 | asdx | amazing |
| 22:12.57 | [TK]D-Fender | BosHaus : You can use any kind of interface you want. You could run pure-voip if you want. |
| 22:12.58 | gerphimum | BosHaus >> you can configure asterisk to work internally only.. |
| 22:13.10 | asdx | can i answer the phone call in my computer? |
| 22:13.27 | asdx | with a softphone |
| 22:13.53 | BosHaus | so if I did pure-voip, who would I have to pay for the PSTN |
| 22:13.54 | gerphimum | asdx >> yes. you would use whats called a softphone, which is a program that runs on your computer and registers itself with asterisk. |
| 22:14.07 | asdx | i see |
| 22:14.09 | gerphimum | BosHaus >> whoever is your provider |
| 22:14.16 | hads | BosHaus: A provider. |
| 22:14.22 | BosHaus | ah, ok |
| 22:14.29 | asdx | how much does it cost a FXO card? |
| 22:14.38 | hads | asdx: 42 |
| 22:14.45 | gerphimum | asdx >> google tdm400p |
| 22:14.55 | JT | digium.com |
| 22:15.08 | asdx | hads: US dollars? |
| 22:15.20 | JT | not for a tdm400P :] |
| 22:15.38 | gerphimum | tdm400p is upwards to about $200, depending on what configuration you get |
| 22:15.40 | asdx | can't wait to buy a card and play with it... |
| 22:16.39 | gerphimum | same here :( |
| 22:16.58 | saftsack | does anybody of you know a european voip stuff seller? |
| 22:17.17 | brookshire | which country? |
| 22:17.22 | [TK]D-Fender | saftsack : jfgi |
| 22:17.39 | asdx | and how do you connect the voip phone into the computer, trough the card? |
| 22:18.06 | JT | if it's a voip phone... ethernet |
| 22:18.24 | brookshire | you use the network |
| 22:18.29 | gerphimum | asdx >> no, you would connect the voip phone the same you would a computer. it would go through the router and into the asterisk box |
| 22:18.29 | asdx | i see |
| 22:18.45 | asdx | nice :) |
| 22:18.45 | JT | gerphimum: there is no need for a router |
| 22:18.57 | JT | in most networks it would be a network switch in place |
| 22:19.01 | asdx | so, the voip phone is just like a computer/interface? |
| 22:19.07 | asdx | computer, interface, client |
| 22:19.09 | JT | if it's voip, yess |
| 22:19.10 | gerphimum | JT >> how would they get ip addresses |
| 22:19.12 | hads | JT: Everything that you plug a network cable into is a router these days. |
| 22:19.20 | JT | voice over INTERNET PROTOCOL |
| 22:19.23 | hads | :) |
| 22:19.24 | JT | hads: rofl |
| 22:19.36 | asdx | nice, this can't be more AWESOME! :D |
| 22:19.40 | asdx | can't wait for try it out |
| 22:19.50 | JT | gerphimum: you can assign them statically or you can have it automatically provided by a DHCP server, no need for a router |
| 22:20.13 | saftsack | [TK]D-Fender, there arent any good sellers listed. |
| 22:20.15 | brookshire | and even provision the phones with dhcp + tftp |
| 22:20.17 | saftsack | ^^ |
| 22:20.37 | asdx | i wish i could switch all the telephony stuff to asterisk in my house xD |
| 22:20.49 | gerphimum | asdx >> you can.. |
| 22:20.56 | gerphimum | :) |
| 22:20.57 | JT | you can, it's just a question of dollars and time |
| 22:20.58 | gerphimum | you just gotta BELIEVE |
| 22:21.06 | gerphimum | and, what JT said |
| 22:21.13 | asdx | i will, i just need the $ for buy all the stuff ;p |
| 22:21.30 | brookshire | so anyone played around with 1.4 yet? |
| 22:21.43 | gerphimum | asdx >> to* buy all the stuff :) |
| 22:21.50 | asdx | gerphimum: yeah ;p |
| 22:21.50 | gerphimum | not for |
| 22:22.07 | asdx | gerphimum: also, thanks for correct me ;p |
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| 22:32.06 | [TK]D-Fender | *b00m* |
| 22:32.06 | awannabe | wow |
| 22:32.07 | asdx | if i have two asterisk pbx running, and the two asterisk pbx are in different countries, and the computers are connected trough the internet, i can call with my phone to the other pbx's? |
| 22:32.07 | JT | yes |
| 22:32.07 | Dovid | yes |
| 22:32.07 | gerphimum | .. looks like yes |
| 22:32.07 | asdx | cool :) |
| 22:32.07 | Dovid | ~trunking |
| 22:32.09 | Dovid | see the wiki |
| 22:32.10 | Dovid | ~wiki |
| 22:32.11 | asdx | this is indeed the future :D |
| 22:32.11 | JT | asterisk is pretty much just a telephony toolkit |
| 22:32.11 | JT | what you do with it, is up to you |
| 22:32.11 | BosHaus | so if I were to go with a pure-voip system, what kind of service would I have to buy from a provider? |
| 22:32.11 | JT | you can use voip protocols including SIP, IAX, H.323, SCCP and a couple of others |
| 22:32.11 | Dovid | depends on what u want to do with it |
| 22:32.11 | JT | SIP or IAX telephony service usually, BosHaus |
| 22:32.12 | asdx | what about if you have two pbx's, and two or more phones, can you call to another pbx and have more than 1 user talking at the same time over the pbx's? |
| 22:32.12 | asdx | like, multiplexing... |
| 22:32.12 | Dovid | asdx: yes |
| 22:32.12 | Dovid | read the book |
| 22:32.12 | JT | yes |
| 22:32.12 | Dovid | ~book |
| 22:32.13 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:32.13 | asdx | wow :) |
| 22:32.14 | JT | these are pretty simple questions |
| 22:32.14 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 22:32.14 | Dovid | wow - bot a lil slow |
| 22:32.14 | asdx | i'm drawing it all in paper as i see in my mind now :P |
| 22:32.14 | JT | assume it's yes unless it's actually something difficult |
| 22:32.14 | JT | bot is always slow, must be hosted on 14kbps modem |
| 22:32.14 | asdx | is really fun |
| 22:32.15 | gerphimum | just hit a huge netsplit |
| 22:32.15 | gerphimum | maybe it didnt come back |
| 22:32.15 | Dovid | ah ok |
| 22:32.15 | JT | it's here |
| 22:32.15 | JT | just wait |
| 22:32.15 | Dovid | ah ok |
| 22:32.15 | Dovid | i will give it time to rest up |
| 22:32.16 | JT | ssh it will coming back with 6 lines about 3 times now :P |
| 22:32.16 | [TK]D-Fender | JT : 14kbps is PLENTY for IRC..... |
| 22:32.52 | [TK]D-Fender | JT : how fast do YOU see chars flying around here? |
| 22:32.52 | JT | [TK]D-Fender: with someone leeching .isos off the link at the same time :P |
| 22:32.52 | [TK]D-Fender | JT : That would do it :) |
| 22:32.54 | JT | that bot is incredibly slow, i want to know what third world connection it has |
| 22:33.43 | [TK]D-Fender | JT : Actually the third world has BETTER tech than most of us because they didn't/coun't invest in the earlier generateions and are thus unburdened by it in their new expansions. |
| 22:33.54 | [TK]D-Fender | JT : for those buying any tech at all taht is. |
| 22:34.11 | JT | i think you'll find that really depends on the part of the third world you're talking about |
| 22:34.29 | [TK]D-Fender | JT : America would be much further ahead were it not burdened by trying to maintain its ancient infrastructure for so long. |
| 22:34.43 | [TK]D-Fender | JT L Quite true, and i did state that. |
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| 22:36.04 | JT | heh |
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| 22:36.39 | *** join/#asterisk lilalinux (i=e-trolle@langweiligneutral.deswahnsinns.de) |
| 22:36.49 | lilalinux | hey guys |
| 22:36.51 | gerphimum | hi. |
| 22:37.27 | lilalinux | is there a possibility to connect my Gigaset 4000 (ISDN) to asterisk and route calls to VoIP? Internal S0? |
| 22:37.44 | [TK]D-Fender | lilalinux : Yes. |
| 22:38.00 | lilalinux | [TK]D-Fender: do you have a starter? |
| 22:38.22 | asdx | does voip phones runs operating systems? |
| 22:38.45 | lilalinux | asdx: I know some, that use embedded linux |
| 22:38.52 | asdx | cool |
| 22:38.56 | asdx | :) |
| 22:39.15 | asdx | lilalinux: what are these ones? |
| 22:39.50 | lilalinux | asdx: e.g. snom |
| 22:40.13 | asdx | lilalinux: thanks |
| 22:40.47 | asdx | the phone looks pretty cool :) |
| 22:40.48 | SomethingISODD | Hey all question accounting to my server its now 17:30 and i have this exten => s,8,GotoIf(Time(09:00-16:30|mon-fri|*|*)?office-open,s,9) |
| 22:40.55 | SomethingISODD | and its still going to the open hours |
| 22:41.00 | saftsack | what is the better conecpt? connecting via a e1 card to the pstn or with a gateway? |
| 22:41.02 | JT | lilalinux: if it's a standard ISDN TE phone, just get an isdn card |
| 22:41.18 | JT | saftsack: do you mean "sip gateway"? |
| 22:41.28 | saftsack | PRI -> SIP |
| 22:41.37 | SomethingISODD | ? |
| 22:41.45 | asdx | how much does a voip phone cost, like the snom one |
| 22:41.56 | [TK]D-Fender | SomethingISODD : Your formatting is off for that function call. |
| 22:42.06 | lilalinux | JT: It's a dect base station |
| 22:42.11 | [TK]D-Fender | asdx : Cost depends on where you are, and what models you are looking at. |
| 22:42.27 | asdx | ok |
| 22:42.32 | JT | lilalinux: if it uses ISDN SO bus in a standard fashion, you can connect to it with the right isdn card |
| 22:42.43 | SomethingISODD | [TK]D-Fender well the way i had it was exten => s,8,GotoIfTime(09:00-16:30|mon-fri|*|*?office-open,s,9) |
| 22:42.45 | JT | saftsack: your question was unclear |
| 22:42.51 | lilalinux | JT: can you recommend one? |
| 22:42.52 | [TK]D-Fender | asdx : I woul dsuggest you start with an ATA likst the Linksys SPA-2002 |
| 22:43.02 | JT | lilalinux: how many ports do you require? |
| 22:43.06 | asdx | [TK]D-Fender: ok :) |
| 22:43.24 | lilalinux | JT: standard euro isdn |
| 22:43.27 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:43.28 | lilalinux | 2 bchannels |
| 22:43.29 | SomethingISODD | is that correct? |
| 22:43.32 | [TK]D-Fender | SomethingISODD : change it back then, and then pastebin your dialpland and the CLI output of its exectution, as well as a proof of the date/time set on your server |
| 22:43.52 | SomethingISODD | ok |
| 22:43.59 | JT | lilalinux: so that's one port. |
| 22:43.59 | *** join/#asterisk enzo (n=enzo@82.225.193.31) |
| 22:44.02 | enzo | hi |
| 22:44.14 | asdx | is there wifi voip phones? |
| 22:44.15 | saftsack | JT, ok would i get more voice quality and configuration comfort / dollar with a gateway for connecting my * to the PSTN via PRI or with an E1 interface card? |
| 22:44.15 | puzzled | evening |
| 22:44.34 | lilalinux | JT: yeah, but the card needs one for the line, and an internal |
| 22:44.37 | enzo | i've launched asterisk in root, now i have problem when i launch asterisk with a normal user, i get db.c:47 dbinit: Unable to open Asterisk database |
| 22:44.43 | puzzled | asdx: did you try to google? |
| 22:44.46 | lilalinux | JT: I guess a standard fritz won't do it |
| 22:44.50 | enzo | what is the file asterisk database ? |
| 22:44.54 | JT | lilalinux: sorry, be clear, how many ISDN2 S0 buses do you require? |
| 22:45.32 | saftsack | lilalinux, did you have a look at the patton gateways? |
| 22:45.35 | JT | saftsack: you've described using the same thing, E1 is PRI too |
| 22:45.41 | lilalinux | JT: 1 internal and 1 external |
| 22:45.51 | JT | explain this internal and external |
| 22:45.54 | lilalinux | saftsack: not yet |
| 22:46.44 | saftsack | JT, yes but i compare these things: * ---SIP----> Gateway -----E1 ----> provider AND * + E1 interface card ----> provider |
| 22:46.55 | enzo | ah yes astbin |
| 22:47.01 | lilalinux | JT: I want to connect the card to isdn (external) so I can receive calls/faxes and want to connect my gigaset base station (internal) for our dect phones |
| 22:47.22 | saftsack | lilalinux, they offer a 2 port (one te and one nt) gateway for 300eu. |
| 22:47.30 | lilalinux | saftsack: thx |
| 22:47.30 | JT | lilalinux: so you want 2 ports |
| 22:47.37 | lilalinux | JT: k |
| 22:47.45 | JT | switchable TE and NT |
| 22:47.51 | saftsack | its more expensive than a fritzbox but a fritzbox doesnt has sip gateway support |
| 22:47.55 | lilalinux | no NT needed |
| 22:48.01 | saftsack | there are hacks for the fritzbox but they dont work properly atm |
| 22:48.10 | JT | lilalinux: NT needed, you are connecting to a dect base station are you not |
| 22:48.12 | saftsack | lilalinux, do you need 2 voice channels or 2 isdn ports? |
| 22:48.22 | JT | the dect base station would usually operate in TE mode |
| 22:48.24 | saftsack | JT, do you have a guess? |
| 22:48.28 | JT | requiring an NT on the other side |
| 22:48.32 | lilalinux | JT: k |
| 22:48.54 | JT | by all means check the specs of you dect station |
| 22:48.58 | lilalinux | saftsack: JT says 2 ports |
| 22:49.13 | JT | but if it normally plugs into the provider's NT1, it's a safe bet it need to connect to a device with NT mode |
| 22:49.18 | saftsack | so 4 voice channels? |
| 22:49.36 | JT | saftsack: yes that's 4 voice channels, it really is irrelevant to this question though |
| 22:49.48 | saftsack | didnt get the question |
| 22:49.52 | lilalinux | :) |
| 22:49.55 | saftsack | ^^ |
| 22:50.00 | SomethingISODD | [TK]D-Fender http://pastebin.ca/247024 |
| 22:50.12 | JT | lilalinux: i'd normally recommend junghanns.net but don't know if they make anything under 4 ports |
| 22:50.27 | lilalinux | thx JT, saftsack |
| 22:50.44 | JT | saftsack: ahh, mentioning the PRI at the SIP provider's end is what got me confused |
| 22:50.54 | JT | again, how they connect to the pstn is usually irrelevant :) |
| 22:51.07 | JT | so it's SIP provider vs pri |
| 22:51.09 | saftsack | but what is the cleverest way? |
| 22:51.15 | JT | clever |
| 22:51.25 | Dovid | does ANI come thru on SIP ? |
| 22:51.25 | JT | depends on the cost of calls using each method in your country |
| 22:51.30 | JT | your reliability requirements |
| 22:51.39 | JT | Dovid: CLI does |
| 22:51.50 | Dovid | do i need to hack asterisk to get it ? |
| 22:51.51 | *** join/#asterisk MoutaPT (n=root@a213-22-40-63.cpe.netcabo.pt) |
| 22:51.56 | JT | not usually |
| 22:52.00 | Dovid | i wana route calls based on ANI |
| 22:52.06 | JT | unless it's sent with a non standard method |
| 22:52.10 | JT | do you mean CLI or ANI? |
| 22:52.18 | lilalinux | JT: can I alternatively use 2 standard ISDN cards? |
| 22:52.30 | JT | lilalinux: i know the cheapies can't do NT mode |
| 22:52.42 | Dovid | so it will only come up in CLI ? |
| 22:53.04 | Dovid | i want to use it int he dial plan :( |
| 22:53.07 | SomethingISODD | could someone take a look at http://pastebin.ca/247024 plzz |
| 22:53.15 | JT | i have no idea if they have ani, i thought that was more a telco thing |
| 22:53.28 | JT | Dovid: you can use incoming CLI in the dialplan |
| 22:54.47 | [TK]D-Fender | SomethingISODD : I told you to change it back to gotoiftim, and not use that function since you're calling it wrong anyways. |
| 22:54.59 | [TK]D-Fender | SomethingISODD : And you have to priority 8's in that context. BAD. |
| 22:55.02 | SomethingISODD | i showed both ways |
| 22:56.07 | SomethingISODD | let me check |
| 22:56.34 | saftsack | gn8 |
| 22:56.39 | SomethingISODD | [TK]D-Fender would this not be the correct function exten => s,8,GotoIfTime(09:00-16:30|mon-fri|*|*?office-open,s,9) |
| 22:56.44 | [TK]D-Fender | SomethingISODD : please clean it up and why do you start numbering your "s" exten all over the place between various contexts? Big mess in there. |
| 22:57.15 | SomethingISODD | what do you mean? |
| 22:58.26 | Corydon76-home | Generally the right way to number contexts is the autonumbering of priority 'n' |
| 22:58.46 | SomethingISODD | ok so what i would just need a number to start it off? |
| 22:58.53 | SomethingISODD | and then set n for the rest |
| 22:58.58 | Corydon76-home | Correct |
| 22:59.04 | [TK]D-Fender | SomethingISODD : You shouldn't be numbering "s" in each of those contexts starting at some freakish number like 7, 8, and 9 like we see in there. |
| 22:59.06 | SomethingISODD | ok thanks let me change all that |
| 22:59.09 | Corydon76-home | and generally you always start with priority 1 |
| 22:59.34 | [TK]D-Fender | And I don't advocate use of "n". Hard-coding is still fine, ust START with 1 for goodness sakes |
| 22:59.45 | SomethingISODD | :-) |
| 22:59.47 | SomethingISODD | ok |
| 22:59.57 | Dovid | i agree with TK |
| 23:00.03 | Corydon76-home | The only reason for you to start with some other number is to create a subroutine common only to that single extension or pattern |
| 23:00.09 | Dovid | if u wana jump around, it gets messy with n |
| 23:00.49 | Corydon76-home | Dovid: auto-jumping is deprecated and will be removed in a future version |
| 23:00.52 | Corydon76-home | Only explicit jumps will remain |
| 23:01.07 | Dovid | TK: have u ever seen issue with astdb where u use a specific variable over and over and eventually u cant set variables any more ? |
| 23:01.18 | Dovid | Corydon: manual jumping |
| 23:01.23 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 23:01.25 | Dovid | and y r they removing it ? |
| 23:01.26 | asdx | for example, if someone calls to my house from the regular telephone... and the call goes to my asterisk pbx, can i answer the call if i am in some other part of the world? |
| 23:01.38 | Corydon76-home | Dovid: because it's prone to error |
| 23:01.42 | Dovid | asdx: PLEASE READ THE BOOK |
| 23:01.46 | Dovid | ~book |
| 23:01.47 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:01.51 | Dovid | corydonL how so ? |
| 23:01.55 | asdx | Dovid: auch, ok... |
| 23:02.06 | Dovid | sorry for shouting, its gettin on me nerves |
| 23:02.13 | *** join/#asterisk ThaZZa (n=me@229.9.233.220.exetel.com.au) |
| 23:02.21 | Corydon76-home | Dovid: people create extensions that are more than 100 long and jumping could happen when they do not expect |
| 23:03.13 | asdx | Dovid: yeah, i agree ;p |
| 23:03.16 | Dovid | ok cause of n+101 |
| 23:03.17 | asdx | Dovid: i will read the book now |
| 23:03.21 | Dovid | :) |
| 23:03.27 | Dovid | TK: see my question ? |
| 23:03.30 | asdx | i'm just really excited ;p |
| 23:03.35 | Corydon76-home | Dovid: correct |
| 23:03.41 | Dovid | so start playin - get ur feet wet |
| 23:03.43 | [TK]D-Fender | Dovid : Variables != ASDTB |
| 23:03.50 | Dovid | so corydon what will replace it ? |
| 23:04.04 | Corydon76-home | Dovid: status variables that can be tested |
| 23:04.07 | Dovid | TK: I dont understand |
| 23:04.09 | Corydon76-home | e.g. DIALSTATUS |
| 23:04.25 | Dovid | so a gotoif |
| 23:04.31 | Corydon76-home | GotoIf($[${DIALSTATUS} = BUSY]?...) |
| 23:04.50 | Dovid | no |
| 23:05.29 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:05.29 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:05.38 | Corydon76-home | That's much easier to follow than trying to remember jumping +101 on a busy status |
| 23:05.56 | CunningPike | russellb: !!! |
| 23:06.06 | Dovid | i have a macro where the person says thier name and it calls the tech's cell and he selects what he wants to do with the call, based on what he pressed a value is stored in astdb and read in a few moments, problem is astb keeps returning an old value and not one that i set moments b4, started happening a few months into production |
| 23:06.10 | russellb | greetings CunningPike |
| 23:06.58 | Dovid | Corydon: n+101 was a lil easier - y not SomeCharacter+n+101 ? |
| 23:07.27 | Corydon76-home | Dovid: eh? |
| 23:07.29 | file | uh oh - a russellb |
| 23:07.32 | gerphimum | can someone explain the process that happens when a voip phone has its hold or transfer button pressed (eg what kinds kind of data is sent and in what protocol) |
| 23:07.59 | Dovid | tK: can u explain Variables != ASDTB ? |
| 23:08.00 | gerphimum | err -kinds* |
| 23:08.08 | JT | astdb is global |
| 23:08.12 | JT | and persistant |
| 23:08.14 | JT | iirc |
| 23:08.19 | JT | variables are per call |
| 23:08.38 | Supaplex | and globals are ... you guessed it, global. |
| 23:08.39 | hads | Except global variables :) |
| 23:08.53 | Supaplex | hads: gmta :) |
| 23:08.53 | Dovid | i know that |
| 23:09.04 | Dovid | JT: however they onli get one call at a time |
| 23:09.15 | Dovid | when i tried on a diff. tech's exten it worked like a charm |
| 23:09.42 | Dovid | seems to only acted up with the tech that gets most calls |
| 23:09.51 | Corydon76-home | Dovid: you might want to consider extending into the realm of an external database |
| 23:09.57 | [TK]D-Fender | Dovid : Stop trying to describe the problem to us and just PASTEBIN your defective code so we can pont out where you screwed up :) |
| 23:09.59 | Dovid | when i added a letter to the var. name it worked - so thats y i think its astdb |
| 23:10.06 | Dovid | thnx |
| 23:10.13 | Dovid | its real bad - it was my orig. code |
| 23:10.14 | Dovid | one sec |
| 23:10.22 | [TK]D-Fender | load chan_ascerbic.so |
| 23:10.40 | SomethingISODD | [TK]D-Fender better http://pastebin.ca/247043 ? |
| 23:11.08 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 23:11.30 | Dovid | client turned off ssh. gona have to have him enable it |
| 23:11.35 | [TK]D-Fender | SomethingISODD : 13-20 don't belong in [hours} from what I can see.... |
| 23:12.21 | [TK]D-Fender | SomethingISODD : Try using "\ |
| 23:12.40 | [TK]D-Fender | SomethingISODD : Try using "|" as your seperator after the ? in your gotoiftime. |
| 23:13.00 | Corydon76-home | SomethingISODD: instead of hardcoding hours,s,5, you might want to create a label there and use that |
| 23:13.06 | *** part/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 23:13.25 | [TK]D-Fender | SomethingISODD : And you'll awnt to change that 5 to 1. |
| 23:13.30 | Corydon76-home | i.e. exten => s,5(continue),GotoIfTime(...) |
| 23:13.40 | Corydon76-home | then: Goto(hours,s,continue) |
| 23:13.49 | SomethingISODD | ok |
| 23:15.21 | [TK]D-Fender | SomethingISODD : Forget lables for now.... just set it to 1 like its supposed to be for your sample. |
| 23:15.44 | SomethingISODD | ok, let me change all the other stuff as well |
| 23:16.31 | *** join/#asterisk rg1_ (n=rg1@www.airlinksystems.com) |
| 23:16.54 | SomethingISODD | [TK]D-Fender you mean like this correct exten => s,5,GotoIfTime(09:00-18:30\mon-fri\*\*?office-open,s,1) |
| 23:17.05 | MoutaPT | why does sometimes 2* servers with host static becomes unreachable and if we change it to host=dynamic and insert register string it starts working? Static IP on both servers. |
| 23:17.06 | rg1_ | There is a variable named ${UNIQUEID} that gets set automatically - anyone know WHEN that gets set? |
| 23:17.16 | russellb | rg1_: when the channel is created |
| 23:17.47 | [TK]D-Fender | SomethingISODD : No, I said AFTER the ? |
| 23:17.48 | JT | on an isdn connection, anyone know how to SET the outgoing MSN? |
| 23:18.22 | SomethingISODD | oh sorry |
| 23:19.17 | rg1_ | russellb - so does the ext have to "answer" to create the channel? |
| 23:19.42 | russellb | rg1_: the channel is created before that happens |
| 23:19.57 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 23:20.00 | rg1_ | ok, thx |
| 23:22.27 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-10.mtco.com) |
| 23:22.32 | *** join/#asterisk mosty (n=mostynm@60-241-198-194.static.tpgi.com.au) |
| 23:23.37 | SomethingISODD | [TK]D-Fender still the same issue |
| 23:24.47 | [TK]D-Fender | SomethingISODD : new complete pastebin please. |
| 23:25.10 | rg1_ | russellb - whats the difference between Set() and SetVar() |
| 23:26.39 | SomethingISODD | http://pastebin.ca/247055 |
| 23:28.37 | [TK]D-Fender | rg1_ : Setver was deprecated for Set. |
| 23:29.37 | [TK]D-Fender | SomethingISODD : Line 11 should read : exten => s,5,GotoIfTime(9:00-18:30|mon-fri|*|*?office-open|s|1) |
| 23:29.48 | [TK]D-Fender | SomethingISODD : Andwhat time does your system reda now? |
| 23:30.04 | SomethingISODD | Sun Nov 12 18:29:45 EST 2006 |
| 23:31.30 | rg1_ | Thx D-Fender |
| 23:31.45 | rg1_ | ok, when does REMOTESTATIONID get set? |
| 23:32.18 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 23:34.04 | [TK]D-Fender | rg_ : Maybe you should put a hold on the 20 questions and just try working with * a bit, and then start on questions directly related to a specific goal you are trying to acheive. |
| 23:34.28 | [TK]D-Fender | SomethingISODD : finished the cahnge, relaoded and retested? |
| 23:34.46 | SomethingISODD | yes |
| 23:35.34 | mosty | i'm having trouble with my internet connection, is there a good util i can use to measure the quality of the net connection (latency, jitter etc) to my voip provider? |
| 23:35.34 | rg1_ | D-Fender - I have been working * quite a lot. My problem is that all of a sudden certain variables are not behaving as before. For example, I used to be able to get the number that was called from ${DNID} - now it is coming up blank |
| 23:35.34 | MoutaPT | rg1_ : SetVar(name1=value1|name2=value2|..[|options]): This application has been |
| 23:35.34 | MoutaPT | deprecated in favor of using the Set application. |
| 23:35.50 | rg1_ | MoutaPT - thank you. |
| 23:36.20 | MoutaPT | on cli of asterisk you can make show application or show function followed by the name of app or func |
| 23:36.25 | MoutaPT | is very useful |
| 23:37.09 | MoutaPT | [TK]D-Fender are you experienced with zapata drivers? |
| 23:38.02 | [TK]D-Fender | rg1_ : most of those have been eaten up by the CALLERID function. |
| 23:38.16 | [TK]D-Fender | rg1_ : Let me guess... you used to work with mostly * 1.0.X, right? |
| 23:38.37 | [TK]D-Fender | MoutaPT : Ask a specific question, and I'll give you a specific answer :) |
| 23:40.56 | MoutaPT | [TK]D-Fender:Imagine you have a TDM10B (only one FXS) and TE110P and you want this FXS phone to be the ZAP/1 device on your server... how would you define this on zaptel.conf |
| 23:40.57 | MoutaPT | :) |
| 23:41.34 | MoutaPT | span=2,1,0,esf,b8zs would be the first line ? |
| 23:41.47 | MoutaPT | for the span definition of TE110P isn't it? |
| 23:42.16 | BosHaus | since the sipura 3000 is discontinued, should I grab a linsys PAP2? |
| 23:42.19 | [TK]D-Fender | MoutaPT : No, it'd be span 1 regardless of whether your TDM card loads first. the CHANNEL numbers may vary however. |
| 23:42.35 | SomethingISODD | [TK]D-Fender anymore ideas ?? |
| 23:42.52 | mosty | does anyone know of a good way to diagnose bandwidth issues? eg how can i test the packetloss and jitter between my asterisk box and a particular server on the net? |
| 23:43.09 | [TK]D-Fender | BosHaus : You can still get aSPA-3000's in a lot of places, and its been replaced byt he SPA-3102. the PAP is FXS only where the 3000 & 3102 are FXS/FXO. So its not a comperable replacement for all the functionality |
| 23:43.18 | MoutaPT | if i want FXS to be ZAP/1 |
| 23:43.19 | [TK]D-Fender | SomethingISODD : Do show how it looks now. |
| 23:43.20 | BosHaus | ok, thanks TK |
| 23:43.27 | MoutaPT | TDM must be loaded first |
| 23:43.29 | MoutaPT | no ? |
| 23:43.43 | MoutaPT | so the "span 1" is my four ports TDM |
| 23:43.48 | SomethingISODD | [TK]D-Fender u never said to do anymore changes since the last change... i pasted that update |
| 23:43.51 | [TK]D-Fender | MoutaPT : I would suggest ensuring that the TE100P be loaded first. |
| 23:43.51 | MoutaPT | so span=2 is for TE110P |
| 23:43.58 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 23:44.07 | [TK]D-Fender | MoutaPT : SPAN is only for digital cards, not, TDM analog |
| 23:44.48 | MoutaPT | yeah but asterisk creates a "span" for the TDM internally |
| 23:44.49 | MoutaPT | no ? |
| 23:44.55 | MoutaPT | i've been reading about it |
| 23:45.07 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-10.mtco.com) |
| 23:46.19 | MoutaPT | lease note that: |
| 23:46.19 | MoutaPT | <PROTECTED> |
| 23:46.19 | MoutaPT | <PROTECTED> |
| 23:46.19 | MoutaPT | <PROTECTED> |
| 23:46.19 | MoutaPT | FRom Beronet Zapata installation Guide |
| 23:49.23 | MoutaPT | <PROTECTED> |
| 23:49.58 | [TK]D-Fender | MoutaPT : do it like I told you |
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| 23:51.00 | MoutaPT | i know what u told me, my point was to accomplish the ZAP/1 for the FXS |
| 23:53.04 | SomethingISODD | http://pastebin.ca/247055 |
| 23:53.19 | SomethingISODD | [TK]D-Fender this is the last one i didi http://pastebin.ca/247055 |
| 23:53.20 | SomethingISODD | did |
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| 23:55.12 | [TK]D-Fender | SomethingISODD : Replace : # |
| 23:55.12 | [TK]D-Fender | exten => s,5,GotoIfTime(09:00-18:30|mon-fri|*|*?|office-open,s,1) |
| 23:55.31 | [TK]D-Fender | with : exten => s,5,GotoIfTime(09:00-18:30|mon-fri|*|*?office-open|s|1) |
| 23:55.52 | [TK]D-Fender | SomethingISODD : You didn't fix it following my prior message |
| 23:56.16 | rg1_ | Doesn't look like CALLERID(name) is working (at least not for me) |
| 23:56.47 | brif8 | Hi All, going to try 1.4 on a gentoo dev machine I see at astrecipes it talks about gcc+g++ and gnutls-devel for CentOS I don't see these under gentoo emerge is there something similar or is this a centos thing ? |
| 23:56.51 | rg1_ | nor does it look like CALLERID(dnid) is working |
| 23:57.23 | rg1_ | I AM getting CALLERID(ani) and CALLERID(num) |
| 23:57.57 | SomethingISODD | what do you mean replace #? |
| 23:58.14 | [TK]D-Fender | rg_ : please show use exact dialplan bits and CLI output to back it up please.... |
| 23:58.31 | [TK]D-Fender | SomethingISODD : Pay attention! |
| 23:58.31 | [TK]D-Fender | [18:55] <[TK]D-Fender> exten => s,5,GotoIfTime(09:00-18:30|mon-fri|*|*?|office-open,s,1) |
| 23:58.31 | [TK]D-Fender | [18:55] <[TK]D-Fender> with : exten => s,5,GotoIfTime(09:00-18:30|mon-fri|*|*?office-open|s|1) |
| 23:59.22 | SomethingISODD | ok doing it now |
| 23:59.42 | rg1_ | <PROTECTED> |
| 23:59.42 | rg1_ | <PROTECTED> |
| 23:59.42 | rg1_ | <PROTECTED> |
| 23:59.42 | rg1_ | <PROTECTED> |