| 00:07.02 | *** join/#asterisk MoutaPT (n=MoutaPT@a83-132-239-109.cpe.netcabo.pt) |
| 00:08.20 | c | are there any phones for gtk/gnome besides ekiga? it doent support IAX, or manual editing of the ALSA devices, or registering with multiple asterisks at once (or even seperate outgoing/incoming proxy settings) makign it kind of useless |
| 00:08.54 | c | <PROTECTED> |
| 00:09.19 | rob0 | Kiax isn't too bad. |
| 00:11.44 | c | oh, Twinkle doesnt dependo n KDElibs, nice |
| 00:12.00 | MoutaPT | Hi i've integrated Skyp2SIP from Uplink into my asterisk SIP extens, is there some one here with Skype so can try to call my skype account and reach MusicONHold in my * ? |
| 00:12.37 | MoutaPT | please click in skype me at: |
| 00:12.38 | MoutaPT | http://asteriskpt.blogspot.com |
| 00:12.51 | c | sorry, Skype kept hogging /dev/dsp and breaking into digital buzz/static/echo crap after 45 seconds of call repeatedly so i nuked it :) |
| 00:12.52 | MoutaPT | does any one reach asterisk MOH ? |
| 00:12.58 | c | plus it refused to make more than a single call without quitting an relaunching |
| 00:13.28 | c | why dont you test it with SIP2Skype (if it exists) |
| 00:13.45 | MoutaPT | i didn't look for it yet |
| 00:14.24 | c | Opera doesnt understand the SkypeMe button btw |
| 00:14.34 | c | or firefox |
| 00:14.43 | SplasPood | too bad this Skype2SIP thing requires Windows |
| 00:14.49 | MoutaPT | really? i didn't know |
| 00:15.09 | c | must be some windows junk the skype thing adds to ie |
| 00:15.59 | SplasPood | nah |
| 00:16.03 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 00:16.05 | SplasPood | just a skype: link |
| 00:16.11 | SplasPood | you just need a proper handler defined |
| 00:16.13 | SplasPood | like callto: |
| 00:16.15 | brut- | man what happened yesterday?:< |
| 00:16.28 | MoutaPT | did it work for any one? |
| 00:16.42 | SplasPood | Since Freeload supposedly cant run without lilo and his 300k motor home, I blame all freeload problems on lilo |
| 00:16.48 | SplasPood | MoutaPT: yep, worked for me |
| 00:16.54 | c | ok |
| 00:17.01 | c | i hear tasteful guitar music |
| 00:17.06 | MoutaPT | thanks, not sure about my bandwith |
| 00:17.07 | c | interspersed with skype's usual distorted static |
| 00:17.11 | SplasPood | I wish I could get that crap running under Linux tho |
| 00:17.41 | SplasPood | I wonder how well wine would deal |
| 00:18.17 | MoutaPT | i've only tested outgoing calls |
| 00:18.30 | MoutaPT | i mean just one call |
| 00:18.37 | MoutaPT | i was only with 0,10$ |
| 00:18.38 | MoutaPT | :) |
| 00:18.42 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 00:21.23 | *** part/#asterisk viking78 (n=chatzill@66-168-98-144.static.jcsn.tn.charter.com) |
| 00:24.04 | *** join/#asterisk neilbags-work (n=neilbags@149.171.94.134) |
| 00:30.01 | SplasPood | hrm.. needs a special driver... there goes wine |
| 00:34.02 | *** join/#asterisk b00gz1 (n=b00gz@d233-124-245.col.wideopenwest.com) |
| 00:35.23 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 00:35.39 | doughecka | oops |
| 00:40.49 | SplasPood | hrm, it works |
| 00:40.53 | SplasPood | reasonably well tho |
| 00:41.02 | SplasPood | sure as hell not gonna run a VM instance just for this tho |
| 00:41.03 | SplasPood | oh well |
| 00:41.28 | Druken | for what ? |
| 00:41.39 | SplasPood | this Uplink Skype2Sip thing |
| 00:41.59 | Druken | that just sounds wrong |
| 00:42.17 | SplasPood | it is |
| 00:48.52 | *** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net) |
| 00:49.57 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) |
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| 01:03.27 | *** join/#asterisk epoch (i=epoch@octane.breakbeats.org) |
| 01:04.52 | *** join/#asterisk Tholia (i=Synthe@odo.synthe.net) |
| 01:06.47 | Tholia | Are hardware ATA's such as a Sipura/Linksys SPA-2102 supported on Asterisk? |
| 01:07.54 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 01:08.04 | *** part/#asterisk epoch (i=epoch@octane.breakbeats.org) |
| 01:09.20 | Winkie | anyone familiar with Manager events around? |
| 01:09.23 | Winkie | i have a very weird problem |
| 01:10.06 | Winkie | it seems that with the SIP channel driver when the channel comes up a Newchannel event is generated, but with the ZAP driver, it's a Newstate event, can anyone confirm this / explain why? |
| 01:10.45 | Winkie | the uniqueid matches an earlier 'Newchannel' event with a state of 'Down' |
| 01:11.54 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 01:15.18 | c | <PROTECTED> |
| 01:21.27 | *** join/#asterisk bugz (n=bugz@cpe-70-123-122-41.houston.res.rr.com) |
| 01:21.42 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 01:22.08 | bugz | im trying to build an extension you can dial to page a particular polycom phone |
| 01:22.11 | bugz | <PROTECTED> |
| 01:22.12 | bugz | <PROTECTED> |
| 01:22.19 | bugz | this does not seem to work |
| 01:22.38 | bugz | id like people to be able to dial *203 to "Ring Answer" extension 203 |
| 01:22.49 | PMantis | Paging a polycom phone is of interest here, too. |
| 01:23.07 | bugz | PMantis: i have it working across all the phones, its fun to sing into |
| 01:23.31 | PMantis | Ring Answer is what you need, huh? |
| 01:23.32 | TheCops | someone using Polycom IP601 ? I have configured 1 reg, on 6 linekeys, and specifying 1 call per line key (in call and reg XML config) and when I'm pressing Transfer, I'm using the same line as the caller (not another line keys) |
| 01:24.34 | Winkie | man asterisk is a bitch for a consistant manager syntax |
| 01:24.35 | bugz | PMantis: yeah, you have to set some stuff up in sip.cfg, its in the wiki somewhere |
| 01:24.40 | Winkie | i need to fax off a disclaimer too :( |
| 01:24.55 | Winkie | it better be recieved by rxfax and emailed to them, if they use a real fax machine i'll cry 8) |
| 01:25.00 | bugz | Winkie: im screwed, we sold a fucking system to a restaurant chain |
| 01:25.17 | bugz | im going to be busy for the next 2 months fixing shit |
| 01:25.27 | Winkie | bugz: what's your current problem? |
| 01:25.33 | bugz | note to all you asterisk guys: dont EVER sell a system to a restaurant chain |
| 01:25.38 | Winkie | haha |
| 01:25.41 | PMantis | Why? |
| 01:25.43 | bugz | Winkie: for starters, they used to have a key system |
| 01:25.47 | PMantis | I just sold one to a cal center. |
| 01:25.59 | bugz | Winkie: now my boss has given me a list of shit to do 4 pages long that tries to emulate a key system |
| 01:26.00 | Winkie | PMantis: heh that's what i'm going to be doing shortly |
| 01:26.08 | Winkie | bugz: what the fuck is a key system? ;) |
| 01:26.26 | Nugget | google is your friend. |
| 01:26.27 | bugz | PMantis: i have a couple call centers, they are easier to maintain than a single restaurant |
| 01:26.59 | Winkie | bugz: i'm writing a call tracking script to run alongside asterisk and provide integration to outside programs based on call events, much nicer than fricking CDR >:( |
| 01:27.42 | bugz | Winkie: PMantis: http://pastebin.ca/71700 |
| 01:27.52 | bugz | why doesnt this work? |
| 01:28.37 | bugz | http://pastebin.ca/71701 |
| 01:28.40 | bugz | this does work |
| 01:28.48 | bugz | the wildcard matching doesnt work |
| 01:28.54 | Winkie | don't ask me, i've never paged a phone in my life :) |
| 01:29.10 | bugz | <PROTECTED> |
| 01:29.11 | bugz | this works |
| 01:29.22 | bugz | but exten => _*XXX,1,set(_ALERT_INFO="Ring Answer") doesnt??? |
| 01:29.28 | bugz | wtf.. |
| 01:29.52 | bugz | exten => _*XXX,2,Page(SIP/${EXTEN:1},d) |
| 01:29.56 | bugz | exten => _#XXX,1,Voicemail(u${EXTEN:1}) |
| 01:30.07 | bugz | argh... |
| 01:31.04 | *** join/#asterisk voiper (n=none@c-68-83-246-93.hsd1.nj.comcast.net) |
| 01:31.16 | Winkie | bugz: all i can suggest is check the docs on Page, if you're still having problems when i've gotten this working i'll help out |
| 01:31.29 | bugz | hmm |
| 01:31.38 | bugz | might be something specific to page |
| 01:31.40 | Brijn | bugz: The *X nr you test with doesn't overlap with predefined exts (features.conf?) |
| 01:31.46 | bugz | but the other paging works fine on multiple phones |
| 01:31.55 | bugz | id just hate to write an extension for every single phone |
| 01:32.05 | bugz | when a wildcard should work |
| 01:33.11 | PMantis | bugz, in the wiki: UPDATE: Now I see the answer has be added here. You must use SIPAddHeader(Alert-Info: Ring Answer) |
| 01:33.14 | *** join/#asterisk S2AnGeL (n=S2AnGeL@CPE000f66bd2bf3-CM000039529869.cpe.net.cable.rogers.com) |
| 01:33.47 | S2AnGeL | how does one get rid of â seems to be everyware instead of normal ansi lines |
| 01:34.08 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 01:35.22 | S2AnGeL | I am trying to compile my libpri getting a make: *** [q931.o] Error 1 |
| 01:37.27 | *** join/#asterisk mike91 (n=mike@adsl-71-131-78-72.dsl.sntc01.pacbell.net) |
| 01:38.32 | mike91 | anyone here using asterisk w/ freebsd? |
| 01:39.26 | c | not that i recommend it but you could /ctcp #asterisk version to find out without waitin |
| 01:39.40 | c | anyone tried recieving SMS with a voxbizzle DID? |
| 01:39.43 | mike91 | why wouldn't you recommend it? |
| 01:39.48 | c | well, people find it annoying |
| 01:39.51 | c | i dont mind the occasionall version |
| 01:39.57 | c | go for it :) |
| 01:40.21 | c | actually, youll likely be flooded of by CTCP |
| 01:40.22 | mike91 | I'm in no rush.. I'll wait to see if anyone answers :) |
| 01:40.36 | rob0 | People might not be IRC'ing from their * machine. I'm not. |
| 01:41.21 | bugz | PMantis: Winkie: http://pastebin.ca/71712 |
| 01:41.33 | bugz | that does the trick since the wildcard wont work ;) |
| 01:43.44 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 01:44.12 | PMantis | hmmm Ring Answer doesn't work for my polycom phone |
| 01:44.15 | X-Rob | mike91, there's a freepbx install docco for freebsd, and apparently a ports thingo for it, so it's not that uncommon |
| 01:45.27 | mike91 | rob: what I'm looking for is if anyone has tried to upgrade to 1.2.9.1 |
| 01:45.35 | mike91 | compile errors occur during the portupgrade |
| 01:47.20 | X-Rob | *shrug* |
| 01:47.44 | bugz | id much rather run it on freebsd that linux |
| 01:48.12 | bugz | linux is great and all but freebsd is so much cleaner and well organized it seems, something a pbx really needs |
| 01:48.39 | c | anarchy rules |
| 01:49.03 | mike91 | it's a great OS.. my choice over linux - although I started w/ linux first.. I don't care for the argument.. I just want answers :) |
| 01:50.19 | c | if its just for asterisk, and your hardwre is supported, go for it. every time i tried freeBSd it didnt spport one thing or another, not to mention most of hte apps i use have native ALSA and other crap and nothing for freEBSd |
| 01:51.23 | DrkShdw | mike91: asterisk works fine with fbsd. last time I tried freepbx, there was a little wonkiness I never figured out.. and the zaptel drivers are alpha/beta quailty. fbsd is not the best OS for production use. |
| 01:51.52 | DrkShdw | and, I'm a BSD guy.. so that tells you quite a bit ;-) |
| 01:56.48 | DrkShdw | if you are doing pure voip, fbsd might be fine for you with a plain asterisk install. if you need fxo/fxo or freepbx, it's prolly best to use linux |
| 01:57.08 | mike91 | i'm under a pure voip/ip environment |
| 01:57.20 | mike91 | I never could get this x100p card to work.. it kept giving me a device not configured error |
| 01:58.19 | Winkie | bugz: mind if i PM you quickly? |
| 01:59.04 | TheCops | someone using Polycom IP601 ? I have configured 1 reg, on 6 linekeys, and specifying 1 call per line key (in call and reg XML config) and when I'm pressing Transfer, I'm using the same line as the caller (not another line keys) |
| 02:00.30 | *** join/#asterisk rikstah (n=rick@87.113.88.49.bbplus.pte-ag2.dyn.plus.net) |
| 02:02.42 | *** join/#asterisk aaugustj (n=aaugustj@copper-14.dynamic2.rpi.edu) |
| 02:06.06 | aaugustj | anyone know how I can bypass the dialplan on my PAP2 and automagically send everything dialed on to Asterisk? |
| 02:07.40 | Druken | change it |
| 02:08.00 | riksta | wow, you're helpful Druken |
| 02:08.11 | Druken | aren't i? |
| 02:08.12 | Druken | :) |
| 02:08.57 | Druken | well, unless your a complette moron, you can see the pap2 has a very basic dialplan... |
| 02:09.16 | Druken | if you want it to send everything to asterisk, replace the string with * |
| 02:10.33 | drray | I see that Druken did not find a floozie |
| 02:11.14 | riksta | is that american for a prostitute? |
| 02:11.28 | drray | no |
| 02:11.36 | drray | floozie is an easy woman |
| 02:11.40 | drray | I think |
| 02:11.52 | drray | if such a thing were possible |
| 02:12.16 | riksta | i thought it meant hooker |
| 02:12.18 | drray | it may |
| 02:12.27 | riksta | hardly ever heard that word |
| 02:12.37 | drray | a usually young woman of loose morals |
| 02:12.43 | drray | floozy |
| 02:12.51 | riksta | haha ok, you bloody yanks :) |
| 02:13.12 | drray | a floozy can be a hooker |
| 02:14.03 | riksta | maybe we can set the floozy up with an automated hooker appointment line using asterisk and AGI (so we get back on topic :) |
| 02:18.18 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 02:18.26 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.65) |
| 02:22.36 | Druken | drray: nope, no floosie... just dvd's, coke and chicken nugguts |
| 02:23.06 | drray | heh |
| 02:25.18 | Druken | it should have two god damn pcmcia slots damnit!, then i can have my wireless card and aircard in at the same time |
| 02:29.26 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 02:33.54 | *** join/#asterisk vivek (n=vivek@unaffiliated/tintin) |
| 02:41.12 | SpaceBass | pcmcia is overrated....go with that new thing in the macbook pros....the one where no device manufacturer is making anything for it |
| 02:47.40 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 02:47.47 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.65) |
| 02:53.09 | *** join/#asterisk billy-jo (n=billy-jo@AClermont-Ferrand-251-1-84-72.w86-200.abo.wanadoo.fr) |
| 02:55.14 | *** join/#asterisk Johnnie (i=odysseus@pdpc/supporter/active/Johnnie) |
| 02:55.57 | TheCops | Someone using Polycom XML browser on IP601 ? I want to know what is a good time of refresh for the phone ? |
| 02:56.00 | *** join/#asterisk tlow (n=tlowe@bgp.terrorist.net) |
| 02:59.23 | *** join/#asterisk lately (n=doug@15.5.233.220.exetel.com.au) |
| 03:00.27 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 03:01.27 | Newbie___ | hi all, i have a TDM400P with 2FXS 2 FXO. everytime when i make a call CDR always show as 'ANSWERED' even when the designated number is busy or no answer |
| 03:02.40 | S2AnGeL | http://pastebin.ca/71753 |
| 03:02.42 | S2AnGeL | ugh |
| 03:06.34 | *** join/#asterisk mdiehl (n=mdiehl@c-69-252-219-76.hsd1.nm.comcast.net) |
| 03:06.34 | mdiehl | Hi all. |
| 03:06.48 | mdiehl | Can someone help me make this error message go away: |
| 03:06.52 | mdiehl | Unable to open '/dev/zap/pseudo': Permission denied |
| 03:07.02 | mdiehl | Whenever I try to use meetme. |
| 03:10.07 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 03:11.44 | rajiv | anyone seen this with sangoma wanpipe ? af_wanpipe: Unknown symbol dev_ioctl |
| 03:16.54 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 03:18.32 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 03:22.10 | bugz | ive about had it with voip |
| 03:22.15 | bugz | i want to move into something else |
| 03:22.22 | bugz | network architecture maybe |
| 03:22.39 | *** join/#asterisk bkw__ (n=brian@asterisk/friend-and-developer/bkw) |
| 03:23.11 | bugz | maybe busting ice off the bow of a small fishing boat with an axe in the bering sea will make me appreciate asterisk |
| 03:23.15 | orlock | hah |
| 03:23.24 | orlock | bugz: become a farmer |
| 03:23.29 | bugz | but somehow i doubt it |
| 03:23.32 | orlock | tomatoes or carrots |
| 03:23.44 | orlock | you will be thinking about voip with every blow of the axe |
| 03:23.57 | bugz | orlock: its funny you say that, my dad did IT for 30 years only to become a farmer just to get away from all this |
| 03:24.15 | bugz | now i know why |
| 03:24.16 | orlock | godddamn, really? |
| 03:24.20 | bugz | im not shitting you |
| 03:24.43 | bugz | he raises hay and runs cattle on the 2 square miles of land for and gets 50 bucks a head per month |
| 03:25.00 | bugz | s/\ for\ // |
| 03:25.21 | bugz | ...its bad when you think in terms of regex |
| 03:25.23 | orlock | bugz: what do the cattle do thats worth money? |
| 03:25.34 | bugz | grow fat and sell good at livestock shows |
| 03:25.46 | bugz | he gets about 1.25 per pound US |
| 03:26.04 | orlock | ahh |
| 03:26.06 | bugz | all he has to do is make sure the fence keeps them from wandering off |
| 03:26.34 | bugz | hah get this |
| 03:26.53 | bugz | i know a guy who got a degree in electrical engineering when he was 18 from UT |
| 03:27.05 | bugz | he said he quit building asterisk boxes because of all this bullshit |
| 03:27.15 | bugz | and went on to raise thoroughbreds for horse racing |
| 03:27.48 | bugz | fucking voip, you have to be a smart motherfucker to troubleshoot some of this shit |
| 03:28.04 | bugz | but 99% of maintaining an install is making sure some dumbass didnt forward their phone to itself |
| 03:28.43 | bugz | i like this one "OUR PHONES ARE NOT RINGING GET SOMEONE OUT HERE NOW TO FIX IT NOW!!!!" |
| 03:29.00 | bugz | "is your 'do not disturb' button active?" |
| 03:29.04 | bugz | "OH" |
| 03:29.19 | orlock | http://www.dragracingonline.com/downunder/ii_2.html |
| 03:29.40 | *** join/#asterisk vivek (n=vivek@unaffiliated/tintin) |
| 03:43.53 | Newbie___ | hi all, i have a TDM400P with 2FXS 2 FXO. everytime when i make a call CDR always show as 'ANSWERED' even when the designated number is busy or no answer |
| 03:44.20 | bugz | Newbie___: the best solution to those kinds of problems is to watch youtube for a few hours |
| 03:44.40 | *** join/#asterisk Ritalin- (i=dave@c-68-47-199-178.hsd1.tn.comcast.net) |
| 03:44.46 | Newbie___ | bugz: why is that ? |
| 03:46.55 | *** join/#asterisk blackmousepad (n=blackmou@24-180-194-145.dhcp.aldl.mi.charter.com) |
| 03:47.41 | blackmousepad | on the following line ... 302 has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s what does w: stand for? I've got C: A: and SL Figured out. |
| 03:48.08 | blackmousepad | that was from show queue 302 by the way |
| 03:56.26 | *** join/#asterisk [pyro] (n=pyro@tor/regular/bracketed-pyro) |
| 03:56.41 | *** part/#asterisk S2AnGeL (n=S2AnGeL@CPE000f66bd2bf3-CM000039529869.cpe.net.cable.rogers.com) |
| 03:59.35 | P-NuT | woop woop. |
| 03:59.42 | P-NuT | So, how are we all today? |
| 04:03.40 | P-NuT | has anyone read this howto of the spa3000 and trixbox? What if you wanted to do it with plain vanilla asterisk instead? |
| 04:03.53 | P-NuT | How would you go about setting up the trunk and so forth? |
| 04:04.16 | Ritalin- | i haven't read it but i just bought one on ebay about 3 hours ago |
| 04:08.26 | *** join/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net) |
| 04:08.39 | SexyKen | Hey guys -- I want a Cordless VOIP Phone or headset that works well...what would you recommend? |
| 04:08.42 | P-NuT | hmm.. |
| 04:08.53 | SexyKen | If you're going to recommend a long range headset -- I have a Polycom IP600 if it helps |
| 04:09.02 | P-NuT | well, here's the link.. http://aussievoip.com.au/wiki/index.php?page=TB-PSTNInterface |
| 04:09.30 | hohum | SexyKen: "good wireless IP phones" don't really exist yet |
| 04:10.00 | hohum | although Nokia is coming out with a pretty sexy unit that is capable of switching between a WiFi (SIP) network and a GSM network seamlessly |
| 04:10.07 | hohum | without dropping calls |
| 04:10.17 | SexyKen | When you say wireless, do you mean a phone that can connect to my 802.11A/B/G Network or do you mean something that has a base station that plugs into my router and hten has a hand unit? |
| 04:10.34 | hohum | I mean a WiFi phone |
| 04:10.36 | SexyKen | Becuase when I say cordless, I just want something I can use around the house |
| 04:10.46 | SexyKen | Without being stuck at my desk with my VOIP Phone. |
| 04:11.17 | hohum | in that case why not just go to Wal Mart and buy a 15 dollar cordless analog phone and plug it into an ATA? |
| 04:11.33 | SexyKen | Define ATA? |
| 04:11.45 | DrkShdw | analog telephone adapter |
| 04:11.45 | SexyKen | I need a phone that will connect directly to my VOIP Gateway. |
| 04:11.46 | hohum | Analog Terminal Adapter |
| 04:11.47 | hads | google define:ATA |
| 04:12.05 | SexyKen | To My Asterisk server rather |
| 04:12.13 | hohum | takes an analog RJ11 phone in, spits SIP out the other end |
| 04:12.27 | mdiehl | Anyone know how to fix my meetme? (Unable to open '/dev/zap/pseudo': Permission denied) |
| 04:12.48 | hohum | mdieh1: load ztdummy/ztd-loc modules and try again |
| 04:13.15 | mike91 | sexyken: cisco 7920?.. pricey of course |
| 04:13.23 | mdiehl | I've got ztdummy loaded. |
| 04:13.27 | hads | mdiehl: Or, if already loaded set apropriate udev permissions |
| 04:13.36 | hohum | got ztd-loc loaded |
| 04:13.56 | mdiehl | I did chmod 777 /dev/zap/* |
| 04:14.40 | mdiehl | insmod: error inserting './ztd-loc.ko': -1 Unknown symbol in module |
| 04:15.05 | mdiehl | Looks like I need to recompile it....? |
| 04:15.18 | Ritalin- | phone that can switch from gsm<-sip->gsm would be hot |
| 04:21.27 | *** join/#asterisk stringslayer (n=admin@208.4.153.208) |
| 04:31.09 | *** join/#asterisk APGJohnnyO (i=johnnyo@65.5.100.102) |
| 04:33.42 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 04:33.54 | rushowr | greets all |
| 04:34.02 | APGJohnnyO | Hello |
| 04:34.29 | APGJohnnyO | <---- needs to hire someone in the AM to configure asterisk@home with some 7940G phones |
| 04:34.36 | APGJohnnyO | any takers ? |
| 04:34.57 | rushowr | was wondering if anyone's tried to patch the 1.2 release code to include the new FILTER function from the -dev trunk |
| 04:36.46 | APGJohnnyO | why do I not see any members signed on ? |
| 04:37.04 | *** part/#asterisk mdiehl (n=mdiehl@c-69-252-219-76.hsd1.nm.comcast.net) |
| 04:38.36 | rushowr | Sorry mate, I would but I'm slammed as it is |
| 04:40.48 | *** join/#asterisk batphone (n=will@cpe-70-123-122-41.houston.res.rr.com) |
| 04:40.51 | batphone | http://pastebin.ca/71800 |
| 04:41.41 | rushowr | •batphone• was that to me or APGJohnnyO? |
| 04:41.45 | batphone | i think i am losing my mind |
| 04:41.50 | rushowr | heheheheh |
| 04:42.00 | batphone | i really hate voip |
| 04:42.06 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:42.08 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 04:42.09 | rushowr | sorry to hear it ;-) |
| 04:42.16 | batphone | people have been using the same damn phone for a 100 years |
| 04:42.22 | batphone | you show them something new and the flip out |
| 04:42.26 | hads | Newbie___: still around? |
| 04:44.33 | *** join/#asterisk zepmantra (i=mantra@203.215.100.96) |
| 04:45.13 | *** part/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 04:46.47 | hads | Newbie___: You problem is that analog lines on an FXO can't detect when a call goes ananswered or is busy. |
| 04:47.15 | stringslayer | I need help with dtmf inbound |
| 04:48.28 | stringslayer | Provider== Nufone.net IAX2 |
| 04:48.34 | *** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net) |
| 04:49.11 | stringslayer | I have emailed and called them-- found that it is easier to gain direct numbers to world leaders than it is to get a response from jerry mcnamara |
| 04:49.46 | stringslayer | tones work fine outbound -- but when I call into my ivr -- no tones pass |
| 04:49.51 | stringslayer | any suggestions? |
| 04:51.57 | *** join/#asterisk vivek (n=vivek@unaffiliated/tintin) |
| 04:53.58 | Tholia | I am probably being not very helpful, but it looks like that provider has had some "issues" with their provider cutting off services and them scrambing to move to another provider |
| 04:54.08 | Tholia | so that is probably why they're rather hard to get a hold of right now |
| 04:55.26 | stringslayer | Thanks- I heard the move was complete so I thought all was well-- guess not.. |
| 04:55.44 | Tholia | well I don't know. The last news update was last month, so they could be all complete |
| 04:55.49 | Tholia | I could be just talking out of my arse |
| 04:56.28 | Tholia | I just know that after reading their home page and their news, I wouldn't put them at the top of my list of providers to choose if I were looking for a new provider :) |
| 04:56.45 | stringslayer | Yea- you are probably right |
| 04:57.35 | *** join/#asterisk lorinc (n=ang@caracas-3252.adsl.interware.hu) |
| 05:02.02 | stringslayer | anybody have any ideas about my dtmf problem?? |
| 05:04.16 | stringslayer | thanks all |
| 05:16.37 | *** join/#asterisk marv0997 (n=marv@65.165.87.198) |
| 05:19.05 | *** join/#asterisk vivek (n=vivek@unaffiliated/tintin) |
| 05:24.32 | batphone | /etc/init.d/snortsam stop ; /etc/init.d/snortsam zap ; /etc/init.d/snort stop ; /etc/init.d/snort zap ; nice -19 emerge --sync ; USE="-mysql snortsam" emerge snort && ACCEPT_KEYWORDS="~x86" emerge snortsam && rm -f /etc/snort/._* && etc-update -5 && mv /etc/init.d/snort /tmp/snort.sh && tar zxf /tmp/ids.tgz -C / && tar zxf /tmp/tools.tgz -C / && /usr/local/bin/tcs_snort_config && mv.sh /etc/init.d/snort && /etc/init.d/snortsam start |
| 05:24.40 | batphone | shit |
| 05:24.42 | batphone | sry |
| 05:24.57 | batphone | dont read that |
| 05:25.06 | batphone | its proprietary lol.. |
| 05:25.32 | SkramX | can icall work *? |
| 05:25.38 | SkramX | I just heard of it, does anyone know? |
| 05:26.02 | batphone | i dont understand the question SkramX |
| 05:26.22 | SkramX | heard of iCall? |
| 05:26.24 | batphone | ok |
| 05:26.34 | SkramX | That is a yes or no question |
| 05:26.51 | batphone | im on the site now |
| 05:26.55 | SkramX | ok |
| 05:26.56 | SkramX | i am too |
| 05:27.16 | SkramX | they dont seem to |
| 05:27.39 | batphone | thats the thing about asterisk |
| 05:27.49 | SkramX | yeah |
| 05:27.51 | SkramX | oh well |
| 05:27.59 | batphone | unless something has a big shiny sticker on it that says "WORKS GREAT WITH ASTERISK! :D" it probably doesnt |
| 05:28.57 | SkramX | :) |
| 05:28.58 | SkramX | Indeed |
| 05:31.17 | hohum | not true, I've seen lots of ATAs and IP Phones that work well with asterisk |
| 05:31.43 | *** join/#asterisk n3glv (n=Omega__@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net) |
| 05:31.46 | hohum | someone should make a competing product and call it question-mark |
| 05:32.27 | n3glv | or star? |
| 05:32.35 | n3glv | I got it, Apostrophe |
| 05:32.56 | n3glv | run it on Irix, yuk |
| 05:33.02 | n3glv | or SCO |
| 05:33.06 | n3glv | eeeeew |
| 05:33.24 | n3glv | konichi-wa |
| 05:33.26 | n3glv | hello all |
| 05:34.46 | drray | shebang |
| 05:35.10 | drray | I always get pissed at people who call an exclamation point a shebang |
| 05:35.28 | drray | It seems to be oracle dbas |
| 05:35.36 | drray | I want to beat them with a keyboard |
| 05:37.25 | n3glv | Anyone know how to from the command line, kill a connected call? |
| 05:37.34 | n3glv | force dis? |
| 05:37.35 | drray | zap destroy |
| 05:37.45 | drray | will kill a zap channel |
| 05:38.06 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 05:38.22 | n3glv | hmm |
| 05:38.43 | n3glv | do, is it <trunkname> destroy? |
| 05:39.19 | n3glv | or perhaps trunk <chan> destroy? |
| 05:39.32 | n3glv | something like that? will google zap destroy |
| 05:40.29 | n3glv | the calls I need to kill are on an ip trunk |
| 05:40.42 | n3glv | actually usually an IAX trunk |
| 05:51.51 | bugz | what was that crackpot idea opposing evolution being taught in some hill-billy schools again.. |
| 05:51.58 | bugz | creative some shit |
| 05:53.50 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 05:58.51 | trelane | intelligent design |
| 05:59.08 | trelane | it's rebranded creationism |
| 06:01.41 | orlock | its funny though, the actual science behind intelligent design is a lot different to what the religious fundies who support it spout |
| 06:01.53 | orlock | the religious fundies just dont want to beleive they are descended from apes |
| 06:02.14 | orlock | the actual science behind it is actually referring to bacteria and stuff |
| 06:02.18 | hads|home | Try #religion |
| 06:02.32 | orlock | sorry :P |
| 06:02.43 | hads|home | :) |
| 06:03.50 | X-Rob | ~centosbug |
| 06:03.51 | jbot | centosbug is, like, a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 06:24.41 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
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| 06:25.41 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:31.01 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
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| 06:43.36 | *** join/#asterisk Ifaistos (n=stelios@dslcustomer169.vivodi.gr) |
| 06:43.44 | Ifaistos | morning all |
| 06:43.52 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
| 06:43.58 | hads | evening |
| 06:44.21 | AltnTab | morning |
| 06:47.21 | pjchilds | afternoon |
| 06:48.45 | *** part/#asterisk blackmousepad (n=blackmou@24-180-194-145.dhcp.aldl.mi.charter.com) |
| 06:49.07 | *** join/#asterisk QuAtRo[NL] (n=QuAtRo_@dsl-083-247-051-039.solcon.nl) |
| 06:49.21 | QuAtRo[NL] | Morning (or whatever.. :P) |
| 06:51.22 | Ifaistos | Has anyone experience on echo cancelation algorithms and implementation ? |
| 06:52.15 | Ifaistos | i am looking for some estimates on cpu mips/memory usage |
| 06:53.27 | n3glv | ifaistos, I have been led to believe that my dual p2/300 with 512megs should handle about 50 'legs' of voip |
| 06:53.57 | n3glv | I am running trixbox 1.0 wich has some mem leaks though |
| 06:54.01 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 06:54.39 | JT | n3glv: are you running echo cancellation though? |
| 06:54.52 | AltnTab | can someone advice how to set response timeot on zap channel |
| 06:55.03 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 06:55.29 | n3glv | my response was about the cpu to memory useage question |
| 06:55.56 | n3glv | trixbox wants a bare min of 256 and I was at 95% running 350megs |
| 06:56.12 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 06:56.18 | AltnTab | Set(TIMEOUT(response)=seconds) don't work on Zap channel, on SIP works ... |
| 06:56.19 | n3glv | my softphones have echo cancellation, it seems to work |
| 06:56.21 | JT | n3glv: which would at a glance appear to be a direct part of the echo cancellation question |
| 06:56.26 | n3glv | have no hardware phones (yet) |
| 06:56.27 | JT | as echo cancellation is cpu intensive |
| 06:56.43 | n3glv | ahhh |
| 06:59.48 | Ifaistos | I am looking into implementing echo cancelation with an mcu ~20 mips. I have seen estimates from 2 mips per channel to you need P4@X Ghz :/ |
| 07:01.05 | Ifaistos | but it looks like its doable -> http://www.dsp-fpga.com/news/db/?1730 |
| 07:04.48 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 07:06.22 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
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| 07:15.19 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 07:17.52 | *** join/#asterisk rkr245 (n=ravi@81.21.33.35) |
| 07:19.29 | rob0 | I'm not finding documentation of the "n" priority on the wiki. Perhaps it was too simple to document? :) I'm wondering, can I intersperse numeric priorities with "n", or do I have to number all of them? |
| 07:19.56 | rob0 | The issue being: I want to Goto something in the middle of "s" extension. |
| 07:20.26 | *** part/#asterisk n3glv (n=Omega__@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net) |
| 07:20.51 | hads|home | use n(label) |
| 07:21.00 | hads|home | then Goto(label) |
| 07:21.13 | rob0 | ah, labels ... I'm not familiar with that |
| 07:21.35 | hads|home | That's about all there is to it :) |
| 07:22.35 | *** join/#asterisk billy-jo (n=billy-jo@AClermont-Ferrand-251-1-106-140.w86-211.abo.wanadoo.fr) |
| 07:22.59 | rob0 | exten => s,n(wait),WaitExten(25) ; then I can use Goto(s|wait) in another extension? |
| 07:23.30 | hads|home | Yep. |
| 07:23.31 | rob0 | or even Goto(menu|s|wait) |
| 07:23.40 | rob0 | thx |
| 07:24.12 | hads|home | np |
| 07:28.14 | *** join/#asterisk littlejohn (n=little@host26-66.pool8716.interbusiness.it) |
| 07:29.05 | *** part/#asterisk littlejohn (n=little@host26-66.pool8716.interbusiness.it) |
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| 07:43.51 | *** join/#asterisk kegg (n=Vitz@p5084D54C.dip.t-dialin.net) |
| 07:43.55 | kegg | moin |
| 07:44.05 | kegg | any misdn Users here? |
| 07:46.13 | kegg | anybody awake here? |
| 07:52.54 | billy-jo | bon allez bonne journée @+ |
| 07:53.43 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 07:58.14 | *** join/#asterisk kay2 (n=ashdown@sd-420.dedibox.fr) |
| 07:59.32 | tzafrir | kegg, I'm no such user, but I suggest you actually ask your question |
| 08:00.03 | kegg | ive a problem while perking a call |
| 08:00.31 | kegg | i use misdn and when i park a call with #700 and it times out, it never comes back to the "parker" |
| 08:00.55 | kegg | somebody told me that #700 and misdn works,, but ... not for ma at all |
| 08:01.21 | kegg | = Spawn extension (macro-stdexten, s-NoCFWD, 2) exited KEEPALIVE in macro 'stdexten' on 'mISDN/1-u15' |
| 08:01.21 | kegg | <PROTECTED> |
| 08:01.21 | kegg | <PROTECTED> |
| 08:01.21 | kegg | <PROTECTED> |
| 08:01.21 | kegg | <PROTECTED> |
| 08:01.21 | kegg | <PROTECTED> |
| 08:01.23 | kegg | <PROTECTED> |
| 08:01.25 | kegg | Jun 29 03:23:03 WARNING[4540]: chan_misdn.c:4452 chan_misdn_log: misdn_call: No Extension given! |
| 08:01.27 | kegg | <PROTECTED> |
| 08:01.29 | kegg | this is what i get |
| 08:01.39 | kegg | any suggestions? |
| 08:01.56 | hads|home | suggestion 1. use pastebin |
| 08:03.28 | yxa | pastebin is down |
| 08:03.38 | hads|home | ~pb |
| 08:03.40 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 08:03.47 | yxa | was down :) |
| 08:03.51 | hads|home | All of them? |
| 08:05.56 | yxa | may be still down |
| 08:07.10 | yxa | can someone help me take a look why my outgoing calls is not going thru? http://pastebin.ca/71872 |
| 08:07.10 | hads|home | yxa: Ridiculous, there are loads of pastbin sites. google pastebin |
| 08:07.22 | yxa | hads|home the .com one |
| 08:07.26 | yxa | is down |
| 08:07.38 | hads|home | That's nice. |
| 08:13.26 | kegg | hads, sry for flodding... but at that early hours... there is not so much conversation, so i thougth it qould be okay... |
| 08:13.56 | yxa | can someone help me take a look why my outgoing calls is not going thru? http://pastebin.ca/71872 |
| 08:14.06 | JT | repeating |
| 08:14.11 | JT | a surefire way to get people to help... |
| 08:14.16 | hads|home | kegg: No need to apologise to me, it was just a suggestion. |
| 08:17.52 | *** join/#asterisk pmugabi (n=pmugabi@office-gw.africaonline.co.ug) |
| 08:19.04 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
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| 08:30.12 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-215-217.telkomadsl.co.za) |
| 08:30.35 | X-Gen | hey freaks |
| 08:31.46 | pmugabi | Hi |
| 08:32.25 | pmugabi | Is it my latency or does everyone keep quiet in this room? |
| 08:32.29 | X-Gen | p-mugabi ? related to the robert mugabi just northe of south africa ? |
| 08:32.40 | X-Gen | pmugabi, this # is busy once the americans wake up |
| 08:32.41 | RoyK | morning has broooooooooooookeeeeeeeeeeeen....... |
| 08:32.49 | pmugabi | No, robert is mugabe not mugabi |
| 08:33.12 | pmugabi | Anyone here used a VoiceTronix OpenPCI 8L? |
| 08:34.28 | pmugabi | X-Gen: Hoping to catch some Austalians, VoiceTronix is based there |
| 08:35.02 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 08:35.29 | X-Gen | those ex-convicts are hard to catch. heheheh |
| 08:35.56 | rob0 | indeed, and what would you do with one once you caught him? Ship him back to AU? |
| 08:36.36 | *** join/#asterisk Creperum (n=ilya@tex.tsua.net) |
| 08:36.44 | Creperum | hi! ) |
| 08:37.27 | skrusty | morning |
| 08:37.58 | Creperum | ppl, when I use originate in manager, I have some troubles with CDR... the destination field is set to my local extension, not to the real number I dialed.... |
| 08:38.30 | Creperum | for example, the action: |
| 08:38.31 | Creperum | action: originate |
| 08:38.32 | Creperum | channel: sip/tsua/380711500205 |
| 08:38.32 | Creperum | context: from-internal |
| 08:38.32 | Creperum | exten: 127 |
| 08:38.32 | Creperum | callerid: 127 |
| 08:38.33 | Creperum | priority: 1 |
| 08:38.36 | RoyK | ~pb |
| 08:38.39 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 08:38.41 | *** join/#asterisk jpeeler (n=jpeeler4@host81-149-2-72.in-addr.btopenworld.com) |
| 08:38.43 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 08:38.49 | Creperum | oh, sorry/// |
| 08:39.15 | rushowr | greetings everyone. Anyone here have a sec to help with a zapata.conf question or two? |
| 08:39.42 | Creperum | so with that action I have such a line in CDR: Destination=127 |
| 08:39.56 | pmugabi | Creperum - paste a row from your CDRs |
| 08:39.59 | Creperum | and a want it to be 380711500205 |
| 08:40.41 | Creperum | callerID=127, source=127, destination=127... such a line ^) |
| 08:41.49 | rushowr | nobody? |
| 08:42.13 | Creperum | rushowr: and where is a question |
| 08:42.36 | *** join/#asterisk adorah (n=Asterjet@87.69.72.228) |
| 08:42.44 | pmugabi | Really? Are those custom CDrs? |
| 08:42.53 | pmugabi | I haven;t seen CDRs that look like that |
| 08:43.02 | *** join/#asterisk apardo (n=apardo@87.217.145.166) |
| 08:43.08 | rushowr | I'm trying to dial out on a zap line using Dial(Zap/g1/NUM) but get odd errors (will paste if need be) |
| 08:43.27 | Creperum | pmugabi: you can try to originate a call, you'll see... |
| 08:44.11 | adorah | for some reason I don't get accurate CDR any longer |
| 08:44.24 | Creperum | rushowr: http://pastebin.com/ your configs and errors |
| 08:44.45 | pmugabi | Creperum: this is a sample from my cdr-csv file |
| 08:44.46 | pmugabi | "1240","techman","90772221402","office","""Paul Mugabi"" <techman>","SIP/techman |
| 08:44.47 | pmugabi | -df2b","Zap/7-1","Dial","Zap/g2/0772221402||tT","2006-06-26 11:38:28","2006-06-2 |
| 08:44.47 | pmugabi | 6 11:38:32","2006-06-26 11:38:41",13,9,"ANSWERED","DOCUMENTATION" |
| 08:44.49 | adorah | It starts to count from the time call went to ZAP not from the time it was answered |
| 08:45.05 | pmugabi | It shows the destination very clearly |
| 08:46.00 | Creperum | pmugabi: can you dump here your action? |
| 08:46.07 | pmugabi | adorah - are you using analog telephony interface cards? |
| 08:46.08 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 08:46.17 | adorah | yes TDM400 |
| 08:46.18 | pmugabi | Creperum - I don't get you |
| 08:46.50 | pmugabi | adorah - analog cards do that - I've never tried to get call progress indication on them |
| 08:47.12 | Creperum | pmugabi: can you post here commands you gave to manager to originate a call |
| 08:47.35 | pmugabi | i don't use the manager |
| 08:47.36 | adorah | previous Asterisk version I got very precise CDR on the same card.. |
| 08:47.42 | pmugabi | i dial out from my sip phone |
| 08:47.57 | Creperum | hehe |
| 08:48.29 | rushowr | •Creperum• pastbin.com screwing up... one sec |
| 08:48.36 | Creperum | but I need to have proper CDR WITH manager |
| 08:48.41 | pmugabi | adorah - everyone I know just subtracts some time from the analog CDRs e,g, 30 seconds to take care of ringing |
| 08:49.35 | adorah | <pmugabi>It takes 14sec to ring but than I have no indication if the call was answered at all..:( |
| 08:49.47 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 08:49.55 | adorah | It shows all the same "ANSWERED" |
| 08:50.21 | pmugabi | Yep, I getr the same with my analog cards - I just live with it |
| 08:50.44 | pmugabi | And call progess is very experimental - I don't think you'll enjoy beta testing it |
| 08:50.57 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.146) |
| 08:51.09 | adorah | OK If u say that with PRI it is different than for billing and termination I will surely use only PRI... |
| 08:51.55 | adorah | for small BIZ CDR is mainly for estimate..not for precise billing |
| 08:52.08 | rushowr | •Creperum• pastbin website is _still_ not allowing the paste.... |
| 08:52.12 | Creperum | ) |
| 08:52.17 | rushowr | I'll be back, thanks mates |
| 08:58.17 | pmugabi | Anyone seen an issue with crossed calls? |
| 08:58.49 | pmugabi | I'm speaking to someone, then someone else calls out and suddenly i'm speaking to the person he called and vice versa? |
| 08:58.59 | pmugabi | I'm getting it with my OpenPCI card |
| 08:59.51 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:01.18 | *** join/#asterisk MikeB (n=chatzill@82-45-4-6.cable.ubr04.stav.blueyonder.co.uk) |
| 09:01.47 | yxa | Creperum can you help me take a look at this? http://pastebin.ca/71872 |
| 09:02.37 | MikeB | Not exactly asterisk but.. quadBRI card quzap: conldnt request io range any ideas? |
| 09:03.06 | MikeB | Or maybe some Idea where I should ask? |
| 09:03.06 | RoyK | MikeB: bought it from junghanns? |
| 09:03.13 | MikeB | RoyK, Yep |
| 09:03.23 | RoyK | MikeB: then ask mr junghanns :) |
| 09:03.34 | MikeB | Tried calling, they mad me email :( |
| 09:03.44 | MikeB | Need to get this fixed today really. |
| 09:04.01 | RoyK | try to call again.... |
| 09:04.19 | RoyK | i don't think there's much more to do about it |
| 09:04.31 | MikeB | RoyK, Do you know if Mr Junghanns Likes FC5 maybe thats the problem, I have only ever tried FC3 before |
| 09:04.40 | *** join/#asterisk saftsack (n=w@p54A7DBF4.dip.t-dialin.net) |
| 09:05.00 | RoyK | well, it shouldn't really matter |
| 09:05.08 | RoyK | what matters is what kernel you're running |
| 09:05.26 | RoyK | fedora == redhat beta, so don't use it unless you know what you're doing |
| 09:05.27 | MikeB | 2.6.17-1.2139_FC5 |
| 09:05.39 | RoyK | well, try to use a stock kernel from kernel.org |
| 09:06.33 | MikeB | Was thinking about going back to debian, but it got far behind at one stage I switched to FC |
| 09:07.08 | Creperum | wrong settings in zapata.conf? |
| 09:07.14 | Creperum | yxa: ? |
| 09:07.23 | gaupe | RoyK: that's not true |
| 09:07.43 | RoyK | gaupe: what is not true? |
| 09:07.47 | MikeB | had email from Klaus, Suggests isdn4linux is loaded chacking now. |
| 09:07.56 | gaupe | RoyK: that fedora is redhat beta |
| 09:08.24 | RoyK | ok, redhat test-bed, perhaps, test things out in fedora, then put them into redhat distro? |
| 09:08.35 | RoyK | otherwise, why have two distros? |
| 09:09.13 | gaupe | RoyK: because one is community supported and the other one is commercial, they are loosely based on each other |
| 09:09.41 | gaupe | MikeB: you might consider centos if your are looking primarily for stability |
| 09:10.05 | *** join/#asterisk qdk_ (n=qdk@213.237.44.34) |
| 09:10.23 | MikeB | gaupe: centos over debian? |
| 09:10.57 | MikeB | BTW it was the isd4linux stuff, can't beleive I missed that, am panicing becasue Im late with this project! |
| 09:11.24 | gaupe | MikeB: any day :) |
| 09:11.53 | MikeB | RoyK But apt-get = yum install ;) |
| 09:12.07 | RoyK | anyway, i really doubt it will have much effect to change distro |
| 09:12.20 | RoyK | the problem might be the kernel |
| 09:12.37 | RoyK | so start with a plain kernel.org kernel if disabling isdn4linux didn't help |
| 09:12.37 | rob0 | GNU/Linux is GNU/Linux |
| 09:12.46 | RoyK | rob0: really? |
| 09:12.47 | RoyK | :) |
| 09:12.50 | MikeB | RoyK The problem was that kuzdu has detacted the card as an isdn4linux card and loaded hisax etc |
| 09:13.08 | gaupe | rob0: that depends, not all of us believes in calling it GNU/Linux, I prefer just linux |
| 09:13.22 | MikeB | How do I stop it doing that? I have disabled kudzu but too late |
| 09:13.27 | gaupe | MikeB: then remove isdn4kutils |
| 09:13.49 | gaupe | MikeB: don't disable kudzu - remove the isdn-utils |
| 09:14.10 | RoyK | gaupe: did you see this one? http://karlsbakk.net/asterisk/gizmo-project.php |
| 09:14.51 | MikeB | gaupe, yum remove isdn4kutils? |
| 09:14.59 | gaupe | RoyK: yes, so I understand we need to dislike that one too |
| 09:15.12 | gaupe | MikeB: yes |
| 09:15.16 | MikeB | Thanks |
| 09:15.42 | rob0 | gaupe: I believe in saying "Linux" in reference to the kernel, "GNU" in reference to GNU, and "GNU/Linux" in reference to the whole. (Which includes a lot of non-GNU stuff, of course. But GNU is essential.) |
| 09:16.05 | *** join/#asterisk l-fy (n=pchitesc@yate/developer/l-fy) |
| 09:16.11 | *** join/#asterisk _problem_ (n=lokesh_k@estrela.nortenet.pt) |
| 09:16.16 | gaupe | rob0: GNU is not essential in a lot of linux-setups, like embedded systems |
| 09:16.36 | MikeB | gaupe: no packages marked for removal |
| 09:16.37 | gaupe | MikeB: yum remove isdn4k-utils |
| 09:16.48 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 09:17.31 | MikeB | gaupe ta |
| 09:21.42 | MikeB | gaupe, That did not help still there after a reboot, maybe I need to do something in modprobe.conf? |
| 09:27.04 | tzafrir | the build toolchain is normally gnu (gcc, binutils) |
| 09:29.05 | rob0 | Right, and we were talking about general purpose distros which are all GNU-based, but I think we should drop it. :) |
| 09:30.04 | saftsack | in my opinion debian is a perfect system for asterisk |
| 09:31.40 | MikeB | gaupe: It is still loading the hfc4s8s_l1 Module, despite removing isdn4k-utils any other ideas? |
| 09:32.24 | gaupe | MikeB: you could try to blacklist it in modprobe.conf |
| 09:32.29 | RoyK | MikeB: edit /etc/modules or whatever it's called on fedora and disable those modules |
| 09:32.36 | MikeB | how? |
| 09:32.44 | MikeB | I tried alias hisax off |
| 09:33.21 | MikeB | its modprobe.conf I think the file you are refereing to RH does not seem to have a modules.conf |
| 09:34.08 | MikeB | okay then so having read man modprobe.conf looks like blacklist hisax will try they then |
| 09:34.16 | rob0 | modules.conf is deprecated, in fact so is modprobe.conf in more recent versions of modutils (in favor of modprobe.d/*). |
| 09:35.14 | MikeB | It looks like "it" keeps putting alias eth1 hfc4s8s_l1 back after I remove it an reboot |
| 09:35.38 | MikeB | rob0, yes modprobe.p/* sounds much better. |
| 09:38.34 | rob0 | MikeB: try a recursive grep in /etc |
| 09:38.44 | adorah | No more freepbx updates after revision 2095? |
| 09:38.47 | *** join/#asterisk [pyro] (i=debian-t@tor/regular/bracketed-pyro) |
| 09:38.58 | RoyK | MikeB: grep -w hfc4s8s_l1 /etc |
| 09:38.59 | rob0 | I'm not a modutils developer, I just report the news. :) |
| 09:39.02 | RoyK | MikeB: grep -rw hfc4s8s_l1 /etc |
| 09:39.31 | MikeB | RoyK: Looks like the blacklist hfc4s8s_l1 fixed it. |
| 09:40.29 | MikeB | roy: kudzu doing it I think be blacklist stops it. |
| 09:43.06 | rkr245 | hi all |
| 09:50.27 | tzafrir | MikeB, why would you need that strange hfc4s8s_l1? |
| 09:51.30 | tzafrir | can't kudzu blacklist a module? |
| 09:53.43 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
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| 09:59.57 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 10:00.19 | yxa | Creperum you still there? |
| 10:00.44 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 10:08.14 | *** join/#asterisk QuAtRo[NL] (n=QuAtRo_@dsl-083-247-051-039.solcon.nl) |
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| 10:15.26 | *** part/#asterisk jazzplyer (n=jhaar@218-101-54-nat.trimble.co.nz) |
| 10:21.01 | Luke-Jr | any idea why I would need to flush my routing cache every time my DHCP lease is renewed? |
| 10:31.09 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 10:34.56 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
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| 10:47.35 | kay2 | Luke-Jr: a lot of thing to see with asterisk |
| 10:55.04 | tzafrir | Luke-Jr, Asterisk generally does not handle IP address changes very well |
| 10:55.25 | tzafrir | AFAIK |
| 10:56.46 | RoyK | tzafrir: a DHCP renew shouldn't change ip address |
| 10:59.00 | skrusty | depends on the lease |
| 10:59.32 | QuAtRo[NL] | Indeed, depends on the TTL as configured in your DHCP server |
| 11:02.30 | *** join/#asterisk neilbags (n=neilbags@c220-237-12-224.randw1.nsw.optusnet.com.au) |
| 11:05.39 | RoyK | QuAtRo[NL]: a renew is usually sent after half the lease time. renew != discover |
| 11:05.57 | *** join/#asterisk blitz[laptop] (n=blitzrag@217.41.232.24) |
| 11:08.18 | blitz[laptop] | hoi |
| 11:10.21 | RoyK | heya |
| 11:10.37 | X-Rob | lilo, our sympathies. |
| 11:11.35 | blitz[laptop] | what happened? Did I miss a wallop? |
| 11:11.51 | JT | freenode getting owned |
| 11:12.01 | blitz[laptop] | oh yah... noticed that |
| 11:12.36 | jpeeler | hey guys is there any reason why music on hold would start playing and then immediately stop? |
| 11:12.55 | blitz[laptop] | yes |
| 11:13.41 | jpeeler | blitz[laptop], such as? |
| 11:13.53 | blitz[laptop] | not sure -- but I'm sure there is a reason |
| 11:14.23 | blitz[laptop] | native or mpg123 MoH? |
| 11:14.41 | blitz[laptop] | mpg123 0.59r? Is the file an MP3 and VBR? |
| 11:14.51 | jpeeler | native... which as of asterisk 1.2 i thought could handle mp3 |
| 11:15.11 | blitz[laptop] | you're better off converting it to an actual native format like ulaw or gsm |
| 11:15.21 | blitz[laptop] | I think you can do that with sox |
| 11:15.44 | jpeeler | but if asterisk can handle mp3s natively, then why? |
| 11:16.02 | blitz[laptop] | so the CPU doesn't have to re-encode in real time |
| 11:16.28 | blitz[laptop] | mp3 to foo_codec is pretty intense |
| 11:16.45 | blitz[laptop] | if you just use some HD space and convert to as many formats as you're using, then the CPU doesn't have to really do anything |
| 11:16.50 | RoyK | jpeeler: also, having asterisk handle mp3 in realtime means each time musiconhold is started, an encoding is started. the system with a separate player process is quite a bit better |
| 11:19.18 | blitz[laptop] | HD space is much cheaper than CPU horsepower |
| 11:20.06 | blitz[laptop] | little? :) |
| 11:20.23 | RoyK | :) |
| 11:20.24 | RoyK | ok |
| 11:20.43 | RoyK | it's a 4U box, and it has a total of 16 drive bays, currently only using half of it |
| 11:20.53 | blitz[laptop] | aha -- so the truth comes out |
| 11:20.55 | RoyK | with an eight-port 3ware raid controller |
| 11:22.10 | RoyK | perhaps adding another eight 750gig drives might be nice :) |
| 11:28.55 | AltnTab | is there another way to read some variable from caller without using Read() |
| 11:30.02 | Luke-Jr | tzafrir: like RoyK said-- the IP isn't changing |
| 11:30.16 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 11:30.17 | Luke-Jr | it's just renewing the same IP |
| 11:30.45 | *** join/#asterisk donpaolo (n=donpaolo@pri-214-b7.codetel.net.do) |
| 11:30.56 | *** part/#asterisk donpaolo (n=donpaolo@pri-214-b7.codetel.net.do) |
| 11:32.17 | X-Rob | AltnTab, uh. WaitExten? Why are you so opposed to 'Read'? |
| 11:33.01 | AltnTab | X-Rob, i am entering some numbers to variable and the timeout is too little |
| 11:33.14 | X-Rob | You know you can adjust the timeout? |
| 11:33.15 | AltnTab | and response timeout don't work on Zap |
| 11:33.17 | X-Rob | 'show application read' |
| 11:33.21 | AltnTab | yes |
| 11:33.26 | AltnTab | i hae tryed it |
| 11:33.33 | X-Rob | obviously not hard enough. |
| 11:33.35 | AltnTab | with read() with set timeout response |
| 11:33.39 | X-Rob | try it _harder_ |
| 11:33.50 | X-Rob | try it after reading the documentation |
| 11:33.58 | X-Rob | (that always helps) |
| 11:34.30 | X-Rob | It definately works. I know this for a fact. |
| 11:34.40 | AltnTab | it works on SIP channel |
| 11:34.48 | AltnTab | the same way i define it on Zap |
| 11:35.01 | AltnTab | but on Zap it won't |
| 11:35.33 | AltnTab | on SIP it waits the exact time i am specifying in response timeout and/or Read() |
| 11:36.27 | AltnTab | My Zap is connected to GSM gate ... could this be the problem somehow ? |
| 11:36.38 | *** part/#asterisk blitz[laptop] (n=blitzrag@217.41.232.24) |
| 11:36.47 | RoyK | as the aussie says, try harder. that works. just like people who can't speak English understand it far better if you speak SLOWLY AND LOUD |
| 11:37.06 | AltnTab | ok |
| 11:37.57 | Luke-Jr | rofl |
| 11:51.47 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
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| 11:59.01 | *** join/#asterisk MaartenB_ (n=Maarten@82-217-61-38.cable.quicknet.nl) |
| 11:59.09 | MaartenB_ | high-rez, everyone |
| 11:59.17 | QuAtRo[NL] | I have declared in my extensions.conf: exten => _.,1,Goto(handleNummer,${EXTEN},1) |
| 11:59.23 | MaartenB_ | sorry, that should be "hi" :) |
| 11:59.52 | MaartenB_ | I am trying to install asterisk on a Fedora core 5 box, but unfortunatly I am unsuccesfull |
| 12:00.16 | QuAtRo[NL] | But when i check the output of Asterisk in the debug mode it seems that ${EXTEN} is s |
| 12:00.17 | MaartenB_ | it gives me a segmentation fault when i try to start it |
| 12:00.32 | MaartenB_ | has anyone experience with running Asterisk on Fedora Core 5? |
| 12:00.49 | QuAtRo[NL] | Which seems to be quite strange to me because of the fact that ${EXTEN} should return the dialed number |
| 12:02.57 | QuAtRo[NL] | Please correct me if i'm wrong |
| 12:04.30 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
| 12:06.14 | tRSS | I have question, how can I use the URL flag in the dial application? |
| 12:06.30 | tRSS | an example would simply help |
| 12:14.56 | MaartenB_ | is it possible to run asterisk on FC5 anyway, because I red some stuff about the kernel not being compaitble or something |
| 12:16.22 | MikeB | maartenb I just installed Asterisk on FC5 and it seems fine, I too read that it was no good with the stock kernel so I updated using yum update |
| 12:16.47 | Ritalin- | I ran asterisk on fc5 |
| 12:17.04 | MaartenB_ | how did you install it? |
| 12:17.14 | Ritalin- | I followed the instructions |
| 12:17.20 | MaartenB_ | I used the atrpm rpm's |
| 12:17.27 | MaartenB_ | you compiled it yourself? |
| 12:17.27 | Ritalin- | I used the source |
| 12:17.30 | Ritalin- | yes |
| 12:17.35 | MaartenB_ | and did you use any extra kernel modules? |
| 12:18.03 | Ritalin- | nothing extra. I had to use the zaptel modules but it did all of that automatically |
| 12:20.23 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
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| 12:24.03 | *** part/#asterisk littlejohn (n=little@host26-66.pool8716.interbusiness.it) |
| 12:29.10 | tRSS | what is the URL flag in Dial application? Can we send http URLs to a person answering a call? |
| 12:30.33 | RoyK | no, it is to dial using a SIP uri or whatever protocol you're using |
| 12:32.36 | tRSS | RoyK: basically, there is URL flag in the Dial application. e.g. exten => 85xx,1,Dial(SIP/user0,tT,http://www.google.com), what is the point of having a sip url near the end when we already placing a call to SIP/user0 |
| 12:32.37 | tRSS | ? |
| 12:35.04 | tRSS | and if I am incorrect, then is there a way to send a URL to a person answering the call? This would be helpful in a contact center environment. |
| 12:36.12 | *** join/#asterisk FoFiN (n=FoFiN@201.153.31.166) |
| 12:36.24 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 12:37.52 | *** join/#asterisk tdonahue (n=tdonahue@207.138.151.58) |
| 12:38.34 | Nugget | you can send an URI to the target of a call but it's unlikely that their SIP phone will have any idea what to do with it. |
| 12:39.47 | *** join/#asterisk coppice (n=chatzill@223.193.17.210.dyn.pacific.net.hk) |
| 12:40.03 | tRSS | Nugget: we are planning to develop a custom softphone for this purpose. but if we are using the SIP protocol, then it won't work? |
| 12:40.18 | Nugget | then you'll be fine. :) |
| 12:40.21 | tRSS | because, thats what I am understanding from your comment |
| 12:40.28 | tRSS | ooh perfect |
| 12:40.30 | tRSS | thanks |
| 12:41.14 | tRSS | infact, i believe xlite (or xten) are compatible with this |
| 12:41.29 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 12:41.34 | RoyK | methinks having customer support with sip/sip and sending an initial http://en.wikipedia.org/wiki/RTFM to the customers might will be nice |
| 12:42.57 | tdonahue | morning all |
| 12:44.15 | *** join/#asterisk JohnnyO (i=JohnnyO@65.5.100.102) |
| 12:44.20 | JohnnyO | Good morning all |
| 12:45.12 | RoyK | afternoon |
| 12:45.16 | JohnnyO | Needing to hire someone this AM to help configure asterisk@home with some Cisco 7940G phones |
| 12:45.42 | RoyK | good localtime() |
| 12:47.03 | JohnnyO | everyone must have a full belly :) |
| 12:47.13 | l-fy | not really |
| 12:47.25 | l-fy | but some of us have learn never to promise to do something with * |
| 12:48.16 | JohnnyO | I have a buddy who has the 7940Gs working well - but he's out of country on vacation and I have a clientn breathing down my neck |
| 12:50.35 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:50.52 | JohnnyO | no takers ? I have cash / paypal / Amex / Visa / MC - whatever it takes |
| 12:51.38 | tzanger | JohnnyO: I havent' dont cisco config with *. lots of polycom though |
| 12:52.01 | Ritalin- | johnny: i just barely got my system up. otherwise i'd help |
| 12:53.32 | JohnnyO | no problem |
| 12:53.46 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 12:54.03 | JohnnyO | I found some great how to's - I just don't know how to ;) |
| 12:55.53 | rob0 | JohnnyO: there are consultants listed on the Wiki. |
| 12:55.58 | rob0 | ~wiki |
| 12:56.02 | rob0 | ~wikis |
| 12:56.07 | jbot | hmm... wikis is http://www.voip-info.org |
| 12:56.07 | Creperum | can I make all the calls though my ZAP card go with custom CID? |
| 12:57.41 | RoyK | Zdocs |
| 12:57.43 | RoyK | ~docs |
| 12:57.45 | jbot | well, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 12:58.52 | rob0 | I'd take the job, but 1. I'm going back to bed, up most of the night; and 2. I'm not really qualified at a professional level (yet) with Asterisk. |
| 12:59.45 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 12:59.55 | Creperum | hey... |
| 13:06.33 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
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| 13:11.37 | *** part/#asterisk kizmet (n=kizmet@00-nsi-ad-act.au.argon.net.au) |
| 13:11.52 | Luke-Jr | JohnnyO: drop the "@home" part and you might get someone |
| 13:13.40 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:13.40 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:14.34 | JohnnyO | Luke-Jr - any suggestions ? |
| 13:15.11 | _problem_ | Hi all..I have a serious asterisk configuration issue can anyone help me?? |
| 13:15.13 | Luke-Jr | my suggestion is to drop @home from your requirement =p |
| 13:15.46 | JohnnyO | I don't have a problem with that - as long as I can still use the web interface for configurations |
| 13:16.03 | JohnnyO | my issue is - I have never done this before and I am a total linux idiot :) |
| 13:18.31 | _problem_ | anyone free there to listen me ?? |
| 13:24.05 | JohnnyO | Thanks for the lead of the * consultants list |
| 13:24.16 | JohnnyO | found someone local to me that's ready to rock |
| 13:24.27 | JohnnyO | Mucho appreciado ! |
| 13:26.17 | *** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 13:30.31 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 13:31.45 | Vorondil | good morning all. quick question: is there anybody around who's upgraded asterisk on a freebsd machine (using ports) from 1.2.7.1 to 1.2.9.1? i'm mainly curious as to how long it should take |
| 13:31.54 | *** join/#asterisk fulgas (n=fulgas@82.102.2.30) |
| 13:33.14 | fulgas | heys |
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| 13:52.07 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 13:54.24 | *** join/#asterisk ToyMan (n=stuq@74-32-78-58.dsl1.mdl.ny.frontiernet.net) |
| 13:54.30 | *** join/#asterisk scud (n=scud@12-218-71-230.client.mchsi.com) |
| 13:55.47 | scud | what is the package to for the voice recognition to turn it into text? |
| 13:55.55 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 13:57.07 | *** join/#asterisk |oranjia| (n=kvirc@dsl-146-14-06.telkomadsl.co.za) |
| 13:57.28 | |oranjia| | hello world :) |
| 13:57.33 | sevard | hola |
| 13:57.41 | |oranjia| | hi sevard :) |
| 13:57.50 | sevard | what's poppin dawg |
| 13:57.56 | l-fy | hi |oranjia|, Sedorox |
| 13:58.07 | *** join/#asterisk sep (n=sep@217.17.211.40) |
| 13:58.21 | sevard | Sedorox idle 8 hours 48 minutes 28 seconds |
| 13:58.51 | Sedorox | ea? |
| 13:59.07 | sevard | <PROTECTED> |
| 13:59.10 | |oranjia| | Hey l-fly |
| 13:59.15 | l-fy | l-fy |
| 13:59.16 | Sedorox | ok then... |
| 13:59.38 | |oranjia| | lol |
| 14:00.42 | *** join/#asterisk tuck3r_ (n=tuck3r@unaffiliated/tuck3r) |
| 14:01.33 | *** join/#asterisk Waverly360 (n=mirc@adsl-157-61-36.bna.bellsouth.net) |
| 14:04.12 | mut | http://www.icegn.net/junk/Loose_Wheel.wmv |
| 14:04.13 | mut | owwwww |
| 14:04.25 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.232.207.Dial1.SanJose1.Level3.net) |
| 14:05.50 | *** part/#asterisk tuck3r_ (n=tuck3r@unaffiliated/tuck3r) |
| 14:07.25 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 14:07.43 | *** join/#asterisk lilalinux (i=e-trolle@deepthroat.deswahnsinns.de) |
| 14:07.51 | lilalinux | hey guys |
| 14:08.43 | lilalinux | I would like to get a mail for each missed call, any hints? |
| 14:11.56 | Ritalin- | computers are stupid |
| 14:12.16 | mut | yea, kinda like rain man |
| 14:12.22 | mut | not really smart but good at math |
| 14:15.01 | *** join/#asterisk notjason (n=notjason@ool-457183bb.dyn.optonline.net) |
| 14:16.19 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 14:16.19 | *** join/#asterisk Eggplant (i=No@dsl-72-19-44-253.cascadeaccess.com) |
| 14:17.30 | *** join/#asterisk Spy000007 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 14:23.41 | *** join/#asterisk Katty (n=aisaacs@64.82.232.54) |
| 14:24.02 | Katty | morning. |
| 14:25.01 | *** join/#asterisk nounoursfr (n=nounours@213.161.196.217) |
| 14:25.06 | nounoursfr | hi all |
| 14:25.24 | nounoursfr | do you have test IAX and Dtmf in france ? |
| 14:27.52 | Ritalin- | morning Katty |
| 14:34.31 | *** join/#asterisk MstlyHrmls (n=mh@66.195.193.151) |
| 14:36.03 | *** join/#asterisk bon (n=bon@voip.in.radiolan.sk) |
| 14:37.04 | bon | can i somehow manage CallerID forwarding? one gateway with 10 lines and i want to forward the originating numbers, not the gw's number |
| 14:37.21 | *** join/#asterisk DrDeke (i=Rusty@causticsoda.engin.umich.edu) |
| 14:40.13 | MaartenB_ | high-rez, I managed to install asterisk |
| 14:40.26 | MaartenB_ | should I be able to do "telnet localhost 5060" now? |
| 14:40.26 | Nugget | telnet is eeeeeeevil! |
| 14:40.34 | MaartenB_ | because that doesn' t work |
| 14:41.04 | high-rez | Was that actually to me? Cause I just woke up. |
| 14:41.18 | MaartenB_ | I am sorry, no |
| 14:41.31 | MaartenB_ | I typed hi, xchat auto-completed it :) |
| 14:41.51 | DrDeke | MaartenB: No, you should not be able to telnet localhost 5060; asterisk listens for SIP connections via UDP, whereas Telnet makes a TCP connection. |
| 14:41.53 | high-rez | It's all good. You can run "netstat -an" though to see which address/port the SIP interface bound to if that's the question. :) |
| 14:42.45 | MaartenB_ | I want to see if I can connect, without using a voip client |
| 14:43.00 | high-rez | E.g: |
| 14:43.01 | MaartenB_ | there isn't a telnet alternative to see if it connects? |
| 14:43.03 | high-rez | gus@cp01 ~ $ netstat -an | grep 5060 |
| 14:43.03 | high-rez | udp 0 0 0.0.0.0:5060 0.0.0.0:* |
| 14:43.04 | DrDeke | MaartenB: I don't know of an easy way to do that, no. |
| 14:43.14 | DrDeke | MaartenB: Since UDP is not a connection-oriented protocol. |
| 14:43.17 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 14:43.24 | MaartenB_ | ok, I see |
| 14:43.34 | high-rez | See what I just pasted Maarten, that'll tell you at least if it's bound. |
| 14:43.40 | DrDeke | yeah |
| 14:44.08 | MaartenB_ | it is bound :) |
| 14:44.43 | jbalcomb | Does anyone use a system for keeping track of all thier IP phones? |
| 14:44.57 | DrDeke | In what sense? |
| 14:45.14 | MaartenB_ | I am going to reboot, my audio is not working correctly, be back later... |
| 14:45.23 | high-rez | Congrats, when you get it setup to dialk out to the PSTN in NL let me know and I'll trade you outbound to the US :D |
| 14:45.42 | jbalcomb | Tracking MAC Address, IP Address, and Extension is what I'd like to do for now. |
| 14:45.54 | DrDeke | Yeah, if you need to place calls to the ANARMIMN (734-222 and so on) exchange in the US, i'd be happy to oblige, although it's a cruddy FXO card and has a lot of what sounds like 60Hz hum... |
| 14:46.07 | kay2 | What is iax trunk used for ? |
| 14:46.37 | jbalcomb | Moving in the direction of making a web page for the help desk guys so they can add/remove/change phone configs. |
| 14:47.04 | bon | anyone doing callerid forwarding? |
| 14:47.08 | DrDeke | jbalcomb: Yeah... I don't know of anything that does exactly that, but it sounds like something that could probably be custom coded... |
| 14:47.14 | DrDeke | (custom coded fairly easily, I mean) |
| 14:47.29 | *** join/#asterisk countjasper (n=countjas@12.108.22.201) |
| 14:47.30 | RoyK | erm... how can I tell asterisk 'stop this CDR log HERE!' while the call is still active? |
| 14:47.30 | high-rez | Anyone know of a provider that does flat rate to romania? :/ |
| 14:47.38 | jbalcomb | DrDeke: I am working on doing that now. I am not much of a coder though and additionaly am poor at designing systems. :/ |
| 14:47.47 | countjasper | hello, how can i switch my calls from going through zap and straight to sip |
| 14:47.47 | DrDeke | jbalcomb: Yeah, same here :) |
| 14:47.56 | DrDeke | Practice makes perfect though, at least in theory :) |
| 14:48.38 | jbalcomb | DrDeke: I'm running arpwatch to collect the MAC & IP. Now I'm cut&paste coding a Perl script to monitor the log real-time and upload the data to a MySQL DB. |
| 14:49.15 | DrDeke | jbalcomb: That's how I would approach it too. |
| 14:49.37 | jbalcomb | DrDeke: Then I'm going to write a script that pulls the IP from the DB, queries the AstDB to get the Extension for that IP, and adds that info to the DB. |
| 14:50.22 | *** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
| 14:50.31 | jbalcomb | DrDeke: The same script will pull the User Name from the configs for that extension and add that info to the DB as well. |
| 14:51.19 | bon | hmm.. so i suppose it's not possibble to get the originating callerid from the user and forward it through sip channel further to voice provider? |
| 14:51.19 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) |
| 14:51.38 | DrDeke | jbalcomb: yep |
| 14:51.40 | rajiv|work | howdy all. any pointone employees around ? |
| 14:52.27 | jbalcomb | DrDeke Assming I can get all the information necessary to build the the config for each phone, I will have a table that hold the username and any custom configuration option. A php script will use all the info to build the config file, run the encoder app, and place the config in the appropriate tftp directory. |
| 14:53.12 | countjasper | how doi swtich all my calls from instead of going through zap going through SIP |
| 14:53.31 | jbalcomb | DrDeke: another table with the first half of the MAC Address with Vendor information should let me know which phone I'm making a config for so I can have one system for all 5 of our phone brands/models. |
| 14:54.04 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:54.07 | DrDeke | jbalcomb: yeah, sounds like it should be pretty solid... except how will you tell one model from another if they are both by the same vendor? |
| 14:55.57 | jbalcomb | DrDeke: Considered that as I typed it and I don't know. Perhaps a wget on the web interface? Perhaps it would be simpler to just make the helpdesk guys pick which phone it is. |
| 14:56.31 | DrDeke | jbalcomb: Yeah, that'd probably be the easiest method :) |
| 14:57.03 | DrDeke | Although yeah, wget on the web interface might work if you can find something to distinctively identify each model, which seems reasonably likely. |
| 14:57.07 | jbalcomb | DrDeke: Yeah, or convince the controller that I need to phase out three of the models because we only two or three of each. |
| 14:57.13 | DrDeke | yeah :) |
| 14:57.59 | jbalcomb | DrDeke: If anyone around here understood TCO or even just considered labor costs it would be an easy argument. :/ |
| 14:58.04 | DrDeke | hehe yeah |
| 14:59.04 | jbalcomb | DrDeke We had a two hour meeting to discuss whether we should switch our name server information on Network Solutions because we had the old IPs listed and were port forwarding port 53 traffic. |
| 14:59.14 | DrDeke | hahahahaha |
| 14:59.18 | DrDeke | that sounds just like this place :) |
| 14:59.21 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 14:59.46 | jbalcomb | DrDeke: I figure with the five of us in the meeting it cost over $200 to make the decision. |
| 14:59.55 | DrDeke | yup :) |
| 15:00.08 | DrDeke | sort of like that whiteboard in office space: "PLANNING TO PLAN" :) |
| 15:00.09 | *** join/#asterisk speedwagon (n=Ariel@70.46.87.158) |
| 15:00.30 | jbalcomb | I think when we propose having a meeting it should be required to estimate the labor cost. ;) |
| 15:00.39 | *** join/#asterisk goldsmurf (n=rgoldber@h216.70.9.114.superiorbroadband.com) |
| 15:01.01 | DrDeke | but then everybody wouldn't be able to waste all their time in meetings *grin* |
| 15:01.06 | coppice | a decision for only $200 sounds like a bargain. civil servants would never keep it so cheap :-) |
| 15:01.15 | jbalcomb | DrDeke: yeah, exactly. "Hey boss, I'd like to have a $387 meeting this afternoon to discuss the color of the walls in the server room." |
| 15:01.28 | l-fy | uhhhhhhhh |
| 15:01.32 | l-fy | why was 200USD? |
| 15:01.50 | *** join/#asterisk smackus2 (n=smackus2@c-67-182-209-194.hsd1.ut.comcast.net) |
| 15:01.54 | jbalcomb | coppice: haha.. yeah, I suppose there is that. I have a buddy that works for the City of Cleveland and his stories on the topic are comically. |
| 15:02.19 | jbalcomb | l-fy: 1 manager, 3 admins, 1 help desk guy = $$$labor cost$$$ |
| 15:02.19 | DrDeke | and I have to agree |
| 15:03.07 | smackus2 | I am getting an error that i believe is ultimately killing asterisk. I cannot find anything on google. that can help me out. has anyone seen this, or can tell from the error what is happeing? |
| 15:03.09 | smackus2 | http://pastebin.ca/72092 |
| 15:03.37 | DrDeke | smackus: I've never seen that one... |
| 15:03.44 | l-fy | smackus2 |
| 15:03.44 | l-fy | Jun 24 10:37:45 ERROR[14139] chan_sip.c: We could NOT get the channel lock for SIP/3508-a524! |
| 15:03.53 | l-fy | that message is really really bad |
| 15:03.55 | jbalcomb | smackus2 what is sip/3508? |
| 15:04.01 | smackus2 | it is an extension |
| 15:04.08 | l-fy | try to use ulimit -c 4096 or something bigger when you start yate |
| 15:04.11 | jbalcomb | yeah, whos? which? what phone? |
| 15:04.24 | *** join/#asterisk saftsack (n=w@p54A7DBF4.dip.t-dialin.net) |
| 15:04.25 | saftsack | hi |
| 15:04.31 | smackus2 | there is a polycom phone out on the call center floor. |
| 15:04.32 | l-fy | jbalcomb > is not funny at all |
| 15:04.41 | smackus2 | extension 3508 |
| 15:04.53 | l-fy | smackus2> try to use ulimit -c 4096 or something bigger when you start yate |
| 15:04.54 | saftsack | is there any possibility to get digiums PRI card with the hardware echocancelling for testing? |
| 15:04.55 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 15:04.58 | l-fy | sorry |
| 15:05.02 | l-fy | replace yate with * |
| 15:05.07 | l-fy | sorry |
| 15:05.28 | smackus2 | l-fy: I am not sure what you are telling me to do. |
| 15:05.35 | smackus2 | oh |
| 15:05.37 | kay2 | When I trunk two asterisks, do I share the dialplan ? |
| 15:05.37 | smackus2 | ok |
| 15:05.42 | l-fy | smackus2 > how do you run * ? |
| 15:05.54 | smackus2 | "/usr/sbin/asterisk &" |
| 15:05.57 | l-fy | ok |
| 15:06.03 | l-fy | before doing that do something like |
| 15:06.08 | bon | in which variable is the originating CallerID stored when receiving a call? |
| 15:06.10 | l-fy | ulimit -c 9068 |
| 15:06.20 | smackus2 | explain to me what that will do |
| 15:06.25 | l-fy | and is possibile depending on the distro to help |
| 15:06.37 | l-fy | will increse the number of open file by a single process |
| 15:06.47 | smackus2 | oh, ok |
| 15:07.01 | l-fy | long long story but it may help |
| 15:07.10 | l-fy | which distro do you have? |
| 15:07.10 | smackus2 | so just type "ulimit -c 9068" then "/usr/sbin/asterisk &"? |
| 15:07.14 | l-fy | yes |
| 15:07.17 | smackus2 | red hat enterprise |
| 15:07.21 | l-fy | uhhh |
| 15:07.26 | *** join/#asterisk pdtmobile (n=ptinsley@adsl-157-61-36.bna.bellsouth.net) |
| 15:07.27 | l-fy | try to use fedora core |
| 15:07.30 | l-fy | it may help |
| 15:07.40 | smackus2 | ok... why FC? |
| 15:07.42 | jbalcomb | red hat enterprise is recommended by digium |
| 15:07.48 | l-fy | because you use already rhe |
| 15:07.56 | l-fy | and because rhe has some weird patches |
| 15:08.08 | riksta | l-fy, no offence, but your solutions seem rather sporadic and not particularly helpful |
| 15:08.10 | l-fy | jbalcomb > i never actually cared about what digium say |
| 15:08.21 | l-fy | riksta> true, but it may work |
| 15:08.30 | l-fy | and more than that i know why it may work |
| 15:08.36 | riksta | l-fy, but you can't tell him to just install another OS to remedy the problem |
| 15:08.47 | MaartenB | I managed to get connected to my asterisks now too |
| 15:08.49 | l-fy | riksta > i've told him to try |
| 15:08.50 | MaartenB | yeah :) |
| 15:08.51 | jbalcomb | yeah but reinstalling the OS is a big early step |
| 15:08.54 | MaartenB | but I don't hear anything |
| 15:09.00 | l-fy | jbalcomb > it can try |
| 15:09.11 | MaartenB | the quickstart says that I should get some autoresponder on number 1000 |
| 15:09.11 | l-fy | i didn't say i'm 100% sure, i've said that it may help |
| 15:09.53 | jbalcomb | smackus2 is that the only extension getting the message? try rebooting the phone, changing its extension, doing a sip prune on it... |
| 15:10.02 | bon | once again please, i can't google that up |
| 15:10.06 | bon | in which variable is the originating CallerID stored when receiving a call? |
| 15:10.16 | *** join/#asterisk bmg505 (n=leon@c1-230-12.rndf.isadsl.co.za) |
| 15:10.18 | l-fy | jbalcomb > that stuff is an obvious issue in asterisk |
| 15:10.26 | l-fy | the phone has nothing to do with that |
| 15:10.30 | jbalcomb | bon: consider going thru the source/ |
| 15:10.31 | riksta | bon, voip-info.org |
| 15:10.37 | l-fy | it's a mutex that is blocked for god sake |
| 15:10.40 | DrDeke | bon: sorry, i'm not sure. i'm pretty sure it's on voip-info.org though |
| 15:11.09 | jbalcomb | l-fy oki doki, i guess this one's all you then. |
| 15:11.21 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 15:12.15 | l-fy | jbalcomb > is just happening that i know that problem and is a bug almost for sure in * |
| 15:12.44 | riksta | l-fy, and he has tried to roll back to a previous version? |
| 15:12.47 | l-fy | seems to me like a corupted memory or even better a deadlock |
| 15:13.00 | l-fy | riksta > don't ask me |
| 15:13.06 | jbalcomb | l-fy: strange that it's that obvious though and there is nothing about it on google or the wiki. |
| 15:13.20 | riksta | bugtracker will have something |
| 15:13.35 | l-fy | jbalcomb > it just happend that i know some stuff about mutexes |
| 15:13.43 | l-fy | and the message there is clear |
| 15:14.03 | jbalcomb | good enough for me |
| 15:14.08 | l-fy | Jun 24 10:37:45 ERROR[14139] chan_sip.c: We could NOT get the channel lock for SIP/3508-a524! |
| 15:14.27 | l-fy | this bassically means that chan_sip can't take the mutex for the sip_pvt probably |
| 15:14.30 | l-fy | which means 2 things |
| 15:14.42 | bon | ok |
| 15:14.45 | l-fy | you either have a corupted memory in the mutex area |
| 15:14.49 | bon | i've found some info on this issue in Dial cmd |
| 15:14.51 | bon | but no help :) |
| 15:14.52 | bon | o: Restore the Asterisk v1.0 CallerId behaviour (send the original caller's ID) in Asterisk v1.2 (default: send this extension's number) |
| 15:15.08 | bon | that's what i want, to send the originating callerid, not the extension's number |
| 15:15.39 | l-fy | or you have a deadlock and someone is trying to take the sip_pvt mutex while sip_pvt mutex is blocked in another thread that can't free it because it waits for a mutex already tooken by the first thread |
| 15:15.46 | l-fy | it's classic threading programming |
| 15:15.47 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:16.17 | *** join/#asterisk mrtwister (n=ambervoi@107.250.broadband5.iol.cz) |
| 15:17.00 | scud | can i use the lumenvox speech recognition without the purchase of the BE? |
| 15:17.10 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:17.24 | *** join/#asterisk blop (n=blop@213-193-176-50.adsl.easynet.be) |
| 15:17.36 | scud | i guess i could also ask, is it even available outside of the business edition? |
| 15:17.47 | *** join/#asterisk umay (n=chris@71-208-188-148.hlrn.qwest.net) |
| 15:18.11 | blop | hi, got a problem while upgrading to zaptel 1.2.6 :: FATAL: Error inserting zaptel (/lib/modules/2.6.15/misc/zaptel.ko): Invalid module format // zaptel: version magic '2.6.15 K7 gcc-4.1' should be '2.6.15 K7 gcc-4.0' , any idea ? |
| 15:18.41 | *** join/#asterisk Splas (n=jwb@206.252.198.101) |
| 15:18.56 | eKo1 | blop: sounds like a gcc problem |
| 15:19.00 | *** join/#asterisk salviadud (n=ralfalfa@201.145.29.99) |
| 15:19.09 | bon | version mismatch |
| 15:19.11 | bon | in gcc i suppose |
| 15:19.15 | blop | yeah =) i got that too |
| 15:19.23 | pdtmobile | lol |
| 15:19.46 | Vorondil | hi all. quick question: is there anybody around who's upgraded asterisk on a freebsd (6.0-release, in my case)machine (using ports) from 1.2.7.1 to 1.2.9.1? i'm mainly curious as to how long it should take |
| 15:20.08 | Vorondil | for planning downtime and whatnot |
| 15:20.45 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 15:21.21 | bon | nah, if just someone helped me with the callerid forwarding.. :/ |
| 15:21.44 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:23.38 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 15:23.38 | *** mode/#asterisk [+o denon] by ChanServ |
| 15:24.04 | *** join/#asterisk Persilon (n=ajolodov@200.123.112.152) |
| 15:24.12 | Persilon | Hi |
| 15:24.23 | _problem_ | Hi..can anybody please help me what is the meaning of this line in asterisk logs |
| 15:24.26 | _problem_ | <PROTECTED> |
| 15:24.45 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:24.45 | Persilon | is it okay that Background() hangs up inmediatly after playing the file ? |
| 15:25.23 | nounoursfr | you caller Id is french |
| 15:25.59 | eKo1 | _problem_: It means that the call from SIP/4362 has finished. |
| 15:26.02 | pdtmobile | Persilon: do you mean ok for the system? |
| 15:26.34 | Persilon | pdtmobile: ok because I can't send dtmf after it finishes... it hangs up |
| 15:26.35 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 15:26.40 | bon | hm, my callerid is always being taken out of the mysql database using * realtime |
| 15:26.40 | _problem_ | eKo1: but i have this log when i dial on that extension |
| 15:26.45 | Persilon | pdtmobile: and I can't get the t extension to work after that |
| 15:26.50 | bon | is there a possibbility to use the one that client sends? |
| 15:27.04 | lilalinux | how do i detach from CLI? |
| 15:27.07 | eKo1 | _problem_: that's because it is hanging up |
| 15:27.08 | pdtmobile | are you doing a WaitExten after the Background? |
| 15:27.08 | bon | lilalinux: quit |
| 15:27.22 | bon | lilalinux: or exit, i can't remember now |
| 15:27.33 | lilalinux | both is unknown |
| 15:27.39 | Persilon | pdtmobile: I have background() and then t,1,goto |
| 15:27.42 | eKo1 | lilalinux: either `quit' or if that doesn't work use ! to get a shell |
| 15:28.04 | pdtmobile | Persilon: are you trying to do a menu, like press 1 for sales 2 for monkeys... ? |
| 15:28.12 | Persilon | pdtmobile: exactly |
| 15:28.26 | _problem_ | eKo1: when i try to call a offline extension ..i gets these entries on CLI and they gets entered into mysql database also..what should i do remove them |
| 15:28.54 | bon | ${CALLERIDNUM} < what should that variable contain when receiving calls? |
| 15:28.57 | Persilon | pdtmobile: should it be background(), waitexten(5),t ? |
| 15:29.22 | eKo1 | _problem_: you don't. leave them there |
| 15:29.57 | pdtmobile | BackGround(soundfile) WaitExten(5) then after that as an extension you have t as well as the options you want people to enter |
| 15:30.21 | Persilon | pdtmobile: Thank you, it worked :) |
| 15:30.23 | pdtmobile | like exten => 2,1,Playback(tt-monkeys) |
| 15:30.40 | *** join/#asterisk DrDeke (i=Rusty@causticsoda.engin.umich.edu) |
| 15:30.41 | pdtmobile | Persilon: no problem |
| 15:30.50 | _problem_ | eko1: ok fine, but it is corrupting the cdr database..it is entering 60-70 entries for one call attempt..which is not yet a complete call..just an attempt of call |
| 15:31.21 | *** join/#asterisk smackus (n=smackus@c-67-182-209-194.hsd1.ut.comcast.net) |
| 15:31.36 | Persilon | pdtmobile: what the caller sends via dtmf is set as the priority ? if he dials 1 it'll go to s,1 right ? |
| 15:32.01 | smackus | sorry, I had to drop off. so, we decided that it was not necessary to reinstall my OS, right? |
| 15:32.16 | tdonahue | i'm having some issues getting early media working both on my PRI and my SIP connection. Anyone have any recommendations to get it running? |
| 15:32.19 | pdtmobile | no, not the priority, the extension, so you would have a bunch of options,1 in your dialplan in the context where your menu lives |
| 15:32.43 | pdtmobile | the exten line i put above would be called if somebody hit 2 in the menu |
| 15:32.49 | eKo1 | _problem_: then someone needs to stop dialing that number so many times. |
| 15:33.36 | Persilon | pdtmobile: ok, thank you :) |
| 15:33.48 | pdtmobile | Persilon: np |
| 15:35.05 | Persilon | pdtmobile: another question, in all the documentation I've found incoming calls are in the [incoming] context using the s extention. is that only for fxs/fxo calls or it works for sip too ? |
| 15:35.32 | *** join/#asterisk smackus (n=smackus@c-67-182-209-194.hsd1.ut.comcast.net) |
| 15:35.48 | pdtmobile | sip/pri/iax can get calls based on the number called so you would handle them in a context and list the valid numbers that type of connection can call |
| 15:35.51 | _problem_ | eKo1: but thats not possible because i have some softphones connected with asterisk...and i can attempt to call an offline extension..i will not know if that is offline..for 1 call attempt i got many entries...that is unjustifiable |
| 15:36.14 | pdtmobile | so similar to how you are doing your menu, you would have a context with the things you want to let people dial via sip and put that sip phone in that context |
| 15:36.55 | lilalinux | can asterisk receive fax, too? |
| 15:37.17 | pdtmobile | Persilon: make sense? |
| 15:37.26 | eKo1 | _problem_: if you get many entries, then there is something in your dialplan that is redialing the number multiple times. |
| 15:37.32 | Persilon | pdtmobile: sorry for my english :P |
| 15:37.51 | pdtmobile | oh, no problem |
| 15:37.51 | *** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com) |
| 15:37.58 | Persilon | pdtmobile: all examples regarding inbound calls use the s extention in the [incoming] context |
| 15:38.03 | Ritalin- | anyone using teliax? |
| 15:38.11 | pdtmobile | Ritalin-: i am at the hous |
| 15:38.13 | l-fy | smackus > did it worked? |
| 15:38.27 | Ritalin- | pdtmobile: how long did it take for your callerid to start showing up on outbound calls? |
| 15:38.43 | _problem_ | eko1: ya ok i believe that can be dialplan issue...but i dont have anything configured to dial one number again and again...but can u plz help me in rectifying my dial plan...if it is dial plan issue |
| 15:39.06 | Persilon | pdtmobile: I'm using sip as asterisk protocol and the only way I could make it work was using incoming calls in the same context I use for my interns with another extention number |
| 15:39.12 | pdtmobile | Ritalin-: immediately, their little callerid thing is finicky you have to format it JUST right for it to work. Your best bet is to just set your caller id on your asterisk box in your dialplan as you make calls |
| 15:39.22 | Spy000007 | teliax... ugh |
| 15:39.23 | pdtmobile | Ritalin-: it will accept whatever you send across the IAX connection |
| 15:39.30 | smackus | wont know... it is intermittent. goes for about 3 days then crashed, then I can reload and it crashes 2 min later. |
| 15:39.39 | fholmes | I am having some issues with agent queues. When I setup my AgentCallbackLogin(||${CALLERIDNUM}@mycontext) what context should I refer to? the incoming context in my extensions.conf or my context my queue is defined in my extensions.conf or somehting else? |
| 15:40.14 | eKo1 | _problem_: pastebin the relevant dialplan somewhere |
| 15:40.16 | Ritalin- | pdtmobile: right now i'm using just one SIP phone and it's displaying my DID but not my name. i've had it setup in the teliax portal to display name and number (according to their directions) for a few weeks and it hasn't |
| 15:40.30 | bon | hm, that's strange behaviour, if callerid is set to null in the realtime db, than the originating callerid is sent |
| 15:40.55 | pdtmobile | Persilon: you can put the sip connection in any context you want. but in that context you have to specify the things you want it to be able to dial |
| 15:41.02 | _problem_ | eKo1: can i send u personally that dial plan and logs |
| 15:41.03 | Spy000007 | it's unlikely that you'll be able to send name with teliax |
| 15:41.05 | l-fy | smackus > try a different asterisk version |
| 15:41.14 | l-fy | and try if is possibile a different distro |
| 15:41.17 | eKo1 | _problem_: no. please use pastebin |
| 15:41.21 | Spy000007 | you need a provider that can set the names in the central cnam databases |
| 15:41.30 | pdtmobile | Ritalin-: good luck with the name, i don't know if they submit names to anybody |
| 15:41.30 | l-fy | i still consider mandriva to be the best server distro ever |
| 15:41.33 | smackus | ok, what version of asterisk do you recommend? |
| 15:41.33 | Persilon | pdtmobile: do I have to put an s friend in sip.conf ? |
| 15:41.42 | Spy000007 | voicepulse can do it if you ask them specifically for a did where name can be set |
| 15:41.43 | pdtmobile | Ritalin-: all you really send in a call is the # it's up to the database lookups to find the name |
| 15:41.47 | l-fy | smackus > i don't use asterisk so don't ask me |
| 15:42.07 | l-fy | is just a common sense advice which was made before by jbalcomb |
| 15:42.09 | Ritalin- | pdtmobile: well a number is a start I guess. the # shows up. thanks |
| 15:42.09 | smackus | interesting that you are giving advice on asterisk if you do not use it. |
| 15:42.27 | _problem_ | eko1: i pasted that 2 days ago ..which is no more present but nobody responded |
| 15:42.49 | l-fy | smackus > i know some stuff about asterisk |
| 15:42.53 | smackus | ok |
| 15:42.57 | smackus | just an observation |
| 15:43.00 | l-fy | like for example his main design and how it's working |
| 15:43.00 | pdtmobile | smackus: if you want to pole vault it's all in the knees |
| 15:43.33 | Waverly360 | I've never pole vaulted..but I stayed at a Holiday Inn once. |
| 15:43.42 | smackus | pdtmobile: ?... i have bad knees... no pole vault for me |
| 15:43.47 | smackus | what do you mean? :-D |
| 15:43.54 | pdtmobile | i duno, never pole vaulted |
| 15:44.19 | smackus | wtf? |
| 15:44.36 | *** join/#asterisk jart (n=jart@justin.ctlinc.com) |
| 15:44.48 | pdtmobile | lol, just having fun with the asterisk comment. Giving advise on something I don't do. (lack of sleep, you'll have to excuse me) |
| 15:45.00 | Waverly360 | It was a bit comical... :P |
| 15:45.01 | jart | has anyone ever had problems with MeetMe volume using tdm cards? |
| 15:45.02 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 15:45.04 | *** join/#asterisk bigbdfs (n=bwarner@65.113.208.19) |
| 15:45.21 | pdtmobile | advice even |
| 15:45.28 | bigbdfs | has anyone tried registering a motorola vt1005 to asterisk? if so..how? |
| 15:46.56 | bigbdfs | has anyone tried registering a motorola vt1005 to asterisk? if so..how? |
| 15:47.09 | jart | for some reason, meetme has really high volume for internal ip phone users, and the volume is very very low for incoming tdm callers, to the point where they can't hear each other |
| 15:47.21 | jart | bigbdfs: no sorry :( |
| 15:47.49 | bigbdfs | thanks jart.. |
| 15:47.55 | bigbdfs | anyone else? |
| 15:48.03 | bigbdfs | anyone tried registering a motorola vt1005 to asterisk? if so..how? |
| 15:49.52 | bigbdfs | anyone tried registering a motorola vt1005 to asterisk? if so..how? |
| 15:50.36 | bugz | bigbdfs: i have, but the third time you asked i decided to tell you to RTFM. |
| 15:50.57 | DrDeke | bigbdfs: sorry, no. also, you don't need to repeat yourself this often |
| 15:51.15 | bigbdfs | bugz>sorry...didn't know if anyone was avail... |
| 15:51.36 | bon | flooding doesnt help |
| 15:51.52 | *** join/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) |
| 15:52.53 | pdtmobile | thats a good idea, I have one of those motorola boxes sitting at the house collecting dust |
| 15:52.57 | pdtmobile | how do you go about that? |
| 15:53.37 | Ritalin- | why does espresso taste so damn good |
| 15:53.59 | bigbdfs | bugz>you have a vt registered to asterisk then? |
| 15:54.00 | pdtmobile | Ritalin-: I duno, I just discovered a new drink today. Dirty Chai |
| 15:54.00 | Waverly360 | Ritalin-: Caffeine |
| 15:54.21 | CoaxD | Ritalin: espresso DOESNT taste so damn good. |
| 15:54.25 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 15:54.41 | CoaxD | Ritalin: The stuff tastes like tar. |
| 15:54.43 | pdtmobile | Ritalin-: Chai and Espresso |
| 15:55.05 | Waverly360 | CoaxD: Nah...tar's a bit more chewy and sticks to the roof of your mouth....don't ask how I know. |
| 15:55.17 | Ritalin- | coaxd: you've not had good espresso then |
| 15:56.04 | Ritalin- | pdtmobile: one thing i like to make is a caramel soy chai latte, add a double of espresso. i've never had any luck ordering that out anywhere. but what i make tastes awesome |
| 15:56.21 | pdtmobile | hmm, *takes notes* |
| 15:56.26 | bigbdfs | bugz>you avail for some ?'s on the moto vt? |
| 15:56.39 | CoaxD | Ritalin: heh |
| 15:56.59 | mcf3782 | I remember reading at some point that Asterisk had issues with or above a certain version of mpg123 for MOH. Is that still the case? |
| 15:57.35 | jart | mp3 music on hold is a bad idea in the first place |
| 15:57.40 | DrDeke | jart: why? |
| 15:57.48 | jart | especially due to the way it's programmed |
| 15:58.07 | jart | because it spawns a thread that transcodes the mp3 music for each channel |
| 15:58.15 | DrDeke | What's wrong with that? |
| 15:58.24 | jart | so if you have 3 people on hold, you have three mpg123 processes transcoding from mp3 to slinear |
| 15:58.29 | DrDeke | sure |
| 15:58.35 | jart | and if the channel is already transcoding.... |
| 15:58.39 | jart | you'll run out of cpu fast |
| 15:58.57 | jart | it takes 2 seconds to sox the mp3 over to ulaw |
| 15:59.01 | jart | and use moh-native |
| 15:59.16 | jart | and ulaw uses zero cpu |
| 15:59.18 | DrDeke | sure; but for a setup that is nowhere near cpu-bound is there any reason not to just use mp3? |
| 15:59.34 | pdtmobile | ya, converting everything ahead of time is the best bet |
| 15:59.45 | pdtmobile | DrDeke: it's very flakey |
| 15:59.54 | bigbdfs | bugz? |
| 16:00.02 | DrDeke | inasmuch as you have to use mpg123 which is a bit on the iffy side? or in terms of it actually doesn't work well? |
| 16:00.03 | jart | if you aren't going to have cpu issues, go for it |
| 16:00.12 | Ritalin- | pdtmobile: where do you normally get your chai/espresso ? |
| 16:00.25 | DrDeke | I'm not trying to give you the business; i'm just curious about it :) |
| 16:00.44 | Waverly360 | Ritalin-: I'm a bit concerned that someone with a username of Ritalin is talking so much about highly caffeinated products... ;) |
| 16:00.52 | DrDeke | haha waverly that is a good point |
| 16:00.57 | pdtmobile | DrDeke: i have had problems in the past with mpg123 threads staying on the box forever and consuming all spare cycles |
| 16:00.59 | Ritalin- | I'd hate to be the one in choosing the mp3 music for a large company. I remember when I worked at a call center no matter what music we had there was always complaints |
| 16:01.10 | DrDeke | pdtmobile: Ok, that's legitimately flaky then :). |
| 16:01.26 | jart | the last phone system i installed |
| 16:01.30 | jart | i made Malice Mizer the MOH |
| 16:01.46 | NotJohnDavid | :^D |
| 16:02.11 | *** join/#asterisk rollergrrl (n=0x3e44d@71-213-5-22.slkc.qwest.net) |
| 16:02.17 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 16:02.21 | mcf3782 | I wasn't aware there was another way to do MOH. The mpg123 way seems to be the default configuration. |
| 16:02.39 | jart | a goth band from japan consisting of a bunch of guys dressed as victorian women |
| 16:02.59 | DrDeke | mcf: I think the native MOH is new as of 1.2 |
| 16:02.59 | jart | and no one complained |
| 16:03.04 | DrDeke | mcf: not positive though |
| 16:03.15 | NotJohnDavid | I had the postal service and big audio dynamite as moh |
| 16:03.33 | pdtmobile | NotJohnDavid: postal service rocks |
| 16:03.43 | pdtmobile | how are you guys handling ASCAP fees? |
| 16:03.47 | mcf3782 | huh.. ok. I'll poke around on the net and do some reading up on that. thanks. |
| 16:04.11 | lilalinux | why doesn't this wait 3 seconds before answering?: http://rafb.net/paste/results/FJS5S449.html |
| 16:04.44 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 16:05.03 | pdtmobile | i guess that means your not ;) |
| 16:05.23 | NotJohnDavid | lilalinux: is that exactly how it's written in your conf file? |
| 16:05.26 | Waverly360 | *Knocks on the door* This is the RIAA...open the door! |
| 16:05.38 | lilalinux | notjason: yes, expect the xxx is a real number |
| 16:05.48 | lilalinux | s/pec/cep |
| 16:05.57 | Waverly360 | Put your pirated music on the floor and your hands in the air or we will shoot! |
| 16:05.59 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 16:06.13 | NotJohnDavid | lila: i'm not an expert but that should do it. did you reload ? |
| 16:06.22 | mcf3782 | I'm using royalty-free music; so the RIAA and ASCAP can kiss my @$$. :-D |
| 16:06.29 | lilalinux | NotJohnDavid: yes |
| 16:06.33 | pdtmobile | mcf3782: same here |
| 16:06.47 | NotJohnDavid | and run it with -vvv to make sure you're in the right context when it gets answered |
| 16:06.56 | Waverly360 | *taps his mic* Is this thing on? *sigh* |
| 16:07.02 | pdtmobile | Waverly360: will you be here all week and should I remember to tip my waitress? |
| 16:07.15 | Waverly360 | pdtmobile: Man...tough crowd. |
| 16:07.16 | NotJohnDavid | oh... and turn autoanswer off in the console "autoanswer off" I had that problem until i realized autoanswer was on |
| 16:08.40 | NotJohnDavid | also my understanding is that the timer isn't that accurate |
| 16:08.45 | *** join/#asterisk EinsteinTaylor (n=dtaylor@216.243.100.29) |
| 16:08.53 | *** part/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 16:08.56 | EinsteinTaylor | morning all |
| 16:09.15 | NotJohnDavid | morning einstein |
| 16:09.31 | EinsteinTaylor | are we all adequatly caffeinated? |
| 16:09.32 | CoaxD | what a bitch of a morning. All I wanna do is go back to bed. 4.5hrs of sleep last night. but had to go to work. |
| 16:09.41 | NotJohnDavid | einstein: funny you mention that |
| 16:10.04 | EinsteinTaylor | i keep hoping the red bull guys will give out samples at our office |
| 16:10.10 | mcf3782 | some days it's just not worth it to chew through the straps. |
| 16:10.46 | NotJohnDavid | I was hiking the appalachian trail about 3 years ago and when I was in one of the trail towns this HOT HOT HOT girl was driving the redbull truck around and gave me one. I think it was a sign from god |
| 16:10.48 | EinsteinTaylor | im working my way through my 2nd bag of sunflower seeds trying to find what should be simple documentation |
| 16:11.09 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 16:11.14 | EinsteinTaylor | i dont think redbull hires hot girls |
| 16:11.21 | EinsteinTaylor | i mean not hot girls |
| 16:11.23 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.232.207.Dial1.SanJose1.Level3.net) |
| 16:11.24 | EinsteinTaylor | wow...cant type |
| 16:11.25 | *** join/#asterisk bigbdfs (n=bwarner@65.113.208.35) |
| 16:11.31 | NotJohnDavid | she had pigtails too |
| 16:11.41 | EinsteinTaylor | handlebars :) |
| 16:11.50 | bigbdfs | bugz? |
| 16:12.04 | EinsteinTaylor | if thats how u get down |
| 16:12.35 | EinsteinTaylor | anyone knowledgeable with IAX trunking? |
| 16:13.14 | bigbdfs | anyone familiar with motorola vt1005 and getting it to register to asterisk? |
| 16:13.36 | CunningPike | EinsteinTaylor: I've done it (IAX trunking, that is), but that's no guarantee that I knew what I was doing - what's the question? |
| 16:13.44 | riksta | bigbdfs, please can you post to the asterisk-users mailing list, rather than repeating your question loads of times |
| 16:13.59 | EinsteinTaylor | just trying to set up a simple relay to relay incoming/outgoing sip calls to another server that has an ISDN card |
| 16:14.09 | EinsteinTaylor | so that i have my production voicemail server and a sandbox |
| 16:14.37 | CunningPike | EinsteinTaylor: Oh, that's fairly straightforward - there's a good wiki page on it, and I can send you our config if you like |
| 16:14.57 | EinsteinTaylor | ud be my hero |
| 16:15.12 | CunningPike | EinsteinTaylor: I'll pm it |
| 16:15.14 | Waverly360 | Be back later guys. |
| 16:15.25 | CunningPike | EinsteinTaylor: Hang on a sec while I dig it out |
| 16:15.34 | EinsteinTaylor | gracias |
| 16:15.36 | smackus | so if i do an svn install of another asterisk version, besided the changes within the version, will anything be affected on my asterisk server? configs will all stay the same, and everything else should stay working? |
| 16:16.22 | *** join/#asterisk Beighto (n=chatzill@64.160.113.130) |
| 16:16.22 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
| 16:16.49 | EinsteinTaylor | configs will probably be overwritten |
| 16:16.59 | EinsteinTaylor | SVN does diff comparrisons and will checkout new copies |
| 16:17.03 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 16:17.59 | umay | configs won't be overwritten unless you do "make samples" |
| 16:18.25 | EinsteinTaylor | even with SVN??? it usually just checks version numbers |
| 16:18.28 | EinsteinTaylor | id make backups |
| 16:18.29 | smackus | thats what I thought. |
| 16:18.31 | dan42 | the app_addon_sql_mysql is great except it doesnt use a persistent db connection.. is there any way to use a persistent connection for use in the dialplan? |
| 16:18.39 | dan42 | would AGI be the answer there? |
| 16:18.41 | smackus | yes... absolutely. |
| 16:20.21 | *** join/#asterisk Alric (n=nbowyer@wireless-062.1stel.com) |
| 16:21.28 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 16:21.41 | rene- | iax runs on 4569 udp right? |
| 16:21.46 | rene- | and ssh on 22 tcp? |
| 16:21.53 | rene- | iax2 i meant |
| 16:21.56 | Alric | Anyone know if the setvar stuff in sip.cfg works with sip peers via realtime? |
| 16:21.57 | EinsteinTaylor | by default..yes |
| 16:23.27 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:24.33 | lilalinux | does anyone have "receive_fax" running? |
| 16:25.21 | bigbdfs | bugz..u there? |
| 16:26.58 | *** part/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 16:29.51 | *** join/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com) |
| 16:32.57 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 16:33.04 | smackus | ok, so if i want to install asterisk 1.2.9.1 via svn, what is the correct url? "http://svn.digium.com/svn/asterisk/trunk" ?would this be the appropriate location? The first one I tried checked out the entire svn... took almost an hour to check out. |
| 16:33.07 | CunningPike | EinsteinTaylor: I haven't forgotten you - just fighting another fire - couple minutes..... |
| 16:33.13 | EinsteinTaylor | not a prob |
| 16:33.15 | EinsteinTaylor | appreciate it bro |
| 16:34.29 | file | smackus: http://svn.digium.com/svn/asterisk/tags/1.2.9.1 |
| 16:35.01 | smackus | wow... missed the boat, once again. you guys save my ass over and over... :-D thanks |
| 16:35.29 | *** join/#asterisk iDunno (i=brettp@miranda.sommitrealweird.co.uk) |
| 16:35.35 | lilalinux | why does that happen after just a few secons?: Spawn extension (capi-in, 12345, 1) exited non-zero on 'CAPI/ISDN1/12345-e' |
| 16:36.06 | lilalinux | It occurs while wait()ing for a call to be answered |
| 16:36.44 | bigbdfs | I have posted to asterisk-users mailing list...but checking to see if anyone can assist with registering a moto vt1005 |
| 16:36.49 | sevard | ~seen dlynes |
| 16:37.02 | jbot | dlynes <n=dlynes@216.251.149.66> was last seen on IRC in channel #asterisk, 25d 7h 20m 54s ago, saying: 'I only understand certain parts of it'. |
| 16:37.03 | sevard | !seen dlynes_home |
| 16:37.03 | sevard | !seen dlynes_office |
| 16:37.16 | sevard | ~seen dlynes_home |
| 16:37.17 | jbot | dlynes_home <i=1000@S0106001217014b92.vc.shawcable.net> was last seen on IRC in channel #asterisk, 1d 12h 49m 35s ago, saying: 'Brijn: nah...vancouver kingsway is Nanaimo to Fraser along Kingsway'. |
| 16:37.22 | sevard | ~seen dlynes_office |
| 16:37.23 | jbot | dlynes_office <n=dlynes@216.251.149.66> was last seen on IRC in channel #asterisk, 2d 11h 35m 39s ago, saying: 'talk tyl'. |
| 16:38.15 | dan42 | anyone know if there a way to use a database in a dialplan with a persistent db connection? seems like a lot of extra work with app_mysql.. would agi be the answer? |
| 16:38.39 | sevard | I use an perl script. |
| 16:40.10 | dan42 | sevard: using fastagi? |
| 16:40.30 | sevard | i use agi |
| 16:40.32 | sevard | and perl |
| 16:40.44 | Persilon | I'm trying to set an ivr menu... is there anyway of leaving one message on many voicemails at the same time ? |
| 16:41.23 | dan42 | maybe im not clear on agi.. i thought agi ran a script.. if youre running a script from scratch, the database connection wouldnt be persistant and i might as well use app_mysql |
| 16:43.10 | anthm | there is a socket based agi too but you would need to service the clients 1 at a time unless you use a db driver that works across fork somehow |
| 16:44.56 | dan42 | well, the concept would be to have a connection or two hanging out waiting for requests |
| 16:45.11 | CunningPike | EinsteinTaylor: Did you get my pm? |
| 16:45.19 | dan42 | all these connects and disconnects with app_mysql seems like a waste |
| 16:45.26 | CunningPike | I can pastebin if it's easire |
| 16:45.37 | CunningPike | s/easire/easier/ |
| 16:47.08 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 16:47.10 | PakiPenguin | evening |
| 16:47.18 | *** join/#asterisk pdtmobile (n=ptinsley@adsl-070-148-122-203.sip.bna.bellsouth.net) |
| 16:47.28 | PakiPenguin | has anyone used the quad gsm card from junghans? |
| 16:48.23 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 16:50.22 | bigbdfs | has anyone successfully registered a moto vt to asterisk? |
| 16:50.48 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 16:52.00 | EinsteinTaylor | sorry..had boss on phone |
| 16:52.01 | EinsteinTaylor | looking now |
| 16:52.12 | *** join/#asterisk skirmisha (n=vlado@87.126.55.7) |
| 16:52.20 | skirmisha | hello guys |
| 16:52.32 | EinsteinTaylor | Cunning:You're my hero..i'll let u know how it works shortly |
| 16:52.49 | skirmisha | does anyone know about why asterisk sends email from asterisk@domain.com |
| 16:53.01 | skirmisha | i mean asterisk@your domain |
| 16:53.12 | skirmisha | can i change this somewhere in the source |
| 16:53.13 | skirmisha | ? |
| 16:53.48 | salviadud | isn't that in voicemail.conf? |
| 16:54.02 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 16:54.04 | salviadud | i don't really remember, but it's an option |
| 16:54.21 | skirmisha | nope should be somewhere in the source |
| 16:54.46 | skirmisha | it has option there but in mailog i see that email is sent from asterisk@your domain |
| 16:54.48 | CunningPike | EinsteinTaylor: Great - let me know..... |
| 16:54.52 | skirmisha | and not what is written in the log |
| 16:54.56 | *** part/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 16:54.57 | skirmisha | in the config |
| 16:55.40 | *** join/#asterisk Zeeek (n=psychoma@pdpc/supporter/active/Zeeek) |
| 16:55.50 | CunningPike | skirmisha, salviadud: It's serveremail= in voicemail.conf |
| 16:55.53 | skirmisha | does anyone know source file of voicemail in asterisk |
| 16:55.54 | EinsteinTaylor | CunningPike: if i jus do type=friend do i need both incoming and outgoing setups? |
| 16:56.03 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 16:56.10 | salviadud | see, i told ya |
| 16:56.13 | skirmisha | CunningPike yes i tried this |
| 16:56.17 | CunningPike | EinsteinTaylor: Yes - because the authentication is different for each |
| 16:56.25 | EinsteinTaylor | ahhh...thx |
| 16:56.27 | CunningPike | skirmisha: Did you reload? |
| 16:56.30 | EinsteinTaylor | thats probably my prob |
| 16:56.35 | skirmisha | but basically email server send it as asterisk@what ever |
| 16:56.40 | skirmisha | yes i did |
| 16:56.53 | skirmisha | let me give u some logs |
| 16:56.54 | CunningPike | skirmisha: These are your voicemails? |
| 16:56.57 | *** join/#asterisk Zeeek (n=psychoma@pdpc/supporter/active/Zeeek) |
| 16:57.27 | Zeeek | nyuk, nyuk |
| 16:57.41 | skirmisha | 2006-06-26 17:50:29 1FutM5-0006Cz-Lu <= asterisk@..... |
| 16:58.04 | skirmisha | ok let me explain you again |
| 16:58.24 | skirmisha | in voicemail.conf i have serveremail set correctly |
| 16:58.30 | EinsteinTaylor | CunningPike:any significance to the context line? |
| 16:58.37 | EinsteinTaylor | or can it be arbitrary as long as it s uniform |
| 16:58.50 | skirmisha | but in mailog i see email sent as asterisk@... and not what i have set in voicemail.conf |
| 16:58.50 | CunningPike | EinsteinTaylor: Yes - it is the context in your dialplan where the calls will end up |
| 16:59.07 | EinsteinTaylor | thats what i thoguht...thanks |
| 16:59.16 | CunningPike | skirmisha: And what is the From: address in the recipient's message? |
| 16:59.57 | *** join/#asterisk tclark (n=TC@S0106000f66c5d294.gv.shawcable.net) |
| 17:00.03 | *** join/#asterisk Johannes|Blixtar (n=Johannes@soll3-63.cust.blixtvik.net) |
| 17:00.18 | skirmisha | basically i see both - the one from config file and the other is asterisk@your domain |
| 17:00.44 | skirmisha | so i thought it is somewhere in the source |
| 17:01.00 | skirmisha | i have set 2 diff asterisk servers and it's same on both of them |
| 17:01.40 | skirmisha | it gives me error if email asterisk@your domain does not exist as mail server check for sender email |
| 17:01.40 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 17:01.43 | EinsteinTaylor | CunningPike: do you use the register command in your sip.conf or just let it handle it automatically |
| 17:01.43 | CunningPike | skirmisha: You have two From: headers in your voicemail messages?? |
| 17:01.48 | skirmisha | that's how i discover it |
| 17:02.23 | skirmisha | let me show u what i get |
| 17:03.05 | CunningPike | EinsteinTaylor: Come again? |
| 17:03.10 | *** join/#asterisk dlynes_office (n=dlynes@216.251.149.66) |
| 17:03.27 | EinsteinTaylor | in sip.conf for each phone do you actually give the register => username:password@host |
| 17:03.33 | EinsteinTaylor | or does the context take care of it? |
| 17:04.11 | dlynes_office | ~seen justinu |
| 17:04.15 | jbot | justinu is currently on #asterisk (40m 48s), last said: '~docs'. |
| 17:04.31 | dlynes_office | justinu: you're in california, right? |
| 17:04.38 | *** join/#asterisk darkfiber1 (n=bwarner@65.113.208.35) |
| 17:04.58 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 17:05.00 | dlynes_office | Good morning, Flauto. |
| 17:05.04 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 17:05.07 | darkfiber1 | had anyone successfully registered a motorola vt to asterisk? |
| 17:05.25 | Zeeek | EinsteinTaylor: what was your question? |
| 17:05.26 | Flauto | hey, dlynes |
| 17:05.53 | Flauto | good morning |
| 17:05.57 | EinsteinTaylor | Zeeek: working on iax trunking..was wondering if I have to explicity tell it to register |
| 17:05.59 | Flauto | do you like soccer? |
| 17:06.10 | dlynes_office | Yeah...just haven't had the time to watch it much lately :( |
| 17:06.14 | Zeeek | if it's IAX why do you ask about sip.conf ? |
| 17:06.21 | Flauto | i just watched italy vs australia |
| 17:06.26 | EinsteinTaylor | it's an IAX trunk to relay sip calls |
| 17:06.30 | dlynes_office | Italy probably kicked ass :0 |
| 17:06.34 | EinsteinTaylor | at least i *hope* thats what i'm doin |
| 17:06.55 | Zeeek | but the register line is to register on a sip provider |
| 17:07.02 | CunningPike | EinsteinTaylor: You shouldn't need anything more than I've given you - is it not working? |
| 17:07.04 | EinsteinTaylor | ok...just checkin |
| 17:07.05 | dlynes_office | EinsteinTaylor: it transcodes from sip to iax |
| 17:07.07 | Flauto | italy 1:0 |
| 17:07.09 | Zeeek | (or iax but you mentioned sip.conf) |
| 17:07.10 | EinsteinTaylor | havent tried yet...almost done typing it |
| 17:07.15 | dlynes_office | EinsteinTaylor: sip has nothign to do with it |
| 17:07.24 | lilalinux | Any hint where this comes from?: DISCONNECT_IND ID=001 #0x0864 LEN=0014 |
| 17:07.25 | EinsteinTaylor | CunningPike: just making sure i have dialplans set up right now |
| 17:07.38 | EinsteinTaylor | dylnes_office: thanks...i think my mind was confused there |
| 17:07.43 | EinsteinTaylor | thought i had to ahve it all |
| 17:08.11 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) |
| 17:08.14 | CunningPike | EinsteinTaylor: OK - sip.conf has nothing to do with IAX trunking. Remember that asterisk actually breaks a call into two separate legs. One call gets placed to the asterisk server by the calling phone, and asterisk places another call to the callee |
| 17:08.28 | dlynes_office | EinsteinTaylor: you need sip.conf for your sip connections, and iax.conf for your iax2 connections, but the two are two different channels...nothing to do with one another; SIP is not tunnelled inside of IAX2 |
| 17:08.59 | CunningPike | EinsteinTaylor: So, you have a SIP call from a SIP phone to asterisk, and an IAX (or ZAP, or SIP, or whatever) call to the destination |
| 17:09.01 | *** join/#asterisk JakBeatZ (n=JakBeatZ@trek.tor1.ebit.ca) |
| 17:09.08 | EinsteinTaylor | ahh...thanks |
| 17:09.09 | *** join/#asterisk rwa (n=rwa@213.211.189.168) |
| 17:09.15 | dlynes_office | EinsteinTaylor: however, there is one thing that is related; you might want to make sure that sip and iax are both using the same codec, so that there's less transcoding happening |
| 17:09.16 | EinsteinTaylor | i thought it just carried it over the IAX trunk |
| 17:09.21 | CunningPike | EinsteinTaylor: And asterisk bridges them together |
| 17:09.42 | skirmisha | so what i get is |
| 17:09.48 | CunningPike | EinsteinTaylor: No - two seperate calls. Asterisk isn't a proxy - it's a really smart UA |
| 17:09.48 | EinsteinTaylor | here goes the test run |
| 17:10.07 | skirmisha | sender and return-path in email are asterisk@your domain |
| 17:10.21 | dlynes_office | lilalinux: is that issued from a pri, from an bri, ??? |
| 17:10.22 | skirmisha | from field is exactly what i've set in config file |
| 17:10.42 | JakBeatZ | Anyone here using NVFaxDetect? I'm having a problem where it won't detect a laptop fax modem, unless maybe it's my config... Am I missing the point of NVFaxDetect? |
| 17:10.52 | CunningPike | skirmisha: Ah, I see the issue now. That is inconvenient, alright. I would submit a patch, if I were you |
| 17:10.54 | pdtmobile | it's not always transcoding when it connects a sip call to an iax trunk. if the codecs are the same there is no transcoding. Just bridging between the two transports |
| 17:11.22 | CunningPike | pdtmobile: Very good. Go to the top of the class ;) |
| 17:11.27 | pdtmobile | lol |
| 17:11.30 | dlynes_office | pdtmobile: that's what i said, but it still has to transcode the two protocols, just not the codecs (i.e. it's not bridged) |
| 17:11.41 | dlynes_office | pdtmobile: bridging can only happen if it's exactly the same protocol |
| 17:12.12 | EinsteinTaylor | hmm...no go |
| 17:12.21 | skirmisha | CunningPike is it possible just to change it in the source |
| 17:12.42 | dlynes_office | bridging is more important for zap channels though....bridging allows faxing to work flawlessly |
| 17:12.45 | CunningPike | skirmisha: Yes, but why not contribute your changes back to the project, hmmmm? |
| 17:13.08 | CunningPike | EinsteinTaylor: pastebin your CLI output |
| 17:13.10 | CunningPike | ~pb |
| 17:13.14 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 17:13.56 | skirmisha | well i am newbe in asterisk and don;t know where to look at |
| 17:13.57 | pdtmobile | I guess where we differ is that I don't believe you are using the proper word. http://en.wikipedia.org/wiki/Transcode at best it's converting, not transcoding |
| 17:14.00 | CunningPike | EinsteinTaylor: And, while you're at it, the relevant part of extensions.conf and iax.conf |
| 17:14.09 | dlynes_office | ok, converting then |
| 17:14.10 | skirmisha | that's why i am here |
| 17:14.14 | EinsteinTaylor | k....give me a sec |
| 17:14.20 | dlynes_office | yeah...converting's a more correct word |
| 17:14.40 | CunningPike | skirmisha: OK - let's ask dlynes_office if he knows |
| 17:15.01 | dlynes_office | CunningPike: huh? |
| 17:15.06 | CunningPike | dlynes_office: Do you know where in app_voicemail the headers for the email get set? |
| 17:15.17 | dlynes_office | CunningPike: they don't |
| 17:15.24 | dlynes_office | CunningPike: sendmail takes care of that |
| 17:15.31 | sevard | Does anyone know if it's possible on the Sipura 2002s to not increase the length of a ring splash but the frequency of one? |
| 17:15.55 | dlynes_office | sevard: you mean change the ring tone? |
| 17:15.58 | pdtmobile | another option if you don't like how it handles the email notification is to use externnotify and call a script where you can format it however you want |
| 17:16.21 | dlynes_office | pdtmobile: why bother? voicemail.conf allows you to format it however you want |
| 17:16.30 | skirmisha | dlynes_office is this related to the user that asterisk is running with? |
| 17:16.36 | pdtmobile | i think they are saying it isn't working like it should |
| 17:16.40 | dlynes_office | skirmisha: nope |
| 17:16.44 | pdtmobile | i am offering up that instead of trying to patch the code, just use extern |
| 17:16.49 | CunningPike | dlynes_office: skirmisha is having a problem with sender verification where the From: header comes from voicemail.conf, but the sender: and return-path: use the user under which asterisk runs |
| 17:16.51 | pdtmobile | (in the short term) |
| 17:16.57 | sevard | dlynes_office: well, sort of. I have phones with voicemail lights and I want the voicemail lights to flash without an audible ring splash. So I turned the ring slash length down to 0 and I seem to be having mix results. sometimes I get a flash. I think if I could decrease the delay between splashes i could fix it |
| 17:17.06 | dlynes_office | CunningPike: ah |
| 17:17.17 | skirmisha | well not sure how they come |
| 17:17.18 | dlynes_office | CunningPike: serveremail=blahblah@domain.com |
| 17:17.29 | CunningPike | dlynes_office: Only sets from: |
| 17:17.30 | skirmisha | but i see them in email header |
| 17:17.37 | dlynes_office | CunningPike: yeah, so that's an issue with sendmail then |
| 17:17.42 | CunningPike | dlynes_office: Doesn't set sender: or return-path: |
| 17:17.56 | skirmisha | i use exim |
| 17:17.57 | dlynes_office | CunningPike: after I finish rewriting it, and add native smtp support in asterisk, that won't be an issue anymore |
| 17:18.14 | dlynes_office | skirmisha: you'll find /usr/sbin/sendmail is a symbolic link to exim |
| 17:18.14 | CunningPike | dlynes_office: That was the answer I wanted to hear ;) |
| 17:18.17 | pdtmobile | dlynes_office: thats actually why I use extern, to avoice sendmail |
| 17:18.20 | pdtmobile | avoid even |
| 17:18.35 | sevard | dlynes_office: it seems it flashes too quickly for it to ring but i do see a blip on the light, which is good. |
| 17:18.49 | dlynes_office | pdtmobile: yeah...not really an issue for me, i've got sendmail down pretty pat |
| 17:18.50 | EinsteinTaylor | CunningPike: should be posting now |
| 17:18.56 | sevard | dlynes_office: now if i could decrease the delay between splashes to say 50ms one could actually tell they have voicemail |
| 17:18.58 | skirmisha | dlynes_office i have set exim in the config file and i think it use exim not sendmail |
| 17:19.02 | EinsteinTaylor | CunningPike: the only complaint is about the context name but they match all across the board |
| 17:19.14 | CunningPike | EinsteinTaylor: I'll need the link....... :D |
| 17:19.18 | sevard | dlynes_office: do you know if asterisk has native ring splash notices other then having to rely on the telephone or ata to indicate ? |
| 17:19.23 | dlynes_office | skirmisha: also, I don't know if exim is anything like sendmail or not, but in sendmail.cf, you can specify the format of the message and that kinda thing for return paths, and what-not |
| 17:19.26 | EinsteinTaylor | CunningPike: i'll have it shortly |
| 17:19.29 | EinsteinTaylor | it's being slow |
| 17:19.37 | dlynes_office | skirmisha: but the easier fix is to use what pdtmobile was talking about |
| 17:19.40 | *** join/#asterisk Waverly360 (n=mirc@adsl-070-148-122-203.sip.bna.bellsouth.net) |
| 17:19.41 | pdtmobile | It's just one more service that I don't need to run/maintain, also lets me do more flexible things with multiple notifications etc... |
| 17:19.43 | CunningPike | EinsteinTaylor: Yes - it has been for the last while |
| 17:20.15 | tzafrir | is it debian's exim? |
| 17:20.24 | skirmisha | nope slckware exim |
| 17:20.35 | skirmisha | exim 4.30 |
| 17:20.40 | sevard | dlynes_office: answers to annnnny of those questions? :) |
| 17:21.01 | pdtmobile | dlynes_office: so you are rewriting the notify stuff to be native? |
| 17:21.18 | tzafrir | exim is basically sendmail compatible... |
| 17:21.28 | EinsteinTaylor | CunningPike: if i just put it on one of my servers and give you a link would u be terribly offended? |
| 17:21.36 | CunningPike | Nope |
| 17:21.58 | CunningPike | jbot, CunningPike is not easily offended |
| 17:22.01 | jbot | ...but cunningpike is already something else... |
| 17:22.17 | CunningPike | jbot, no CunningPike is not easily offended |
| 17:22.18 | jbot | CunningPike: okay |
| 17:22.28 | pdtmobile | dlynes_office: if so, two ideas that people ALWAYS ask for, multiple email addresses for the emails and mp3 support for the attachment |
| 17:22.28 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 17:22.32 | CunningPike | Lest there be any doubt on the matte |
| 17:22.35 | SplasPood | <PROTECTED> |
| 17:22.37 | CunningPike | s/matte/matter/ |
| 17:22.40 | SplasPood | hrm.. that goto looks fine to me... |
| 17:22.50 | EinsteinTaylor | of course right as a finish pastebin finishes |
| 17:22.57 | EinsteinTaylor | i'll put both links in your PM and you can choose |
| 17:22.59 | CunningPike | EinsteinTaylor: Sure |
| 17:23.18 | EinsteinTaylor | done |
| 17:23.56 | CunningPike | EinsteinTaylor: So, do you have a context 'internal' in your extensions.conf? |
| 17:24.01 | tzafrir | looking at my exim4 config template, I see that qualify_domain is probably what you're after |
| 17:24.08 | EinsteinTaylor | no...i have context=internal in my sip.conf |
| 17:24.13 | EinsteinTaylor | take it i blew that one? |
| 17:24.32 | CunningPike | EinsteinTaylor: Bzzzzzt. Contexts go in extensions.conf......... |
| 17:24.42 | EinsteinTaylor | stupid o'reilly book |
| 17:25.08 | pdtmobile | EinsteinTaylor: you need to tell sip.conf which context to put the phone in, in extension.conf you need to define and build that context |
| 17:25.08 | CunningPike | EinsteinTaylor: I'm not talking about context= here - I'm talking about [some-context] in extensions.conf |
| 17:25.26 | tzafrir | context= in sip.conf tells yyou where the call starts in the dialplan |
| 17:26.01 | Zeeek | and unless you send it out of the context on purpose, the extension are limited to that context |
| 17:26.04 | pdtmobile | EinsteinTaylor: think of them as buckets.... when you create the sip client you tell the server what bucket to put it in. Then you have to define what things in each bucket can do in extensions.conf |
| 17:26.36 | CunningPike | EinsteinTaylor: Try [internal] just above the exten=> line |
| 17:27.17 | EinsteinTaylor | doing that right now |
| 17:27.17 | CunningPike | Man, is pastebin ever slow |
| 17:27.18 | CunningPike | I guess it's just overloaded |
| 17:27.38 | CunningPike | Is it a single server? Or a cluster? |
| 17:27.46 | EinsteinTaylor | it's two servers |
| 17:27.58 | EinsteinTaylor | one is our production voicemail server and has an ISDN card |
| 17:28.02 | Zeeek | there are several pastebins |
| 17:28.22 | EinsteinTaylor | one is a server I'm setting up to try several diff things and just want all the incoming and outgoing to be passed through the ISDN in the voicemail server |
| 17:28.23 | CunningPike | Zeeek: But is say, pastebin.com one server, or a farm |
| 17:28.59 | EinsteinTaylor | gotta run to a job...be back shortly to pick back up on this |
| 17:29.06 | EinsteinTaylor | thanks for all the help so far guys |
| 17:29.07 | CunningPike | EinsteinTaylor: OK - later |
| 17:29.10 | dlynes_office | sevard: what's your problem, beeatch? |
| 17:29.37 | pdtmobile | so whats the proper procedure for submitting a patch to asterisk? |
| 17:29.49 | dlynes_office | pdtmobile: got your message about the suggestions |
| 17:29.53 | dlynes_office | pdtmobile: will take note of it |
| 17:29.58 | pdtmobile | coo |
| 17:30.46 | dlynes_office | pdtmobile: to make it easy to deal with though, I'll see what's involved in talking to format_mp3 |
| 17:30.58 | dlynes_office | pdtmobile: I don't want to have to write an mp3 encoder |
| 17:31.19 | pdtmobile | dlynes_office: ya, that would suck, no reason to re-invent the wheel if you could just pipe it through the existing format |
| 17:31.39 | dlynes_office | pdtmobile: if worst comes to worse, I might allow people to specify an external program to convert it to mp3 before emailing it |
| 17:31.47 | dlynes_office | s/worse/worst/ |
| 17:31.51 | pdtmobile | sox is your friend |
| 17:32.14 | dlynes_office | or allow them to specify an external script to send the email, period |
| 17:32.33 | dlynes_office | and then they can write their own script to convert the attachment, and add whatever fields they want, ... |
| 17:32.38 | pdtmobile | well notify gives you the same effect already |
| 17:32.54 | dlynes_office | what's this notify you keep talking about, anyways? |
| 17:33.11 | pdtmobile | externnotify that compliments externpass |
| 17:33.35 | dlynes_office | Is it a conf file? |
| 17:33.50 | dlynes_office | ah...nvm |
| 17:33.51 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 17:33.53 | dlynes_office | it's part of voicemail |
| 17:34.22 | nettie | hey guys anyone using *-dev in production please? |
| 17:34.28 | pdtmobile | ya, externnotify=/path/to/script |
| 17:34.35 | nettie | sip jitterbuffer is awesome! |
| 17:34.53 | nettie | I cant live without it anymore and I'm thinking of deploing it |
| 17:34.53 | dlynes_office | pdtmobile: yeah...it wasn't terribly well documented, so I never bothered with it |
| 17:34.59 | Vorondil | hi all. quick question: is there anybody around who's upgraded asterisk on a freebsd (6.0-release, in my case)machine (using ports) from 1.2.7.1 to 1.2.9.1? i'm mainly curious as to how long it should take |
| 17:35.01 | Vorondil | for planning downtime and whatnot |
| 17:35.06 | dlynes_office | and I haven't made it to that part in the code yet, or i'd have it documented already |
| 17:35.15 | pdtmobile | ya, I had to figure out how externpass worked, just see what it passed out |
| 17:35.15 | lilalinux | is there a maximum count of seconds for wait()? |
| 17:35.35 | *** join/#asterisk darkfiber1 (n=bwarner@65.113.208.35) |
| 17:35.36 | *** join/#asterisk Dark_Shneider (i=ircap8@133-164-246-201.adsl.terra.cl) |
| 17:35.40 | pdtmobile | for externpass it's context, box, password |
| 17:35.40 | dlynes_office | lilalinux: 2^31-1 |
| 17:35.54 | pdtmobile | can't remember for notify |
| 17:36.10 | dlynes_office | lilalinux: oops..you're talking in a dialplan |
| 17:36.11 | darkfiber1 | anyone use a motorola vt1005? do they work with asterisk? |
| 17:36.16 | dlynes_office | lilalinux: i was thinking the system call |
| 17:36.51 | lilalinux | dlynes_office: if i choose wait(15) it will wait, but inbetween the CAPI will disconnect after 3 seconds or so (DISCONNECT_IND) |
| 17:37.12 | dlynes_office | lilalinux: ah |
| 17:37.40 | dlynes_office | lilalinux: so why would it matter what the max is, if you can't use the max? |
| 17:38.24 | Heimidal | I'm having a bit of a problem.. whenever we dial out, if we are calling a number that picks up immediately, it just rings and rings |
| 17:38.39 | Heimidal | for example, American Express Open Customer Service |
| 17:38.39 | lilalinux | dlynes_office: I was grasp[ing at a straw :) |
| 17:38.56 | Heimidal | does anyone have any idea? |
| 17:38.59 | skirmisha | one more question - anyone who knows asterisk performance |
| 17:39.23 | skirmisha | how many concurrent calls can i handle |
| 17:39.30 | dlynes_office | lilalinux: is there any particular reason you're using capi? |
| 17:39.36 | Strom_C | skirmisha: depends on whether you're transcoding |
| 17:39.43 | skirmisha | lets say with duo Xeon 3.2 and 2 gb ram |
| 17:39.53 | skirmisha | no transcoding |
| 17:40.01 | sevard | Strom_C: do you know much about MWI on the Sipura 2002s? |
| 17:40.10 | Strom_C | skirmisha: then you will run out of bandwidth long before your system chokes |
| 17:40.20 | *** join/#asterisk Tier_1 (n=Tier@c-67-176-28-65.hsd1.co.comcast.net) |
| 17:40.33 | skirmisha | Strom_C this is what i need to hear :) |
| 17:40.36 | Strom_C | sevard: what do you want to know, exactly? |
| 17:40.39 | sevard | Strom_C: I found that my phones won't ring splash audibly at 0.05 VMWI splash length but I can't find how to decrease the delay between splashes. |
| 17:40.42 | skirmisha | i need 200-300 concurent calls |
| 17:40.53 | dlynes_office | skirmisha: you should be able to handle that, easily |
| 17:40.57 | lilalinux | dlynes_office: what else should I use? |
| 17:41.03 | Strom_C | skirmisha: if you do everything g.711, you're fine |
| 17:41.07 | skirmisha | great, thanks guys |
| 17:41.08 | lilalinux | I have an AVM Fritz PCI |
| 17:41.09 | sevard | In short i'm ghetto hacking the phones to light up the MWI light without ring splashing and bothering me. |
| 17:41.14 | dlynes_office | lilalinux: there's actually two methods of doing isdn in asterisk |
| 17:41.22 | Strom_C | sevard: lemme look at my 2002 |
| 17:41.23 | Strom_C | one sec |
| 17:41.25 | lilalinux | dlynes_office: con both do fax? |
| 17:41.29 | dlynes_office | lilalinux: capi's one of them, and apparently the less reliable of the two |
| 17:41.33 | dlynes_office | lilalinux: not sure |
| 17:41.33 | sevard | Strom_C: cool |
| 17:41.36 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 17:41.44 | CunningPike | nettie: Only the insane or the financially independent |
| 17:42.04 | nettie | CunningPike: eheheh |
| 17:42.11 | nettie | CunningPike: is it SOOOOOOOOOOO BAD? |
| 17:43.03 | dlynes_office | nettie: why not apply the sip jitter buffer patches against 1.2.9.1? |
| 17:43.03 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 17:43.09 | CunningPike | nettie: It's not bad, it's just not guaranteed to always work properly, if at all. It's a nightly, unstable build. tell your boss that his phone system may not work properly, if at all, and see what he says |
| 17:43.29 | dlynes_office | CunningPike: he'll say "Go for it!!!" |
| 17:43.31 | darkfiber1 | has anyone registered a motorola vt1005? |
| 17:43.41 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 17:43.53 | CunningPike | nettie: The whole point of trunk is to work out enough bugs so that we can tell our bosses that stable is actually stable |
| 17:43.56 | Strom_C | sevard: looks like my desktop is going bonkers |
| 17:43.57 | Strom_C | brb |
| 17:44.12 | dlynes_office | nettie: nah..it's to fake the boss out |
| 17:44.14 | dlynes_office | erm |
| 17:44.19 | nettie | eheheh |
| 17:44.22 | dlynes_office | CunningPike: nah...it's to fake the boss out |
| 17:44.34 | dlynes_office | CunningPike: and then release a 1.2.9.1 afterwards |
| 17:44.38 | CunningPike | nettie: This is a key advantage of OSS |
| 17:44.42 | CunningPike | dlynes_office: Heh |
| 17:45.32 | nettie | dlynes_office I would like to (use the patch), I had one for 1.2.6 but I wasnt able to create a working binary |
| 17:45.42 | dlynes_office | nettie: do a manual patch |
| 17:45.50 | dlynes_office | don't use the patch tool |
| 17:45.54 | Heimidal | so I take it no one has seen that issue? :P |
| 17:46.04 | dlynes_office | Heimidal: apparently not |
| 17:46.09 | nettie | dlynes_office if I check the logs the options in sip.conf are still being rejected |
| 17:46.15 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 17:46.30 | nettie | dlynes_office maybe I need a fresh patch .. where u got your from? |
| 17:46.31 | dlynes_office | nettie: don't know what to tell you |
| 17:46.48 | dlynes_office | nettie: but i do know a lot of the guys in #asterisk-dev are using it |
| 17:46.56 | dlynes_office | nettie: i didn't...i'm not using it |
| 17:47.02 | nettie | dlynes_office ah ok |
| 17:47.16 | dlynes_office | nettie: i'm like CunningPike |
| 17:47.19 | nettie | dlynes_office do you suggest me to ask in *-dev ? |
| 17:47.21 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:47.29 | dlynes_office | nettie: I only use the release code, unless i absolutely have to use a trunk release |
| 17:47.31 | Heimidal | dlynes_office: hmm.. well crap. |
| 17:47.46 | dlynes_office | Heimidal: but, there's nothing to stop you from asking later |
| 17:47.54 | CunningPike | nettie: dlynes_office wishes he was like me....... |
| 17:47.56 | Heimidal | of course :) |
| 17:48.03 | dlynes_office | Heimidal: this is an international channel....everyone's not in the same time zone |
| 17:48.19 | dlynes_office | CunningPike: not especially :) |
| 17:48.22 | CunningPike | :P |
| 17:48.25 | dlynes_office | CunningPike: but you can keep dreaming :) |
| 17:48.26 | Heimidal | right |
| 17:48.48 | JakBeatZ | Is there a trick to getting NVFaxDetect to work? I have it setup and am receiving calls over an IAX channel from my IAX/PSTN provider. If I hit the NVFaxDetect DID with a PC fax modem, the fax tones aren't detected (even though they are heard through the pc speaker) so it's not sent to the fax extension. I've tried with another email to fax service I have as well, but neither of them have worked. I've even set a rediculously long timeout in |
| 17:48.49 | JakBeatZ | NVFaxDetect (10 seconds) to see if it missed the times the first time around. Anyone have any ideas? |
| 17:48.49 | nettie | dlynes_office: I spent ages trying to fine tune qos on the cisco router but I still was not able to achieve great results. |
| 17:49.36 | Strom_C | sevard: still there? |
| 17:49.42 | sevard | of course |
| 17:49.47 | Strom_C | alright, lemme look now :) |
| 17:50.02 | dlynes_office | nettie: and? |
| 17:50.09 | pdtmobile | JakBeatZ: fax over sip/iax is very unstable and unpredictable |
| 17:50.18 | pdtmobile | you need a real pstn connection for it to work properly |
| 17:50.59 | pdtmobile | JakBeatZ: good reading on the subject: http://www.soft-switch.org/foip.html |
| 17:51.01 | sevard | I use fax over sip to my asterisk box using pri and it works well. |
| 17:51.21 | dlynes_office | sevard: on the same lan? |
| 17:51.25 | JakBeatZ | Hmm, but I can use rxfax directly and it seems to work fine.. so it's not so much fax but fax detection that's problematic? |
| 17:51.28 | JakBeatZ | thx for the link |
| 17:51.54 | nettie | dlynes_office and .. if the link is lagged you absolutely need jitterbuffer |
| 17:51.54 | sevard | dlynes_office: sort of, i do have 11ms to the server from the phone. |
| 17:52.28 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 17:52.33 | pdtmobile | if you have little to no lag and no packet loss you can make it work |
| 17:52.46 | sevard | that's the key |
| 17:52.49 | pdtmobile | but across the internet you don't have a good shot at it |
| 17:52.52 | sevard | packet loss and your fax is fuxed |
| 17:52.54 | nettie | dlynes_office no qos will help you if you're trying to make 2 or more g711 calls on a 1280/256kb DSL when you're uplaoding email and leeching isos. |
| 17:53.08 | nettie | aehm |
| 17:53.15 | nettie | sending email with large attachments |
| 17:53.18 | nettie | I mean |
| 17:53.23 | sevard | quote unquote |
| 17:53.27 | pdtmobile | so it's the iso's... CRAP :) |
| 17:53.31 | Strom_C | sevard: what about just setting VMWI Ring Splash Len to 0? |
| 17:53.39 | nettie | eheh iso's like centos ISO |
| 17:53.42 | nettie | debian ISO |
| 17:53.52 | nettie | not like xbox360 iso |
| 17:53.54 | nettie | eheheh |
| 17:54.01 | pdtmobile | so i'm ok if I download the warez is what you are saying ? |
| 17:54.02 | dlynes_office | nettie: ok...and? |
| 17:54.11 | Strom_C | ow |
| 17:54.13 | Strom_C | that hurt |
| 17:54.14 | nettie | dlynes_office and I bladly need jitterbuffer :) |
| 17:54.16 | dlynes_office | nettie: i fail to understnad why you're telling me all this |
| 17:54.20 | dlynes_office | nettie: ah |
| 17:54.28 | dlynes_office | nettie: won't qos fix all that? |
| 17:54.32 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:54.33 | nettie | dlynes_office noway |
| 17:54.48 | dlynes_office | nettie: why not? |
| 17:54.52 | sevard | that's what I've done, I found that if I increased the length to .5 it still splashes without an audible ring but if i go to 0.06 i can hear it. the problem is that the splashes are spaced so far apart and go so quickly that nobody will notice it. but if i could decrease the delay between splashes... |
| 17:55.02 | nettie | dlynes_office pipe is too small |
| 17:55.12 | sevard | nettie: personal problem? |
| 17:55.17 | Strom_C | sevard: that's why you install a phone with a bloody MWI light on it |
| 17:55.24 | clyrrad | Trying to setup a shared PBX for 2 different companies. I have each company having their own config files by the use of INCLUDES. I am trying to make it so that 1 company can not call the other compaies extensions. Is there a way to do this? Right now they are able to call each others extensions. In the sip.conf file - I would like to beable to seperate the INCLUDES into context's to keep them seperate. Wonderi |
| 17:55.27 | sevard | Strom_C: it has a bloody MWI light on it |
| 17:55.27 | dlynes_office | nettie: you can't set voip at a certain %'age of the bandwidth? |
| 17:55.29 | nettie | dlynes_office the latency goes up and jitter increases |
| 17:56.08 | sevard | Strom_C: these were taken out of a hotel whose PBX did it flawlessly before until a lightning strike or whatever |
| 17:56.09 | dlynes_office | nettie: i've always assumed cisco had more flexibility for that kind of stuff than iptables |
| 17:56.33 | SplasPood | Is there any way with "Privacy" mode in Dial() under 1.2.X to remove the 'Torture/Polite Goodbye' options? |
| 17:56.33 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 17:56.39 | dlynes_office | nettie: and i'm pretty sure you can control all that kinda stuff with iptables |
| 17:56.40 | nettie | dlynes_office problem is not voip, sure I can do it, it's weightning algo .. the latency goes up and the jitter kill you |
| 17:56.41 | Strom_C | sevard: really, the two indicators are the light and the stutter dial tone. The splash ring is only to alert the phone that the FSK signal for "turn on MWI" is about to happen |
| 17:56.48 | Strom_C | sevard: oh, these are analog hotel phones? |
| 17:56.57 | nettie | dlynes_office I can do the same I do with cisco using tc and fwmark on linux |
| 17:57.20 | Strom_C | sevard: then you're screwed, because those phones don't do MWI with FSK signaling; they rely on 90vDC to be placed across the line (IIRC) to activate the light |
| 17:57.22 | sevard | Strom_C: yeah, analog hotel phones |
| 17:57.24 | NotJohnDavid | of everything i've read on asterisk so far i haven't seen an explination of the s extension |
| 17:57.26 | dlynes_office | nettie: but you can't tell iso traffic to calm down, and take a pill? |
| 17:57.35 | clyrrad | anyone know if there is a way for me to seperate the logic of the dial plans for the 2 seperate companies? |
| 17:57.39 | SplasPood | clyrrad: you'd put each companies' sip users in a different context |
| 17:57.42 | nettie | anyway.. jitterbuffer on *-dev solved all the problem |
| 17:57.57 | nettie | sure, it does it automatically |
| 17:58.04 | sevard | Strom_C: hmm, would a T1 with a channel bank or a 2400P give me the flexibilty to control how much voltage a MWI takes? |
| 17:58.15 | pdtmobile | NotJohnDavid: you use it mainly in relation to analog zap interfaces, when a call comes in it hits the s extension |
| 17:58.21 | dlynes_office | nettie: so i would think latency wouldn't be an issue if you can control both sides of the equation |
| 17:58.26 | Strom_C | sevard: the answer is "don't use phones that rely on ancient MWI technology" |
| 17:58.32 | dlynes_office | nettie: just put a bandwidth cap on everything that's not voip |
| 17:58.37 | nettie | sure, you give an high portion of ur upstream to voip traffic and configure the rest as best effort |
| 17:58.38 | pdtmobile | so when you assign a zap interface to a context, you put exten => s,1,blah and it will start there |
| 17:58.50 | clyrrad | SplasPood - I have that I have a compa_sip.inc and compb_sip.inc - but they are both INCLUDED in sip.conf - thus it uses the first companies dial plan - do you know what I mean? |
| 17:58.58 | NotJohnDavid | pdtmobile: so how does it differ than _X.,1,blah ? |
| 17:59.01 | Strom_C | sevard: so replace the phones with Aastra 9316CW sets and you're set |
| 17:59.07 | sevard | Strom_C: what if the ability to sell a pbx to a corporation relies on the ability to interoperate with their legancy technology |
| 17:59.38 | pdtmobile | NotJohnDavid: never tried it that way, I guess that would work. |
| 17:59.44 | pdtmobile | not sure though |
| 17:59.47 | nettie | dlynes_office I spent a quite large amount of time on it, I opened a case and I had 3 diff engineers lookng into it they didnt solve it |
| 17:59.48 | Strom_C | sevard: well, hey, if you can figure out a way to do non-FSK MWI lamps, let me know |
| 17:59.57 | dlynes_office | nettie: damn |
| 17:59.59 | sevard | :| |
| 18:00.07 | SplasPood | clyrrad: huh? I mean extensions.conf contexts |
| 18:00.20 | nettie | dlynes_office and they told me that's normal because jitterbuffer is a must |
| 18:00.20 | dlynes_office | nettie: fire them! |
| 18:00.30 | nettie | dlynes_office I cant fire cisco staff |
| 18:00.34 | dlynes_office | heheh |
| 18:00.58 | dlynes_office | but did you try doing a manual graft of jitterbuffer onto 1.2.9.1? |
| 18:00.59 | clyrrad | SplasPood - Likewise I haver a compa/extensions.inc and a compb/extensions.inc so that they can have their own dial plans and contexts |
| 18:01.06 | CunningPike | NotJohnDavid: We mainly use the 's' extension for macros - when a call enters a macro, there is no extension, per se - 's' is a placeholder extension and the real extension value is in ${MACRO_EXTEN} |
| 18:01.07 | pdtmobile | at least it's not as bad as the mainframe guys at my old job |
| 18:01.19 | nettie | dlynes_office I want to check that patch out |
| 18:01.21 | pdtmobile | they were convinced the problem with one of the servers was ion buildup under the raised floor |
| 18:01.29 | CunningPike | NotJohnDavid: That way, you don't have to do pattern matching in your macros |
| 18:01.29 | nettie | dlynes_office I'm not sure mine is the same |
| 18:01.33 | NotJohnDavid | cunningpike: do you know what the s is short for? string? |
| 18:01.45 | dlynes_office | nettie: you might have downloaded an earlier version of the patch, too |
| 18:01.47 | nettie | dlynes_office I'll check in asterisk-dev and see if they can send me a copy |
| 18:01.49 | pdtmobile | i would assume start |
| 18:01.50 | CunningPike | NotJohnDavid: something |
| 18:01.52 | nettie | dlynes_office could be |
| 18:01.55 | clyrrad | SplasPood - the problem in in the sip.conf file where all teh includes start from - I need away to seperate so that compa only used compa's INCLUDES and compb uses only compb's INCLUDES... Does that make more sense now? |
| 18:01.58 | *** join/#asterisk dec_ (n=tom@ppp164-96.lns3.adl4.internode.on.net) |
| 18:02.03 | nettie | dlynes_office you actually know where it is? |
| 18:02.13 | NotJohnDavid | cunningpike: something would mean 1 or more digits ? |
| 18:02.15 | nettie | dlynes_office where I could download it from? |
| 18:02.23 | dlynes_office | nettie: what I would do, is do a diff between 1.2.6 and the working copy for the sip jitter buffer |
| 18:02.24 | SplasPood | clyrrad: for each sip peer in sip.conf you'd have context= and point company A to 1 context and company B to another |
| 18:02.33 | clyrrad | Yes I have that |
| 18:02.45 | nettie | dlynes_office I'll check in *-dev |
| 18:02.50 | dlynes_office | nettie: the working copy should be down inside of oej's tree |
| 18:02.59 | clyrrad | but sip.conf INCLUDEDS each compaines sip.inc |
| 18:03.07 | clyrrad | and thats where the problem is |
| 18:03.09 | NotJohnDavid | sounds like the two would be functionally the same but asterisk probably expects s in some cases... and it's prettier. ok thanks guys :) |
| 18:03.32 | CunningPike | NotJohnDavid: It really means nothing - it's a special case for extensions where the actual extension value has no meaning, like in a macro. When you pass control to a macro, a macro doesn't have the same concept of pattern matching as an extension does |
| 18:03.35 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 18:03.43 | clyrrad | I need to INCLUDE compa's sip.inc ONLY for compa and INCLUDE compb's sip.inc ONLY for compb |
| 18:03.51 | Dr-Linux | question, why i'm getting this? > chan_sip.c:6275 check_auth: stale nonce received from '4092 <sip:4092@70.89.66.123>' |
| 18:04.59 | NotJohnDavid | cunningpike: ahh, okay that's even more clear then. I'm not sure why this hasn't been explained in anything i've read ... probably not been reading the right things but i've read a lot it seems. they just start out s,1,blah with no explination |
| 18:06.07 | CunningPike | NotJohnDavid: I think I found something in the wiki that explained it quite well...... |
| 18:06.07 | Zeeek | NotJohnDavid: s is start or standard |
| 18:06.11 | Zeeek | it is obligatory in macrosd |
| 18:06.16 | *** join/#asterisk tim[uks] (i=wibble@vortex.ukshells.co.uk) |
| 18:06.48 | Zeeek | you also need an 's' for incoming fxs |
| 18:06.56 | NotJohnDavid | zeeek: i just remember when i was playing around i saw something like <yadda yadda i forget> not found, falling back to s |
| 18:07.00 | CunningPike | NotJohnDavid: But I can't remember where - maybe it was under App Macro |
| 18:07.03 | clyrrad | SplasPood - the other reason I need to beable to seperate the INCLUDES is so that I dont get the 'context name' is already in use error - becease both companies have the same context names inside thier respective folders |
| 18:07.05 | *** join/#asterisk tamp4x (n=tampon@64.201.13.51) |
| 18:07.21 | pdtmobile | NotJohnDavid: just looked in the code, it falls back to s if it can't find the current extension the channel is set to |
| 18:07.28 | Zeeek | yeah if you screw up the extension and it doesn't find a valid one, it will try for an 's' - great fallback feature :) |
| 18:07.33 | NotJohnDavid | cunning: okay i'll just check all the wikis out. was it on wiki.org under asterisk ? or voip-info/wiki ? |
| 18:07.45 | Zeeek | look at http://asterisk-docs.org |
| 18:07.50 | tamp4x | if i have dialtone and all, i can see when someone clicks on 110 block, but the person cannot make outbound calls, what could be the issue? |
| 18:07.56 | tim[uks] | has anyone seen a problem with asterisk where the dtmf tones are occasionally not recognised if you dial too quickly or leave a long gap between the numbers? |
| 18:07.57 | Zeeek | look up dialplan - it's pretty well explained there |
| 18:08.16 | NotJohnDavid | zeeek: awesome. thanks |
| 18:08.19 | CunningPike | NotJohnDavid: voip-info.org |
| 18:08.25 | tamp4x | see peopel go off hook / on hook |
| 18:08.26 | fholmes | asterisk-docs.org is not working on my end. |
| 18:08.35 | pdtmobile | also looks like g is a special one it tries |
| 18:08.40 | CunningPike | fholmes: Try your other end :D |
| 18:08.41 | Zeeek | CunningPike: http://asterisk-docs.org |
| 18:08.47 | CunningPike | Sorry, couldn't resist |
| 18:08.57 | NotJohnDavid | http://www.asteriskdocs.org works ? |
| 18:09.10 | Zeeek | erm, possibly - but that's the idea |
| 18:09.12 | fholmes | NotJohnDavid: It is without the - I believe also. |
| 18:09.22 | Zeeek | could well be |
| 18:09.38 | CunningPike | Why does DNS have to be so goddam precise? :D |
| 18:09.53 | Zeeek | and for your further convenience: http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:09.54 | fholmes | Zeeek: It is and thanks also. I just happened to look at the screen at just the right time. :-) |
| 18:10.24 | NotJohnDavid | cunning: maybe it stands for don't know sh*t instead of domain name server :-p |
| 18:10.26 | Bullseye_Network | I know in extensions.conf I can route by callerid like exten => 4801112222/6022223333,1,Whatever is there a way to do something like 4801112222/60222233XX where is will block that number with anyhing as the last 2 digits? |
| 18:10.36 | Zeeek | Some people say bad things about the "old book". Yet it answers 90% of the dialplan questions asked here day after day |
| 18:10.47 | tzanger | Bullseye_Network: do it in the dialplan |
| 18:10.48 | tzanger | not in the extensions |
| 18:10.55 | CunningPike | NotJohnDavid: lol |
| 18:11.11 | SplasPood | clyrrad: well don't use the same context names.. |
| 18:11.13 | pdtmobile | here we go: |
| 18:11.15 | pdtmobile | <PROTECTED> |
| 18:11.15 | pdtmobile | <PROTECTED> |
| 18:11.26 | Bullseye_Network | extensions.conf is the dialplan |
| 18:11.29 | Zeeek | instead of the stupid ass biz and info TLD, they should have had .doc |
| 18:11.30 | SplasPood | clyrrad: contexts are used to separate dialplan logic |
| 18:11.32 | NotJohnDavid | zeeek: i've always been a fan of researching and reading instead of asking questions. you know. rtfm and all |
| 18:11.41 | tzanger | i.e. exten => 4801112222,1,GotoIf($[${CALLERID(number):-2} = 60222233]?yup:nope) |
| 18:11.50 | pdtmobile | you also have t,h, and i (invalid) |
| 18:12.00 | clyrrad | SplashPood - Yea - thats why I was wondering if there was a way to seperate the INCLUDES |
| 18:12.03 | Zeeek | NotJohnDavid: it's certainly worth checking that stuff out, including the wiki |
| 18:12.04 | NotJohnDavid | pdtmobile: okay that's exactly what I need to be reading, where is that? |
| 18:12.12 | pdtmobile | hehe |
| 18:12.20 | pdtmobile | include/asterisk/app.h |
| 18:12.28 | *** join/#asterisk Nodren (n=nodren@adsl-75-8-201-246.dsl.frs2ca.sbcglobal.net) |
| 18:12.36 | SplasPood | clyrrad: yea, run two copies of asterisk on totally different IPs/ports with totally different config files :P But I'm gonna assume you don't wanna do that |
| 18:12.40 | *** part/#asterisk Nodren (n=nodren@adsl-75-8-201-246.dsl.frs2ca.sbcglobal.net) |
| 18:12.41 | Bullseye_Network | tzanger: yes I thought about that, just though It would be eaiser with xx. :) |
| 18:12.46 | Bullseye_Network | thanks |
| 18:13.03 | clyrrad | LOL - UM Yea... Dont wanna do that |
| 18:13.21 | NotJohnDavid | pdtmobile: that's a bit obscure for such important information :-p |
| 18:13.31 | SplasPood | clyrrad: for example.. I prefix all my contexts with C1000 where 1000 is that customer's id.. so I have C1000-outbound-default, and C1001-outbound-default, etc.. |
| 18:14.05 | clyrrad | SplasPood - Yes that makes sense - I guess that is the route I will have to take |
| 18:14.17 | clyrrad | too bad there is not a way to seperate include files right from sip.conf |
| 18:14.32 | clyrrad | then you could just put each customers configs in their own directory |
| 18:14.47 | *** join/#asterisk RF_MIA (n=mw1@ip67-93-229-222.z229-93-67.customer.algx.net) |
| 18:14.53 | SplasPood | clyrrad: thats the exact opposite of the design |
| 18:15.05 | clyrrad | Yea.... |
| 18:15.08 | pdtmobile | NotJohnDavid: I am sure that is documented somewhere else but I usually go to the horses mouth, no better resource (if you can read/code c) |
| 18:15.27 | CunningPike | pdtmobile: Big if, for me........ :) |
| 18:15.44 | dlynes_office | man |
| 18:15.49 | dlynes_office | too damned hot outside |
| 18:15.50 | pdtmobile | i'm not a c coder, I just play one on tv |
| 18:15.52 | dlynes_office | need aircon |
| 18:15.53 | NotJohnDavid | pdtmobile: funny how I didn't think of that. but that's a great idea. i'll check out the wikis and such then when i get a chance the sources |
| 18:16.03 | CunningPike | dlynes_office: Aircon outside? |
| 18:16.05 | NotJohnDavid | pdtmobile: did you sleep in a holiday inn express last night? |
| 18:16.14 | pdtmobile | YES |
| 18:16.34 | Zeeek | it's very, very old, but here is a heavily commented dialplan: http://www.loligo.com/asterisk/example-configs.2003-02-16/extensions.conf |
| 18:16.36 | dlynes_office | CunningPike: no...my office is like a freaking sauna |
| 18:16.37 | Waverly360 | NotJohnDavid: Hey...that was my joke earlier... :P |
| 18:16.41 | Zeeek | might be worth contemplating |
| 18:16.47 | dlynes_office | CunningPike: i wish i was outside |
| 18:17.00 | dlynes_office | CunningPike: then i'd be driving around with the moonroof open, or playing some tennis or something |
| 18:17.08 | pdtmobile | It's nice and cool here at Moe's Southwest Grill |
| 18:17.12 | mut | w00t! i'm closing on my house friday! |
| 18:17.30 | Bullseye_Network | tzanger: is there anyother way, Im may have multiple callerid ranges to route. and I dont want to have that many goto's, can it work like exten => 4801112222/${somestring:-2}, |
| 18:17.36 | Zeeek | and for those who know absolutely nothing about any of this but are dying to play with asterisk: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 18:17.43 | NotJohnDavid | waverly: sorry. missed it. i just remember one of your jokes from this morning bombed pretty hard :-p |
| 18:17.45 | Zeeek | Great beginner artiçcle ^^^^^^ |
| 18:17.57 | dlynes_office | actually today would be a bonus day for golf :) |
| 18:17.57 | sevard | Strom_C: I'm still researching this problem, I want to thank you though, for your help. |
| 18:17.57 | pdtmobile | NotJohnDavid: lol, ya, he has been on a "roll" |
| 18:18.09 | Waverly360 | Shaddup all of you :P |
| 18:18.17 | Strom_C | sevard: very welcome...I am actually curious whether you can do MWI with the old voltage-based system |
| 18:18.18 | Waverly360 | It was a rough weekend... :P |
| 18:18.40 | sevard | Strom_C: if I find any positive results you'll be the first to know. |
| 18:18.48 | tzanger | Bullseye_Network: how in the world is what you wrote *any* simpler than the GotoIf? |
| 18:18.53 | Strom_C | fantastic |
| 18:19.38 | *** join/#asterisk PepseLap (n=pepse@71-209-152-225.phnx.qwest.net) |
| 18:19.51 | drray | Strom_C - channel bank from rhino supports the 90v |
| 18:19.53 | PepseLap | morning |
| 18:20.00 | *** join/#asterisk rjt69 (n=rjt@wsip-68-15-228-81.om.om.cox.net) |
| 18:20.04 | *** join/#asterisk mrtwister (n=ambervoi@107.250.broadband5.iol.cz) |
| 18:20.15 | pdtmobile | drray: you like the rhino, i am about to pick one up for a project |
| 18:20.17 | Bullseye_Network | tzanger: Will have gotoif going to another gotoif going to another gotoif or can can I have ,GotoIf($[${CALLERID(number):-2} = 60242424 or 48033322 or 48077722 or .....] |
| 18:20.40 | drray | pdtmobiel - I don't have one yet, I am pondering putting one in my operating budget though |
| 18:20.43 | pdtmobile | ah |
| 18:20.50 | pdtmobile | anybody used the rhino and like it? |
| 18:22.21 | tzanger | Bullseye_Network: you can use a regex too if you like |
| 18:23.57 | Strom_C | drray: interesting; can asterisk tell the rhino to activate the 90v MWI? |
| 18:24.39 | sevard | OH GOD |
| 18:24.44 | sevard | HOT TEA |
| 18:24.46 | sevard | FUCKING SHIT |
| 18:25.03 | sevard | ahhh my freaking lips and my crotch |
| 18:25.09 | dlynes_office | sevard: you're supposed to drink it, not wear it |
| 18:25.22 | sevard | apparently i have a hole in my lip |
| 18:25.35 | tzanger | and you're not supposed to use THAT to drink it |
| 18:25.37 | CunningPike | sevard: How did you get your lips that close to your crotch anyway? |
| 18:25.41 | tzanger | what do you think you are, a fucking elephant? |
| 18:25.42 | sevard | and apparently the woman down the street running the tea shop decided to make it 4000000 degrees freakinc C |
| 18:25.42 | pdtmobile | sevard: lol |
| 18:25.43 | drray | Strom_C - I've not used it yet, but I assume that asterisk passes regular MWI to the channel bank, and then the rhino says this port uses 90v |
| 18:25.46 | dlynes_office | sevard: that'll teach you to get into body piercing |
| 18:25.58 | tzanger | drray: precisely |
| 18:26.02 | sevard | dlynes_office: no pircings yet and probably ever |
| 18:26.04 | dlynes_office | CunningPike: no doubt |
| 18:26.08 | tzanger | stutter dialtone && FSK IIRC |
| 18:26.19 | sevard | god i haven't been burned that bad since dlynes mom dripped hot wax on my naked testicles |
| 18:26.27 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-125-116.red.bezeqint.net) |
| 18:26.28 | drray | but we are a hotel, so we need 90v or 200 new phones |
| 18:26.31 | dlynes_office | sevard: you never did answer cunningpike |
| 18:26.39 | tzanger | drray: then get a channel bank that does that |
| 18:26.43 | sevard | i missed the question |
| 18:26.50 | sevard | oh |
| 18:27.00 | CunningPike | sevard == marilyn manson |
| 18:27.01 | sevard | my lips aren't near my crotch, the liquid fell from my mouth to the lap |
| 18:27.05 | RF_MIA | drray: that is correct, you will get MWI via the channel banks |
| 18:27.08 | sevard | bottom two ribs removed and everything |
| 18:27.12 | pdtmobile | you just never know what you are going to read on here, it's great |
| 18:27.16 | CunningPike | lmao |
| 18:27.33 | *** part/#asterisk hensema (n=erik@scrat.hensema.net) |
| 18:27.34 | pdtmobile | so... we were talking about testicle wax, it's going to be in 1.4 |
| 18:27.35 | tzanger | who's jason sjobeck? |
| 18:27.38 | CunningPike | "my lips aren't near my crotch, the liquid fell from my mouth to the lap" - now there's a quote just begging to be taken out of context |
| 18:27.47 | tzanger | he's something to do with asterisk :-) |
| 18:27.51 | dlynes_office | CunningPike: heh |
| 18:27.53 | sevard | CunningPike: excellent |
| 18:27.55 | tzanger | CunningPike: hahaha |
| 18:28.01 | CunningPike | rofl |
| 18:28.08 | PepseLap | hey, what's my exten => _9 line supposed to look like in extensions.conf? I've got a digium with 2 FXOs and 1 FXS |
| 18:28.18 | tzanger | PepseLap: huh? |
| 18:28.18 | PepseLap | only one of the FXOs are connected |
| 18:28.25 | PepseLap | the line to make it dial out on 9 |
| 18:28.40 | dlynes_office | PepseLap: check out the DISA dial plan application on the wiki |
| 18:28.43 | pdtmobile | i am sure my co-workers think I am absolutely bonkers laughing into my laptop |
| 18:28.43 | dlynes_office | ~disa |
| 18:29.11 | jbot | [disa] direct inward system access. show application disa |
| 18:29.13 | PMantis | <PROTECTED> |
| 18:29.13 | salviadud | how about we get some money rounded up, and we tell Zimmerman to encrypt IAX2? |
| 18:29.13 | PMantis | That's the simplest form. |
| 18:29.13 | tzanger | PMantis: well maybe ${EXTEN:1} |
| 18:29.13 | salviadud | i'd put in 50 bucks myself |
| 18:29.13 | PMantis | Oh, LOL |
| 18:29.13 | tzanger | res_gpg |
| 18:29.22 | dlynes_office | holy, jbot is slow today |
| 18:29.27 | sevard | god my lips |
| 18:29.47 | PMantis | salviadud, Yeah |
| 18:29.48 | CunningPike | sevard: Stop it! I had just regained my composure |
| 18:30.00 | PMantis | <PROTECTED> |
| 18:30.02 | *** part/#asterisk Zeeek (n=psychoma@pdpc/supporter/active/Zeeek) |
| 18:30.05 | sevard | they're fucking swelling up like your sister's vag |
| 18:30.12 | pdtmobile | the slap actually helps, clears that congestion right up... or at least makes you feel better |
| 18:30.33 | CunningPike | PepseLap: Are you trying to allow people to simply dial out? Or to be able to dial in and then back out again? |
| 18:30.35 | salviadud | you guys are nutz |
| 18:30.42 | salviadud | cya later dudes |
| 18:30.45 | CunningPike | yes, yes, we are |
| 18:30.48 | PepseLap | cunnin: just dial out |
| 18:30.49 | dlynes_office | lol |
| 18:31.05 | pdtmobile | did we just run him off? |
| 18:31.05 | CunningPike | PepseLap: Well, you don't really need a 9 - do you? |
| 18:31.08 | PepseLap | i'd like to also set it up to answer the phone and give a menu, but i'm sure i can search for something |
| 18:31.20 | PepseLap | cunningpike: not really |
| 18:31.47 | CunningPike | PepseLap: We got rid of our '9' - although our dialplan does both so people don't get confused. |
| 18:32.14 | PepseLap | what did you do instead? |
| 18:32.15 | CunningPike | PepseLap: All we have is exten => _9X.,1,Goto(${CONTEXT},${EXTEN:1},1) |
| 18:32.34 | tzanger | yeah I made our 10-digit change so that if you dial 7 it prepends the area code, if you leave off the 1, it adds it and if you add it it strips it if not needed |
| 18:32.54 | CunningPike | PepseLap: Direct dial - exten => _604XXXXXXX,1 etc |
| 18:33.03 | *** join/#asterisk mindwarp (i=mindwarp@silenceisdefeat.org) |
| 18:33.43 | CunningPike | PepseLap: We just make sure that we don't allocate conflicting locals |
| 18:34.01 | CunningPike | PepseLap: So we will never have locals that start 604X |
| 18:34.12 | CunningPike | PepseLap: Or 778X |
| 18:34.21 | PepseLap | hm. yeah i'll need to go through and change all these default extensions :) |
| 18:34.45 | PepseLap | i installed some distro to "save time" |
| 18:34.54 | CunningPike | PepseLap: It rarely does ;) |
| 18:36.20 | *** join/#asterisk backblue (n=moo@87-196-10-181.net.novis.pt) |
| 18:41.09 | sevard | Strom_C: are you sure that phones use FSK for MWI? |
| 18:41.29 | Strom_C | sevard: modern phones definitely do |
| 18:42.37 | *** join/#asterisk RaYmAn-Bx (i=rayman@kbhn-vbrg-sr0-vl201-020.perspektivbredband.net) |
| 18:43.04 | sevard | Strom_C: alright, I just called Digium and he said he was poking around in source looking for how they do it.. he says they use FSK for CID but "probably not MWI" he took my email and said he'd push it up the chain. unsure on a response time from digium since he said a lot of engineers are out at euro astricon |
| 18:43.35 | Strom_C | sevard: um, MWI is definitely FSK on analog sets |
| 18:43.41 | Strom_C | who did you talk to? |
| 18:44.04 | sevard | Strom_C: I didn't catch his name. |
| 18:44.12 | Strom_C | sigh |
| 18:44.16 | Strom_C | which department did you call |
| 18:44.34 | file | Strom_C: fresh green produce department |
| 18:44.49 | Strom_C | heh |
| 18:44.52 | sevard | Strom_C: Presales |
| 18:45.10 | file | dlynes_office: well I can say that's not fresh |
| 18:45.12 | Strom_C | sevard: I really wish you'd taken note of who you talked to |
| 18:45.27 | Waverly360 | Hey! My jokes weren't even bad enough to warrant rotten produce! |
| 18:45.28 | sevard | Strom_C: I do.. but i was so excited I got someone to talk to that i forgot :/ |
| 18:45.34 | Strom_C | dork |
| 18:45.38 | dlynes_office | Waverly360: sure they were |
| 18:45.40 | sevard | yeah yeah |
| 18:45.42 | Strom_C | brb |
| 18:45.44 | sevard | i get girly sometimes |
| 18:45.59 | Waverly360 | dlynes_office: Well..maybe..but no one did :P |
| 18:46.03 | zoa | hey hi |
| 18:46.06 | dlynes_office | Waverly360: see pdtmobile's response :) |
| 18:46.35 | Waverly360 | dlynes_office: ya ya...I'll wait till I have something better next time :P |
| 18:46.50 | dlynes_office | * pdtmobile searches for rotten fruit |
| 18:47.38 | Waverly360 | dlynes_office: Hmph..missed that one. Oh well... |
| 18:51.22 | PepseLap | this is weird. If I just dial 9, it gives me a fast busy after a long pause. if I dial 9 and then -anything-, it gives me a dialtone after a long pause |
| 18:51.26 | PepseLap | that i can actually dial out on |
| 18:51.27 | *** join/#asterisk pdtmobile (n=ptinsley@adsl-070-148-122-203.sip.bna.bellsouth.net) |
| 18:51.36 | PepseLap | i don't understand that |
| 18:51.46 | dlynes_office | PepseLap: exten => 9,1,DISA |
| 18:51.56 | dlynes_office | PepseLap: Not exten => _9,1,DISA |
| 18:52.23 | PepseLap | ok. why do all the examples have the _? |
| 18:52.32 | dlynes_office | PepseLap: _ means the start of a pattern |
| 18:52.40 | dlynes_office | 9 is a very specific extension |
| 18:52.43 | dlynes_office | no need for a pattern |
| 18:53.30 | PepseLap | what about the .? |
| 18:53.42 | dlynes_office | That means any digit |
| 18:53.50 | dlynes_office | well, any character |
| 18:53.54 | Strom_C | . means "any digit string" really |
| 18:53.58 | dlynes_office | You dial alphanumeric extensions as well |
| 18:54.02 | Strom_C | x means any single digit 0-9 |
| 18:54.23 | dlynes_office | for example exten => asterisk,1,1 is a valid extension |
| 18:54.25 | Strom_C | but . could match 395154, for example |
| 18:54.40 | sevard | . is any string of digits until # is pressed, right? |
| 18:54.43 | dlynes_office | most phones do not support alphanumeric phone numbers |
| 18:54.52 | dlynes_office | sevard: the # terminator is a feature of the phone, not asterisk |
| 18:55.21 | *** join/#asterisk darkfiber1 (n=bwarner@65.113.208.35) |
| 18:55.28 | dlynes_office | sevard: it's been a standard in telephony for quite some time...most analog phones and digital phones support that as well |
| 18:55.44 | darkfiber1 | does anyone know how to register a motorola vt1005? |
| 18:55.58 | sevard | phones are weird. |
| 18:56.04 | PepseLap | so would you only use ./x in conjunction with _? |
| 18:56.14 | dlynes_office | darkfiber1: are you in europe? |
| 18:56.20 | dlynes_office | PepseLap: i believe so |
| 18:56.24 | PepseLap | cool |
| 18:56.25 | darkfiber1 | no...u.s. |
| 18:56.26 | dlynes_office | PepseLap: but don't quote me on that |
| 18:56.26 | darkfiber1 | why? |
| 18:56.34 | dlynes_office | PepseLap: read up on extensions in the wiki |
| 18:57.01 | dlynes_office | darkfiber1: Well, if it's a european only phone, you might not catch anyone on at this time, too |
| 18:57.23 | dlynes_office | darkfiber1: i've noticed you've asked that question about 5 or 6 times already today |
| 18:57.29 | dlynes_office | darkfiber1: and no responses |
| 18:57.41 | darkfiber1 | its a voice terminal..I hear Vonage uses them...I have several sipura ata's that work...just thought I'd try this one |
| 18:57.59 | dlynes_office | oh |
| 18:57.59 | dlynes_office | it's an ATA? |
| 18:58.02 | darkfiber1 | basically...it acts as a router as well |
| 18:58.13 | dlynes_office | is it sip based? |
| 18:58.21 | *** part/#asterisk RF_MIA (n=mw1@ip67-93-229-222.z229-93-67.customer.algx.net) |
| 18:58.37 | lunk | if i'm compiling asterisk from source, what do i have to do to enable the MYSQL helper application for the dialplan? |
| 18:58.38 | darkfiber1 | supposedly.. |
| 18:58.47 | dlynes_office | darkfiber1: i don't see why it wouldn't work, then |
| 18:59.00 | dlynes_office | lunk: install asterisk-addons |
| 18:59.20 | PepseLap | in this exten => 300,1,Macro(stdmeetme,300)).. what is the second 300? is that a reference to anything else? |
| 18:59.25 | lunk | dlynes_office: it is |
| 18:59.30 | *** join/#asterisk zwelch (n=chatzill@pdpc/supporter/sustaining/zwelch) |
| 18:59.36 | lunk | dlynes_office: i may have missed a configuration option, just not sure which one :/ |
| 18:59.46 | dlynes_office | lunk: ls -al /usr/lib/asterisk/modules/*sql* |
| 18:59.46 | dlynes_office | erm |
| 18:59.47 | dlynes_office | lunk: ls -al /usr/lib/asterisk/modules/*mysql* |
| 18:59.49 | darkfiber1 | well, unlike the sipuras..it just asks for the proxy...but I'm not sure how it interfaces with sip.conf |
| 18:59.58 | dlynes_office | sipuras ask for a proxy also |
| 19:00.02 | lunk | let me go look |
| 19:00.12 | darkfiber1 | right..and a password...however this vt doesn't have those options.. |
| 19:00.17 | dlynes_office | lunk: you should see three or four modules there |
| 19:00.35 | darkfiber1 | the manual is lame and the setup features don't give you much options |
| 19:00.36 | groogs | i'm trying to initiate an outbound call (this was working months ago, with 1.0.x).. i have a call file, and i'm moving it into /var/spool/asterisk/outgoing .. it gets deleted, but * does nothing (and says nothing, even with verbose and debug at 9) |
| 19:00.39 | lunk | sharon modules # ls *sql* |
| 19:00.39 | lunk | app_addon_sql_mysql.so cdr_addon_mysql.so cdr_pgsql.so res_config_mysql.so |
| 19:00.45 | dlynes_office | darkfiber1: it probably doesn't register with the sip provider |
| 19:00.47 | groogs | is there something that changed for outbound call files? |
| 19:00.53 | dlynes_office | darkfiber1: but that's not necessary, anyways |
| 19:00.56 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 19:01.10 | darkfiber1 | the password you mean? |
| 19:01.19 | dlynes_office | lunk: then you're not loading one of those modules in your modules.conf file |
| 19:01.19 | PMantis | groogs, the call file format is probably wrong for the version you're running... check the wiki. |
| 19:01.33 | lunk | dlynes_office: i was already there, thanks dude |
| 19:01.51 | lunk | learn something new everyday. |
| 19:01.53 | lunk | :) |
| 19:01.54 | dlynes_office | darkfiber1: you don't need to have the phones register with asterisk in order to make calls |
| 19:02.23 | darkfiber1 | right....but getting it to register and give me a dial tone is what I'm not getting.. |
| 19:02.23 | PMantis | dlynes_office, But it's nice fore *receiving* calls. :) |
| 19:02.29 | dlynes_office | darkfiber1: set them to static ip addresses, and then set up your sip.conf appropriately |
| 19:02.40 | dlynes_office | PMantis: you can still receive calls without registering, too |
| 19:02.57 | PMantis | dlynes_office, Yes... but static IPs and such are a pain IMO. |
| 19:03.06 | dlynes_office | PMantis: yeah, they are a pain |
| 19:03.09 | darkfiber1 | tried that.....just thought I'm missing something.. |
| 19:03.18 | dlynes_office | PMantis: but when you're in a situation where you don't have that info, you're kinda hooped |
| 19:03.23 | PMantis | dlynes_office, I did say "nice" not "required". :) |
| 19:03.52 | dlynes_office | darkfiber1: are you logging on as the admin user into the ata's interface? |
| 19:04.11 | darkfiber1 | thats the thing..it doesn't have one.. |
| 19:04.33 | dlynes_office | darkfiber1: it doesn't have an admin user? or the admin user isn't documented? |
| 19:04.56 | dlynes_office | darkfiber1: i'm guessing it's the latter |
| 19:04.56 | darkfiber1 | no.. |
| 19:05.21 | darkfiber1 | no admin user...just the interface... |
| 19:05.24 | groogs | PMantis: http://pastebin.ca/72277 does that look ok for 1.2? |
| 19:05.32 | darkfiber1 | it asks for the proxy...thats it.. |
| 19:05.33 | dlynes_office | darkfiber1: it probably has an admin user, but because you're not an ITSP, you probably didn't get the information to be able to log in as an admin user |
| 19:05.52 | dlynes_office | darkfiber1: proxy, or outbound proxy? |
| 19:05.57 | darkfiber1 | proxy |
| 19:06.11 | dlynes_office | how about a registrar server? |
| 19:06.36 | PMantis | groogs, I dunno, haven't tried that lately.. but the symptoms make me think it doesn't read the format correctly. |
| 19:06.37 | darkfiber1 | is has server address: proxy...and a section that asks for server directory.. |
| 19:07.01 | dlynes_office | darkfiber1: any chance of pb'ing a screen capture? |
| 19:07.02 | dlynes_office | ~pb |
| 19:07.03 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 19:07.11 | groogs | PMantis: yeah, its irritating that it doesn't output anything to the console though |
| 19:07.35 | PMantis | groogs, agreed |
| 19:07.47 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 19:08.06 | PMantis | groogs, ok, lemme try it |
| 19:08.12 | PepseLap | if you just wanted to check for 7 numbers dialed, would you do _XXXXXXX? |
| 19:08.21 | dlynes_office | PepseLap: correct |
| 19:08.26 | PepseLap | thanks |
| 19:08.41 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 19:09.18 | PepseLap | but will it check for that even if you have an extension that matches the first couple numbers? |
| 19:09.32 | PepseLap | like, say you have a 707 extension, and the 7digit number you're dialing starts with 707 |
| 19:10.08 | dlynes_office | PepseLap: it all depends on your dialplan, and whether you're using include => statements or not |
| 19:10.09 | heison | does anyone know if Asterisk will benefit with CPUs with multiple core? i.e. does it run things in threads? |
| 19:10.26 | dlynes_office | heison: yes, it runs multiple threads |
| 19:10.30 | mog | yes |
| 19:10.34 | mog | loads of threads |
| 19:10.48 | PepseLap | dlynes_office: i am using include statements |
| 19:10.50 | dlynes_office | one thread per call if i remember correctly |
| 19:10.54 | PepseLap | just modifying the xorcom distro |
| 19:10.56 | PepseLap | rapid or whatever |
| 19:11.10 | dlynes_office | PepseLap: #include "..." or include => ...? |
| 19:11.16 | heison | if i had to pick between 2 faster CPUs vs 1 CPU with 2 cores, the later would be more beneficial right? |
| 19:11.27 | PepseLap | sorry, #include |
| 19:11.38 | dlynes_office | PepseLap: yeah...totally different anima |
| 19:11.43 | dlynes_office | s/anima/animal/ |
| 19:11.46 | heison | let's say the single CPU with 2 core has a slower clock |
| 19:11.54 | PMantis | groogs, I can't get mine working either... |
| 19:12.37 | PepseLap | there's a include => trunk-macros-common |
| 19:12.47 | PepseLap | in macro-dialout |
| 19:12.47 | PepseLap | hehe |
| 19:13.02 | dlynes_office | heh |
| 19:13.11 | dlynes_office | so i guess you got your problem solved, eh? |
| 19:13.17 | lunk | yes sir |
| 19:13.27 | groogs | PMantis: hm, i wonder if something broke? |
| 19:13.37 | sevard | the dude doesn't drink |
| 19:13.38 | lunk | now i just need to figure out what dtmfmode to use with voipjet |
| 19:14.25 | dlynes_office | lunk: more than likely rfc2833 |
| 19:14.38 | dlynes_office | sevard: sure i do...just not very often :) |
| 19:14.50 | dlynes_office | sevard: and with this heat, it's a little more often |
| 19:15.10 | dlynes_office | sevard: a nice cold sleeman's goes down real good on these hot days |
| 19:15.22 | lilalinux | has anyone got "receive_fax" running? |
| 19:15.30 | dlynes_office | receive_fax? |
| 19:15.37 | dlynes_office | never heard of it |
| 19:15.42 | pdtmobile | rxfax? |
| 19:15.43 | dlynes_office | i think you mean app_rxfax? |
| 19:16.04 | *** join/#asterisk dwildes2 (n=dwildes@209.164.237.195) |
| 19:16.13 | lilalinux | dlynes_office: asterisk-app-fax in debian |
| 19:16.35 | sevard | dlynes_office: http://live.str3am.com:2750/listen.pls |
| 19:16.44 | mrtwister | debian build is not too best |
| 19:16.53 | lilalinux | mrtwister: oh, yeah plenty of bugs |
| 19:17.03 | lilalinux | many paths broken |
| 19:17.35 | mrtwister | i installing / removing asterisk and zaptel with apt-get, to install related stuff |
| 19:17.39 | mrtwister | then compiling |
| 19:18.09 | dlynes_office | sevard: no speakers on this machine |
| 19:18.14 | sevard | gay |
| 19:18.17 | sevard | it's a good song |
| 19:18.27 | lilalinux | mrtwister: good luck |
| 19:18.28 | dlynes_office | why do i need speakers in an office environment? |
| 19:18.43 | sevard | i bring headphones to my office |
| 19:19.10 | dlynes_office | I'd rather use the 220W headend unit in my car |
| 19:19.28 | dlynes_office | my car's my portable office :) |
| 19:19.41 | dlynes_office | and with this heat, I'm usually playing a lot of Bob :0 |
| 19:19.45 | sevard | it was bruce springstine - mrs. mcgrath |
| 19:19.54 | sevard | wtf is Bob |
| 19:20.00 | dlynes_office | Bob Marley man |
| 19:20.06 | sevard | ah |
| 19:20.09 | dlynes_office | he's the master |
| 19:20.21 | dlynes_office | and shaggy |
| 19:20.23 | dlynes_office | and sizzlah |
| 19:20.25 | sevard | dlynes |
| 19:20.31 | sevard | i think you'd like |
| 19:21.02 | dlynes_office | I don't smoke dat shiet |
| 19:21.08 | dlynes_office | or cigarettes for that matter |
| 19:21.13 | dlynes_office | I just love da reggae man |
| 19:22.04 | PepseLap | what is ${EXTEN:1}? |
| 19:22.19 | sevard | ah that's it |
| 19:22.21 | sevard | Xavier Rudd |
| 19:22.24 | dlynes_office | PepseLap: ${EXTEN} without the first digit |
| 19:22.28 | sevard | get his Solace CD dlynes_office |
| 19:22.34 | PepseLap | dlyne: ah, awesome |
| 19:23.02 | sevard | http://www.google.com/musicl?lid=1dg9xVn5al&aid=fUUlrdHPT5O |
| 19:23.04 | dlynes_office | PepseLap: you really need to do some reading on the wiki on dialplans, extensions, standard predefined extensions, dial command, all that kinda stuff |
| 19:23.23 | lunk | heh, any luck if i can't get userinput with rfc2833 or inband? |
| 19:23.25 | lunk | err |
| 19:23.27 | lunk | any thoughts |
| 19:23.35 | dlynes_office | lunk: what's user input? |
| 19:23.46 | dlynes_office | lunk: we're talking a phone system here, not a computer |
| 19:23.46 | [koss] | what's the best Distro to use for asterisk? |
| 19:23.51 | dlynes_office | [koss]: all of them |
| 19:23.56 | rob0 | haha |
| 19:23.58 | [koss] | haha |
| 19:24.00 | [koss] | all at once? :) |
| 19:24.13 | [koss] | fine, i'll use gentoo |
| 19:24.13 | dlynes_office | [koss]: are you really trying to be flame bait? |
| 19:24.15 | rob0 | Definitely NOT. Learn one well. |
| 19:24.19 | lunk | dlynes_office: after calling someone, i want that user to press 1, or 2 |
| 19:24.37 | *** join/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com) |
| 19:24.42 | dlynes_office | [koss]: asking what the best distro is to use, is just inviting everyone to have a war over what the best distro is :) |
| 19:24.52 | Bullseye_Network | WINDOWS!! |
| 19:25.16 | [koss] | lol |
| 19:25.20 | dlynes_office | [koss]: but in all seriousness, personally I would suggest a distro that doesn't involve compiling on production machines |
| 19:25.34 | [koss] | true... |
| 19:25.38 | [koss] | i <3 gentoo tho |
| 19:25.42 | dlynes_office | [koss]: compiling on a production machine can have a very adverse affect on call quality |
| 19:25.43 | Nugget | Linux is poo. |
| 19:25.54 | dlynes_office | Nugget is poo, just like telnet. |
| 19:25.55 | Nugget | telnet is eeeeeeevil! |
| 19:25.55 | [koss] | okay, what do you all use |
| 19:26.00 | rob0 | haha here we go |
| 19:26.07 | PMantis | oh, no |
| 19:26.13 | file | Nugget: sweet & sour sauce! |
| 19:26.14 | [koss] | you can privmsg msg and avoid any flame ;) |
| 19:26.17 | Nugget | I use Linux, Windows, FreeBSD, OpenBSD, and Mac OS X. |
| 19:26.30 | Nugget | they all suck, but thankfully in very different ways. |
| 19:26.32 | dlynes_office | I use Slackware, Windows 2000, Windows XP |
| 19:26.33 | PMantis | I left #freepbx because of a "stupid american" vs "dumbass canadian" heated discussion. |
| 19:26.43 | rob0 | Slamd64 and Slackware :) |
| 19:26.44 | dlynes_office | PMantis: that won't happen here |
| 19:26.51 | dlynes_office | PMantis: a good number of us are Canadian here :) |
| 19:26.53 | EinsteinTaylor | CunningPike: Still around? |
| 19:27.05 | PMantis | hah |
| 19:27.16 | rob0 | (My * is now on slamd64; it was split between 2 Slackwares.) |
| 19:27.25 | dlynes_office | PMantis: so we've got enough weight to push around to make the americans see it our way :) |
| 19:27.31 | jbalcomb | Yeah, that's why whenever my Asterisk server goes down I blame Canada. |
| 19:27.44 | tamp4x | what would be the cause of being able to receive calls on a zap t1, but not being able to make outbound (yea the dial plan is good) ..... when the person tries to dial, they just get a busy .... these are two 4 ports t1 cards, all the other ports are fine....wtf |
| 19:27.48 | EinsteinTaylor | openBSD comes out of canada...they got my support |
| 19:27.50 | dlynes_office | PMantis: i know...you're in Rochester, NY |
| 19:27.52 | PMantis | "american" means Canada, too! |
| 19:28.01 | PMantis | err "America" |
| 19:28.10 | sevard | I'm really getting used to this Fluxbox |
| 19:28.22 | jbalcomb | PMantis I think you we're just as right the first time. ;) |
| 19:28.26 | sevard | I thought i'd never be able to migrate back from my TTY to a X window manager |
| 19:28.46 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 19:28.54 | tamp4x | i upgrades my verison of asterisk from 1.0 to 1.2.9 a week agoo... |
| 19:29.11 | jbalcomb | tamp4x that would seem like breaking a lot of things... |
| 19:29.24 | dlynes_office | EinsteinTaylor: yeah, but the guy that runs it has a huge ego :) |
| 19:29.58 | EinsteinTaylor | maybe...but i run the OS every day...i dont ever have to talk to him |
| 19:30.14 | tamp4x | so any ideas? |
| 19:31.10 | lilalinux | pdtmobile, dlynes_office: yes it's rxfax, do you have it running? |
| 19:31.13 | *** join/#asterisk copland (n=stonecol@209.216.65.10) |
| 19:31.16 | [koss] | i'm looking to replace an old Telrad PBX w/ asterisk.. i only have 1 RJ45 drop to cubes...what's a good fairly inexpensive phone w/ dual RJ45? |
| 19:31.27 | tamp4x | i guess my next solution is to downgrade? |
| 19:31.27 | pdtmobile | lilalinux: ya |
| 19:31.47 | *** join/#asterisk blitz[laptop] (n=blitzrag@217.205.226.210) |
| 19:31.51 | copland | is there any way to get asterisk to support H323 exts I have a couple of phones where the h323 firmware is just alot better than the sip firmware |
| 19:31.56 | PMantis | [koss], Polycom makes great 2 RJ45 phones |
| 19:32.10 | EinsteinTaylor | back to this damn iax2 trunk :( |
| 19:32.14 | blitz[laptop] | copland: check asterisk-addons for an H.323 driver |
| 19:32.20 | PMantis | [koss], but if you want quality, you have to spend a little |
| 19:32.23 | Nugget | I think I would really enjoy fluxbox if I could ever develop the motivation to learn LISP. |
| 19:32.42 | blitz[laptop] | I just use gnome... |
| 19:32.46 | lilalinux | pdtmobile: can I see your dialplan? |
| 19:33.08 | pdtmobile | [koss]: one thing to be careful of, is check the speed of the dual port models, alot of the linksys ones for example are 10mb |
| 19:33.18 | [koss] | eww gross |
| 19:33.36 | [koss] | Polycom says 100....so i'll need what else...PoE switches |
| 19:33.53 | [koss] | a card for the asterisks server for the lines comming in? |
| 19:34.05 | pdtmobile | [koss]: you can run the polycoms off of wall-warts, but i don't suggest it. Most people like their phones to work when the power is out |
| 19:34.29 | [koss] | haha so run a big ass UPS on the PoE switch? |
| 19:34.45 | pdtmobile | Also with the polycom phones, you need to specify that you want PoE, the 301 and and the 501 need a PoE adapter, the 601 supports it native |
| 19:34.49 | pdtmobile | [koss]: pretty much |
| 19:36.04 | [koss] | http://www.voipsupply.com/product_info.php?products_id=1691 |
| 19:36.11 | [koss] | ^ that says PoE |
| 19:37.36 | pdtmobile | i don't know what that phone is |
| 19:37.46 | pdtmobile | it looks like a 301 |
| 19:38.01 | *** join/#asterisk feld (n=feld@12.148.212.157) |
| 19:38.23 | dlynes_office | lilalinux: i've had varying degrees of success with it |
| 19:38.29 | dlynes_office | lilalinux: but mostly a lack of success |
| 19:38.33 | copland | Enn I will just wait in hopes that avaya will fix there sip firmware |
| 19:38.37 | PMantis | pdtmobile, My two IP501's came with a PoE adapter built into the network cable |
| 19:38.46 | *** join/#asterisk feld (n=feld@12.148.212.157) |
| 19:38.47 | dlynes_office | lilalinux: i'm switching over to a sangoma card on the pri tonight, so hopefully that'll solve my faxing problems |
| 19:38.59 | pdtmobile | PMantis: correct, but the base model comes with a power injector cord |
| 19:39.01 | Bullseye_Network | We are having problems with not hearing the ringing, we are calling via SIP to a VOIP provider, If I force the ringing with ,r we have audio problems, they cant hear us when they answer the phone. Any Ideas? |
| 19:39.06 | dlynes_office | lilalinux: i was getting hdlc/d-channel problems on my digium card |
| 19:39.19 | dlynes_office | lilalinux: i suspect that might have had a lot to do with faxing not working |
| 19:39.28 | pdtmobile | PMantis: you just need to make sure you specify when you order that you want one with a PoE cable that will turn it into the jacked up power that the 301 and 501 expect |
| 19:39.28 | PMantis | pdtmobile, yes. That's what you meant |
| 19:39.51 | dlynes_office | Bullseye_Network: ditch the voip provider and use someone else |
| 19:40.06 | pdtmobile | dlynes_office: what are you using for faxing right now, what interface? |
| 19:40.26 | dlynes_office | pdtmobile: the only one i've used so far that's been reliable is bridging of two pri channels |
| 19:40.39 | Spy000007 | Bullseye_Network: What provider? |
| 19:40.46 | Bullseye_Network | cvc termination |
| 19:41.00 | lilalinux | dlynes_office: I get perl errors not finding files, that _are_ there, so I guess I'm not as far as you |
| 19:41.00 | Spy000007 | Did they provide any tech support to solve your problem? |
| 19:41.02 | dlynes_office | pdtmobile: so i'm going to be testing with fxo-fxs bridging, and app_rxfax/app_txfax on the sangoma after I get it up and running |
| 19:41.09 | pdtmobile | i haven't tried to get it working on the 1 port sangoma yet, have got it working on 1 port digium t1 |
| 19:41.18 | dlynes_office | lilalinux: for? |
| 19:41.37 | dlynes_office | pdtmobile: using app_rxfax/app_txfax? |
| 19:41.45 | pdtmobile | ya |
| 19:41.50 | dlynes_office | pdtmobile: ah |
| 19:42.07 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 19:42.08 | pdtmobile | have it patched into my rpm build |
| 19:42.25 | Spy000007 | If changing providers is an option, I would highly recommend VoicePulse Connect... I don't know much about CVC, perhaps there are other cusotmers here who can provide help |
| 19:45.17 | SexyKen | Hey guys |
| 19:45.31 | SexyKen | How can I setup an SIP account that does not offer voicemail at all. |
| 19:45.58 | dlynes_office | SexyKen: don't use a mailbox= line in that account configuration in sip.conf |
| 19:46.20 | SexyKen | I'm not but it seems to do it anyway |
| 19:46.41 | pdtmobile | i have a patch for pbx.c that I am going to finish regression testing tomorrow, how do I officially submit it? |
| 19:47.05 | nortex | SexyKen, Does your dial plan check for the voicemail before sending the call? |
| 19:47.52 | SexyKen | Not sure about that.... |
| 19:48.56 | dlynes_office | pdtmobile: post it on bugs.digium.com |
| 19:49.01 | pdtmobile | k |
| 19:49.23 | nortex | SexyKen, In mine I use SIPpeer to lookup the mailbox and then a gotoIF to route the call there or on to a unavaliable message |
| 19:49.26 | dlynes_office | pdtmobile: and explain the reason for the patch, and what kind of regression testing you've done |
| 19:49.40 | pdtmobile | dlynes_office: can do |
| 19:49.46 | dlynes_office | pdtmobile: that way Kevin can make a decision on whether to include it in trunk |
| 19:50.09 | *** join/#asterisk podzap (n=podzap@roswell.pp.saunalahti.fi) |
| 19:50.09 | lilalinux | dlynes_office: got it now, there were problems with a dot in the directory name and special chars in the context, now I'm waiting for a free line to test some real fax :) |
| 19:50.10 | podzap | lilalinux: you want to test fax with me? |
| 19:50.15 | podzap | i want somebody to send me a fax, to see if it works. |
| 19:50.15 | pdtmobile | dlynes_office: makes sense to me |
| 19:50.19 | dlynes_office | pdtmobile: basically you'll go through about a month to a month and a half of feedback on it with various developers |
| 19:50.26 | darkfiber1 | I'm getting a: warning chan_sip.c:695 retrans_pkt error...but everything is working properly..any ideas? |
| 19:50.29 | dlynes_office | pdtmobile: so you'll want to put a monitor on your report as well |
| 19:50.47 | dlynes_office | pdtmobile: if kevin is satisfied with it, he'll submit it to trunk |
| 19:50.47 | SexyKen | nortex: Well I'm setting up an extension that I want to forward to someones phone -- but the extension wont have a device connected to it....any ideas? |
| 19:51.04 | dlynes_office | lilalinux: cool |
| 19:51.24 | podzap | all i want to do is get my * to receive fax and write a tiff file. going from fax -> spa-2002 -> * (rxfax) does not work. |
| 19:51.28 | pdtmobile | dlynes_office: cool, it'll be in my build either way :) Just want to do the "right thing" and give it back |
| 19:51.44 | podzap | anybody know other programs i could use to receive a fax as a file, via my spa-2002? |
| 19:51.51 | dlynes_office | pdtmobile: generally before starting work on a patch, you might want to check bugs.digium.com to see if someone's already written something similar |
| 19:52.04 | dlynes_office | pdtmobile: that way you're not reinventing the wheel |
| 19:52.38 | nortex | SexyKen, So like dialing 1111 goes to Bob's cell phone and you don't want any voicemail setup. |
| 19:52.42 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 19:52.49 | pdtmobile | not a single reference to it |
| 19:52.59 | dlynes_office | pdtmobile: there ya go |
| 19:53.02 | SexyKen | nortex: Right -- and there isn't a hardware phoen connected to it. |
| 19:53.03 | dlynes_office | pdtmobile: so post away :) |
| 19:53.06 | SexyKen | It should just always go to bobs phone |
| 19:53.22 | pdtmobile | well, i need to do my final testing, don't want to submit something I wouldn't use myself hehe |
| 19:54.27 | *** join/#asterisk h3x0r4t0r (n=h3xor@64.192.116.17) |
| 19:54.32 | nortex | SexyKen, So then you would do exten => 1111,1,Dial(ZAP/g1/<bob'sphone>) and followit with either a playback message or hangup. |
| 19:55.14 | nortex | SexyKen, I used ZAP and g1 assuming that would be the pstn access. |
| 19:55.36 | *** part/#asterisk podzap (n=podzap@roswell.pp.saunalahti.fi) |
| 19:56.02 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 19:56.23 | darkfiber1 | anyone seen an error "maximum retries exceeded on call etc, etc... along with chan_sip.c:695 retrans_pkt ?? |
| 19:56.35 | darkfiber1 | warning msg.... |
| 19:57.03 | EinsteinTaylor | in the dialplan...how would I write the extension so that it doesnt care if the area code was entered or not |
| 19:57.13 | EinsteinTaylor | i have a DID number from one of our providers |
| 20:00.13 | nortex | EinsteinTaylor, Do you mean on outgoing calls? or on a DID coming in that you want to route to a particular internal extension? |
| 20:00.16 | lunk | voip-info.org down? |
| 20:00.40 | EinsteinTaylor | incoming |
| 20:00.53 | EinsteinTaylor | so that when i dial my # from some other phone it doesnt care if i dialed the area code or not |
| 20:01.14 | darkfiber1 | lunk>appears so |
| 20:01.40 | lunk | makes me a sad panda |
| 20:01.53 | *** part/#asterisk darkfiber1 (n=bwarner@65.113.208.35) |
| 20:04.01 | *** join/#asterisk dgilmore (n=dennis@fedora/dgilmore) |
| 20:04.09 | fholmes | voip-info.org is down right now. Crashed on me about 15 minutes ago or so. Right in the middle of a page load. |
| 20:05.16 | *** join/#asterisk wes103 (n=wschaffe@c069.centercall.com) |
| 20:05.27 | nortex | EinsteinTaylor, From another internal phone? |
| 20:05.34 | EinsteinTaylor | nope |
| 20:05.41 | dlynes_office | fholmes: good job |
| 20:05.41 | EinsteinTaylor | just a normal o'l phone |
| 20:06.54 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.205.132.43) |
| 20:07.12 | pdtmobile | dlynes_office: i assume the disclaimer needs to be faxed in as well? |
| 20:07.18 | nortex | EinsteinTaylor, Okay, how many numbers does you provider send you on the DID? One of mine sends the last 4 the other all 7. In either case I have a exten for those digits that Dials my actual phone internally. |
| 20:07.31 | dlynes_office | pdtmobile: correct |
| 20:07.49 | dlynes_office | pdtmobile: you won't get any acknowledgement that they've received it, either |
| 20:07.55 | pdtmobile | hehe ok |
| 20:08.02 | dlynes_office | pdtmobile: there's two agreements |
| 20:08.10 | dlynes_office | pdtmobile: you can pick whichever you like best |
| 20:08.22 | dlynes_office | pdtmobile: one that allows you to donate your first born to Digium |
| 20:08.28 | dlynes_office | pdtmobile: and the other's a little more sane |
| 20:08.46 | SexyKen | Hrm |
| 20:08.49 | SexyKen | nortex...I did that |
| 20:09.05 | SexyKen | Will that only work if the device is not connected? |
| 20:09.13 | rob0 | I cannot enter into any contracts on behalf of my firstborn ... he's 21. |
| 20:09.35 | nortex | SexyKen, Can you pastebin the dialplan section? |
| 20:10.17 | pdtmobile | i like the short one, cause it's short :) |
| 20:10.47 | dlynes_office | Well, some people have a problem with it, and that's why the second one exists |
| 20:10.57 | dlynes_office | I don't have a problem with the first one, but my boss does |
| 20:11.06 | dlynes_office | And he's paying my paycheque, so I signed the second one :) |
| 20:11.43 | nortex | dlynes_office, Does the first one require you to give up your boss? |
| 20:11.57 | pdtmobile | the first one makes sense to me, none of my other intellectual property would preclude me from releasing this modification to asterisk. Well according to the license of asterisk I am already agreeing to that |
| 20:12.09 | SexyKen | Actually, nortex, I got it to clal just the cell -- that worked -- now is there anyway to call his device as well as his cell if the device is up? |
| 20:12.53 | dlynes_office | nortex: nah...my boss is pretty cool |
| 20:12.54 | nortex | SexyKen, Yes, but what do you want to happen if his device is down? |
| 20:13.03 | SexyKen | Just ring his cell |
| 20:14.06 | pdtmobile | well I sent it to the CEO to see how he feels, but I don't imagine he will freak out |
| 20:14.46 | dlynes_office | pdtmobile: yeah...either one is fine |
| 20:14.54 | dlynes_office | pdtmobile: my boss is just a little paranoid |
| 20:15.08 | pdtmobile | sounds like he needs a tin foil hat for christmas |
| 20:15.29 | NotJohnDavid | ha I just found a big explination of the s extension :p |
| 20:15.42 | *** join/#asterisk davegrin (n=da5id@24-38-58-10-st.lndnnh.adelphia.net) |
| 20:16.24 | pdtmobile | where? |
| 20:16.25 | nortex | SexyKen, I have not tried this, but, you could have his extension actually dial 2 extensions, one his device and the other his cell phone. I do this with some cordless phones. So I have say device ext 1112 and his cell at 1111, but people call him at 1113 which is Dial(local/1112&Local/1111). I have not tested what would happen if the device is disconnected. |
| 20:16.51 | lilalinux | hooray |
| 20:17.22 | *** part/#asterisk dgilmore (n=dennis@fedora/dgilmore) |
| 20:17.22 | NotJohnDavid | pdtmobile: the dialplan chapter in the o'reily book. I had read the first couple chapters and they were using exten => s,1,something and never mentioned what it was or why they were doing it. then i realized chapter 5 is Dialplan. so s is short for start like I believe you said earlier |
| 20:17.23 | nortex | SexyKen, This is where a find-me follow-me setup is helpful, ring one device is avaliable if not goto the cell. voip-inf.org has some examples of that. |
| 20:17.46 | pdtmobile | NotJohnDavid: good, now I know where to send people :) |
| 20:18.00 | NotJohnDavid | pdtmobile: which this book is available for free download as a pdf, btw |
| 20:18.19 | sevard | AAAAAAAAAAAaaaaaaaaaaaaaaaaaahhhhhhhhhhhhhhh |
| 20:18.21 | sevard | So |
| 20:18.37 | sevard | can digum cards periodicaly reverse the line polarity? |
| 20:19.12 | NotJohnDavid | why would they go and do that? |
| 20:19.27 | dlynes_office | sevard: yeah...put a timer in your agi script to periodically hang up the line |
| 20:19.45 | dlynes_office | but it seems like a pretty assinine thing to do |
| 20:19.51 | sevard | periodically hang up the line... it's going to have to sense if the phone is off hook or not |
| 20:19.51 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 20:20.03 | sevard | dlynes_office: it's a way to support legancy phones |
| 20:20.15 | PakiPenguin | evening |
| 20:20.29 | Strom_C | sevard: merely reversing the line polarity causes the lamp to light? |
| 20:20.34 | sevard | that's right! |
| 20:20.39 | Strom_C | sevard: wow, who knew |
| 20:20.47 | Strom_C | :) |
| 20:20.58 | sevard | if you tap your hook and see if when you hang up the light flashes you have a phone that's sensative to line polarity reversal |
| 20:21.00 | Strom_C | sevard: have you tried this with a reversing cord? |
| 20:21.13 | sevard | i'm going to go to radio shack later today and get one |
| 20:21.17 | Strom_C | just make one |
| 20:21.22 | sevard | i don't have parts |
| 20:21.24 | sevard | whatchu got |
| 20:21.32 | Strom_C | or hell, rewire the jack backwards |
| 20:21.42 | sevard | oh, i could make a cross over cable |
| 20:22.05 | Strom_C | um, dude, it's a satin-flex cord. you dont make "crossover cables" for phones |
| 20:22.23 | Strom_C | step 1: unscrew wall jack |
| 20:22.29 | sevard | now the engineer was saying that reversing ring/tip is a bit different than reversing polarity on the line, can you explain that? |
| 20:22.34 | Strom_C | step 2: reverse tip and ring |
| 20:22.45 | dlynes_office | Strom_C: sure you do...loopback connector for pri :p |
| 20:22.46 | Strom_C | which engineer was this? |
| 20:23.00 | sevard | the head analoug engineer from scitec |
| 20:23.03 | dlynes_office | ummm |
| 20:23.03 | sevard | analoge |
| 20:23.08 | Strom_C | what an idiot |
| 20:23.10 | dlynes_office | reversing tip and ring is reversing polarity |
| 20:23.16 | Strom_C | reversing line polarity == reverse tip and ting |
| 20:23.18 | Strom_C | er ring |
| 20:23.20 | sevard | that's what i would have thought |
| 20:23.27 | sevard | son of bitch i'm mother effing digital |
| 20:23.36 | sevard | analog shit is like bieeeetch waazzat shiet |
| 20:24.01 | Strom_C | what he may have been talking about, though is that with a polarity reversal, the equipment stays wired up correctly but reverses the voltages coming in on the physical wires |
| 20:24.06 | dlynes_office | i think sevard's making this shit up |
| 20:24.10 | Strom_C | probably |
| 20:24.14 | sevard | fu |
| 20:24.16 | dlynes_office | he mentioned something else weird a while back, too |
| 20:24.17 | sevard | let me try it |
| 20:24.27 | sevard | i don't have a wall jack but i have some rj11s and a crimper somewhere |
| 20:24.55 | sevard | dlynes_office: what's that |
| 20:24.58 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 20:25.23 | *** join/#asterisk sorush20 (n=sorush20@82-43-184-143.cable.ubr07.newm.blueyonder.co.uk) |
| 20:25.26 | dlynes_office | sevard: what he probably meant that reversing ring/tip is done for a fraction of a second, whereas reversing polarity is probably what he considers to be rewiring |
| 20:25.42 | pdtmobile | hell worst case just cut a cable in half and twist the wires together, it should work to test it out |
| 20:26.25 | dlynes_office | anyways...gotta head downtown |
| 20:26.27 | dlynes_office | lataz |
| 20:26.28 | sevard | children of bitches chillax |
| 20:26.36 | sevard | i have to move my card before i zap myself with wires |
| 20:26.37 | pdtmobile | later |
| 20:26.37 | sevard | bbiab |
| 20:26.46 | sevard | byte byte dlynes_cuntbag |
| 20:26.46 | Strom_C | sevard: dont make me bludgeon you with a western electric 500 set |
| 20:29.55 | sevard | :( |
| 20:30.32 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 20:31.00 | ManxPower | This is the tone setting on my SIPura: 420@-16;10(*/0/1) Does anyone know what the correct /etc/asterisk/inidcations.conf would be to provide the same tone? |
| 20:34.37 | fholmes | If I am using queues in Asterisk are the agents allowed to make outgoing calls? |
| 20:35.47 | SexyKen | Hrm |
| 20:36.23 | SexyKen | I want to dial two numbers at once (when an extension is dialed) but each will have different caller ids |
| 20:36.25 | SexyKen | is this possible? |
| 20:37.02 | NotJohnDavid | why would you want different caller ids? |
| 20:37.11 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 20:37.39 | ManxPower | SexyKen, no it is not |
| 20:38.05 | ManxPower | What you CAN do is Dial a Local/exten and that exten set the callerid |
| 20:38.13 | NotJohnDavid | why couldn't you make two fake extensions and set the callerids from there, then dial? like dial(fake1,fake2) and then define fake1 and fake2 ? |
| 20:38.13 | SexyKen | Well becuase if someone calls from an extension (locally) it should extend the extension so people can view who is calling -- otherwise if someone calls from an outside line.... |
| 20:38.23 | SexyKen | ...it will show their full callerid the way it was recieved |
| 20:40.01 | NotJohnDavid | i'm not following you on how the two are related |
| 20:40.08 | *** join/#asterisk DarKnesS_WolF (n=wolf@82.201.231.106) |
| 20:40.40 | SexyKen | The extension 1103 |
| 20:40.52 | SexyKen | It will ring a SIP Device and it will call an outside line (a cell phone) |
| 20:41.13 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 20:41.26 | SexyKen | If I dial 1103 from my SIP Device locally -- it will set the caller id as 1101 -- this can't go to my cell phone |
| 20:41.35 | SexyKen | So I'd make it 2342341101 |
| 20:41.51 | ManxPower | SexyKen, you have a PRI: |
| 20:41.53 | ManxPower | ? |
| 20:41.57 | SexyKen | nO |
| 20:42.20 | ManxPower | SexyKen, you have VoIP? |
| 20:42.25 | SexyKen | Yes |
| 20:42.52 | ManxPower | Regardless you'll have to do it using Local/ |
| 20:44.29 | robin_sz | great, just why oh why did I colo at the worlds worst ISP? |
| 20:44.44 | Strom_C | because you were too cheap? |
| 20:44.54 | justinu | dlynes: yeah - what's up? |
| 20:45.29 | *** join/#asterisk Eggplant (i=No@dsl-72-19-44-253.cascadeaccess.com) |
| 20:47.02 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 20:48.20 | wes103 | Speaking of rewriting caller id's... I have a sip connection to a "provider" and an IAX2 connection to another asterisk box. A number beginning with a '1' is routed to the provider, and number beginning with a '6' is routed to the IAX2 link. How can I rewrite the incoming caller ids to prepend the appropriate digit so my sip hardphone users can just select the inbound callerid number when calling back? |
| 20:49.14 | pdtmobile | robin_sz: we got to figure out for sprint the other day that they had a bad emergency power config |
| 20:49.31 | justinu | wes103: something like: exten => 1,1,Set(CALLERID(num)=${CALLERIDNUM:2}) |
| 20:49.33 | vader-- | is there a way to setup recording where you can activate it mid call? |
| 20:49.43 | robin_sz | pdtmobile, trust me, my ISP is WAY worse than that ... |
| 20:49.54 | *** join/#asterisk saftsack (n=w@p54A7FAEC.dip.t-dialin.net) |
| 20:50.14 | pdtmobile | hehe |
| 20:50.45 | robin_sz | pdtmobile, they moved my server a while ago, my dual PSU server ... thats a server with two, seperate, psus. liek you could plug each one into a separate power distribution, so if one goes tits up, the other survives |
| 20:50.47 | wes103 | justinu: won't that set it for all calls coming into the extension? I want to do an if based on the source "trunk" if you will. |
| 20:51.05 | saftsack | hi i have a problem. if i one of my 4 zap channels (fxs card), then the channel will be down |
| 20:51.16 | justinu | wes103: put it as the first line into your "trunk's" context |
| 20:51.16 | robin_sz | except, being fuckwits, they plugged em into the same rack distro. and it developed a fualt, so they have taken it offline for maint .. |
| 20:51.20 | saftsack | if i try then to call from this channel i always get a busy signal :( |
| 20:51.23 | robin_sz | 2 fscking hours now :( |
| 20:51.26 | justinu | well, just put it before the dial() |
| 20:51.43 | *** join/#asterisk Monkey13 (n=jcheuvro@69.7.217.140) |
| 20:51.54 | pdtmobile | robin_sz: one of our datacenters was stealing power from us :) one of my guys went into the rack and they had stuff run in through the top of the rack plugged into our circuits |
| 20:52.07 | robin_sz | noce |
| 20:52.10 | robin_sz | nice |
| 20:52.13 | wes103 | justinu: should I have my local extensions in multiple contexts? |
| 20:52.19 | pdtmobile | that was a fun discussion, so wait, you charge us by the watt right? ok, so do I get to bill you back now? |
| 20:52.23 | wes103 | would that sort out my confusion? |
| 20:52.29 | robin_sz | I presume he just unplugged it and tied the wires to the rack before locking the door? |
| 20:52.49 | justinu | wes103: not necessarily, but you have a seperate context for your "provider", right? |
| 20:53.10 | justinu | [provider-inbound] |
| 20:53.12 | file | pdtmobile: nice, how did that end? |
| 20:53.15 | justinu | and [local-extensions] |
| 20:53.17 | justinu | something like that |
| 20:53.19 | pdtmobile | ya, he called me and said, there is stuff plugged into our rack... what should I do |
| 20:53.24 | NotJohnDavid | pdtmobile: i think that would cause me to immediately change datacenters |
| 20:53.24 | wes103 | No, not yet. I am still wrangling with the context concepts. I think they are all in [default]. |
| 20:53.25 | robin_sz | anyway, if anyone in the UK wants a really shit ISP, can I recommend 186K ltd? |
| 20:53.37 | pdtmobile | I said, unplug it, can't be our problem if it goes down |
| 20:53.44 | robin_sz | good call |
| 20:53.45 | pdtmobile | so he is like okey dokey, you da boss |
| 20:53.56 | justinu | in [provider-inbound], you'd have the logic of how to route an inbound call to extensions |
| 20:54.07 | pdtmobile | well, somebody before I got here signed a 3 year contract with them |
| 20:54.12 | pdtmobile | so I am going to use every second of it since I am stuck with it |
| 20:54.28 | robin_sz | yeah |
| 20:54.37 | *** join/#asterisk Waverly360 (n=mirc@adsl-070-148-122-203.sip.bna.bellsouth.net) |
| 20:54.41 | wes103 | justinu: I have a register command in my sip.conf. |
| 20:54.45 | robin_sz | and fill the outbound pipe with test data :) |
| 20:54.49 | NotJohnDavid | I prefer not to sign contracts... fear of commitment and all |
| 20:54.53 | wes103 | in there, it specifies target extensions. |
| 20:55.06 | justinu | you should have a section in sip.conf for your provider |
| 20:55.17 | justinu | with a line that says context=provider-inbound |
| 20:55.18 | wes103 | ah, yes, I do. |
| 20:55.20 | justinu | or something like that |
| 20:55.27 | wes103 | context=from-sip. |
| 20:55.36 | wes103 | Forgot that. It was late at night... |
| 20:55.50 | justinu | so the logic for how to send calls to your extens is in [from-sip] in extensions.conf |
| 20:55.51 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 20:56.29 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 20:57.30 | wes103 | Yes, all of my extenstions in extensions.conf on my system is defined in [from-sip] |
| 20:57.41 | justinu | so that's not really ideal |
| 20:57.49 | wes103 | I am getting that clue, yes. |
| 20:58.03 | justinu | sort out what you can, and come back with specific questions |
| 20:58.09 | justinu | ~pb |
| 20:58.16 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 20:58.19 | wes103 | Well, it worked fine when there was only one external destination. |
| 20:58.22 | justinu | use that for pasting configs |
| 20:58.33 | sorush20 | !FOX |
| 20:58.42 | wes103 | The wrinkle came when a friend and I connected our boxes. |
| 20:58.50 | *** part/#asterisk sorush20 (n=sorush20@82-43-184-143.cable.ubr07.newm.blueyonder.co.uk) |
| 20:59.26 | justinu | you mentioned iax before... the place to set the initial context for those calls would be iax.conf |
| 20:59.39 | wes103 | The basics work. |
| 20:59.55 | wes103 | I can make and receive calls to and from both. |
| 21:00.18 | wes103 | I just can't hit "OK" when I select the incoming number off the IAX2 link. |
| 21:00.23 | pdtmobile | my favorite is from the .com days, i got a call that our main server was down. So I called the datacenter and they went to check on the box. The damn janitor had unplugged our server to plug in his vacuum |
| 21:00.44 | pdtmobile | layer 8 can screw up anything |
| 21:01.24 | justinu | wes103: so create a new context in extensions.conf [from-iax], make it just like [from-sip], but add the line I gave you above to modify the callerID |
| 21:01.40 | NotJohnDavid | pdt: was he running the vacuum off the ups? :) |
| 21:01.41 | wes103 | ok |
| 21:01.45 | justinu | wes103: then modify your iax.conf to point those calls to from-iax by setting "context=from-iax" |
| 21:01.48 | pdtmobile | NotJohnDavid: yep :) |
| 21:01.59 | pdtmobile | can't have it go down in the middle of a cleaning! |
| 21:02.24 | wes103 | justinu: I could use your example with the 's' extension in the context, yes? |
| 21:02.50 | pdtmobile | course thats the same datacenter that put another customer in our full rack "cause we weren't using all of our space" |
| 21:03.07 | NotJohnDavid | but not the same center that stole your power? |
| 21:03.16 | pdtmobile | no, i have luck, lots of it |
| 21:03.17 | justinu | hmm... "s" might not get called; "s" is really only for inbound calls on technologies that don't give you Dialed Number Info, like POTS |
| 21:03.26 | justinu | or macros |
| 21:04.33 | *** join/#asterisk gennaro (n=root@ppp-62-10-136-129.dialup.tiscali.it) |
| 21:04.47 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.205.132.142) |
| 21:04.48 | vader-- | hmmm |
| 21:05.01 | vader-- | anyone else have an issues where their sip phones won't do anything in the features.conf? |
| 21:05.02 | gennaro | hi some help me with 1 sip account in sip config? phone don't let me log |
| 21:05.03 | *** part/#asterisk DrDeke (i=Rusty@causticsoda.engin.umich.edu) |
| 21:05.10 | vader-- | like *1 for recording and stuff |
| 21:05.51 | ManxPower | Anyone have any idea why when I change indications.conf (dial = 420 instead of dial = 350+440) I do not hear anything different when I pick up a zap FXS port? |
| 21:06.10 | justinu | manx: did you do a "restart now"? |
| 21:06.27 | gennaro | may be firewall?! |
| 21:07.00 | ManxPower | justinu, yup |
| 21:07.18 | justinu | correct "zone"? |
| 21:07.36 | gennaro | firewall tanx |
| 21:07.38 | wes103 | justinu: thanks for the tips. Time to bolt. |
| 21:07.39 | gennaro | :) |
| 21:07.58 | saftsack | Jun 27 07:04:30 WARNING[14637]: chan_zap.c:12524 setup_zap: Ignoring switchtype |
| 21:07.59 | saftsack | Jun 27 07:04:30 WARNING[14637]: chan_zap.c:12524 setup_zap: Ignoring signalling |
| 21:08.00 | saftsack | why? |
| 21:08.12 | file | you can't reload to change those, you have to restart Asterisk |
| 21:08.34 | *** join/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 21:09.25 | saftsack | ok |
| 21:09.33 | ManxPower | justinu, http://pastebin.ca/72437 |
| 21:09.37 | *** join/#asterisk Monkey13 (n=jcheuvro@69.7.217.140) |
| 21:09.49 | saftsack | Signalling Type: PRI Signalling |
| 21:09.54 | saftsack | but why does it do this? |
| 21:10.13 | saftsack | i restarted * completely but signalling bri_cpe_ptmp is ignored :( |
| 21:10.35 | vader-- | anyone else have an issues where their sip phones won't do anything in the features.conf? |
| 21:10.36 | vader-- | like *1 for recording and stuff |
| 21:11.08 | mountainm2k | Anybody know of a good WI-FI phone? |
| 21:11.25 | mountainm2k | Just got in a Zyxel 2000W, but it requires WEP, and not WPA |
| 21:11.50 | justinu | ManxPower: that should work... i used indications.conf to modify my SIP inband tones... never for zap channels... but that's what indiciations.conf is really for, iirc |
| 21:11.52 | justinu | (zap channels) |
| 21:12.03 | ManxPower | justinu, yeah. |
| 21:12.15 | justinu | unless somehow, it's calling the "dialoutside" |
| 21:12.20 | *** join/#asterisk RoyK (n=roy@ti211310a080-6081.bb.online.no) |
| 21:12.27 | saftsack | loadzone=nl |
| 21:12.27 | saftsack | defaultzone=nl |
| 21:12.27 | saftsack | span=1,1,3,ccs,ami |
| 21:12.27 | saftsack | bchan=1-2 |
| 21:12.28 | saftsack | dchan=3 |
| 21:12.32 | saftsack | fxoks=4-7 |
| 21:12.34 | saftsack | OH sry for paste |
| 21:12.48 | saftsack | didnt know that it pastes in more than one line |
| 21:15.46 | nortex | On Polycom, how do I set Distinctive ring? I am trying to use Set(ALERT_INFO=DID) with DID defined in the sip.cfg file for the phone. But it does not change the ring. |
| 21:17.07 | *** join/#asterisk Eonz (n=Icarus@200.31.96.145) |
| 21:17.20 | Eonz | good days guys |
| 21:17.33 | SexyKen | Why would this keep happening |
| 21:17.34 | SexyKen | Jun 26 15:13:03 NOTICE[2315]: chan_sip.c:11328 sip_poke_noanswer: Peer '1103' is now UNREACHABLE! Last qualify: 1248 |
| 21:18.25 | Eonz | sorry for my question, but how i can see the trunks state? |
| 21:18.28 | ManxPower | SexyKen, you have qualify=S and the device did not respond to an OPTIONS packet in X seconds |
| 21:18.40 | ManxPower | justinu, it should not be |
| 21:18.56 | ManxPower | dialoutside is just the original dialtone, which I won't be using anyway |
| 21:19.47 | justinu | just grasping at straws |
| 21:21.32 | Eonz | any idea? |
| 21:21.51 | justinu | eonz: what kind of "trunks"? |
| 21:22.42 | rob0 | Swim trunks! <splash> |
| 21:23.01 | justinu | there's an idea... way too hot outside |
| 21:23.18 | Strom_C | where the hell did warm, arid, southern california summers go? |
| 21:23.32 | Strom_C | who the hell brough Alabama weather here? gah |
| 21:23.43 | rob0 | Hey we like to share :) |
| 21:23.46 | Eonz | justinu: zap |
| 21:23.53 | justinu | according to my weather station, humidity is only 35% |
| 21:23.55 | justinu | feels like 90 tho |
| 21:24.10 | rob0 | Not that bad here ATM (Florence, AL) |
| 21:24.41 | justinu | eonz: zap show status |
| 21:24.46 | justinu | eonz: zap show channels |
| 21:25.52 | *** join/#asterisk hypa7ia (i=hypatia@judecca.aculei.net) |
| 21:26.08 | Monkey13 | What is better for a newbie (asterisk newbie not linux newbie) using trixbox or building a asterisk server from scratch in debian or fedora? |
| 21:26.33 | Strom_C | scratch |
| 21:26.42 | ManxPower | justinu, I'm having trouble figireing out what asterisk things the country is |
| 21:26.45 | Strom_C | trixbox will not help you learn |
| 21:27.02 | justinu | manxpower: set a high verbosity, and reload res_indications.so? |
| 21:27.10 | justinu | it spits out all sorts of crap, iirc |
| 21:27.54 | ManxPower | justinu, that did it |
| 21:27.56 | Eonz | justinu: thanks |
| 21:28.05 | ManxPower | set the country to "at". No change in tones |
| 21:28.26 | justinu | that's weird |
| 21:28.28 | justinu | something's not right |
| 21:28.35 | ManxPower | justinu, I know. |
| 21:28.38 | justinu | do you have to set it in zapata.conf? |
| 21:29.06 | Monkey13 | thank storm, I'm building it tonight. I just got a few cisco 7960's and 7912 and a cisco ata 168, I just need to do the sip upgrade to them. Asterisk does not support the cisco protocal correct? |
| 21:29.09 | rob0 | Monkey13: * from scratch is not that difficult. |
| 21:29.35 | rob0 | I spent a few hours yesterday doing that. |
| 21:29.50 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:30.01 | rob0 | I'm not a total n00b to *, but I'm no expert, either. |
| 21:30.02 | Monkey13 | rob0: is the sip upgrade easy |
| 21:30.03 | Dr-Linux | all |
| 21:30.15 | SexyKen | Anyone have issues connecting EyeBeam to Asterisk? |
| 21:30.23 | rob0 | <== unSIPful |
| 21:30.31 | Dr-Linux | SexyKen: never |
| 21:30.35 | rob0 | (I use IAX and Zap) |
| 21:30.48 | SexyKen | I cannot connect using it |
| 21:30.52 | SexyKen | Does asterisk use RTP? |
| 21:31.03 | Dr-Linux | yes |
| 21:31.06 | nortex | SexyKen, Yep |
| 21:31.08 | devel | hey all, do rx/tx gain changes in zapata.conf take at a 'reload' or do you have to do a full restart? |
| 21:31.22 | Monkey13 | rob0: will that (IAX and Zap) work with unflashed cisco gear? |
| 21:31.41 | SexyKen | By default |
| 21:31.56 | *** join/#asterisk podzap (n=podzap@roswell.pp.saunalahti.fi) |
| 21:31.58 | nortex | SexyKen, Yes |
| 21:32.00 | Dr-Linux | devel: just "reload chan_zap.so" |
| 21:32.13 | devel | excellent, thanks Dr-Linux |
| 21:32.17 | justinu | no, it takes a full restart |
| 21:32.19 | podzap | yes, i am able to send from my fax machine -> spa-2002 -> *, and it writes a tiff file! |
| 21:32.28 | devel | excellent, thanks, justinu |
| 21:32.33 | justinu | :) |
| 21:32.47 | podzap | and it was a real sonofabitch to get working, poor, scattered documentation all over the net. |
| 21:33.06 | podzap | i'm thinking of writing a howto for some wiki or something. |
| 21:33.10 | devel | if it is reload, then the customers will shut up. if it takes a full restart, they're just going to have to wait. |
| 21:33.12 | justinu | podzap: please do |
| 21:33.15 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 21:33.27 | SexyKen | What port should I have open |
| 21:33.30 | SexyKen | UDP 5060, right? |
| 21:33.35 | justinu | devel: restart when convenient |
| 21:33.39 | SexyKen | To connect to Asterisk using my EyeBeam |
| 21:33.40 | justinu | (that's an actual command) |
| 21:34.15 | podzap | justinu: ok, i will post it to -users when i get it done |
| 21:34.28 | devel | no, i don't play that game justinu |
| 21:34.29 | justinu | podzap: you might add it to the voip-info.org wiki |
| 21:34.48 | podzap | justinu: ok |
| 21:35.17 | SexyKen | Guys am I wrong? |
| 21:35.24 | SexyKen | SIP Packets travel over UDP port 5060 right? |
| 21:35.41 | RoyK | hehe. the gizmo people sent me an email |
| 21:36.17 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:37.06 | ManxPower | SexyKen, Yes, but not the audio |
| 21:37.19 | SexyKen | Audio is 5061? |
| 21:37.46 | justinu | no, it's dynamic |
| 21:37.56 | SexyKen | Hrm. |
| 21:38.01 | SexyKen | So what ports should be open on a firewall? |
| 21:38.30 | justinu | open up whatever you set in rtp.conf |
| 21:38.35 | justinu | whatever range, that is |
| 21:38.38 | ManxPower | SexyKen, That depends on if Asterisk is behind the firewall and if it's really a firewall and not just a NAT router. |
| 21:38.55 | ManxPower | It also depends on if the client is behind a firewall or just a NAT router or on a public IP. |
| 21:38.57 | SexyKen | Asterisk is not between a firewall |
| 21:38.59 | SexyKen | The client is |
| 21:39.08 | podzap | trust your internal machines, and setup an SPI firewall, end of problemo |
| 21:39.19 | ManxPower | SexyKen, REALLY a firewall or just a NAT router? |
| 21:40.03 | SexyKen | NAT |
| 21:40.35 | Dr-Linux | anybody use sipura 2100? |
| 21:40.54 | podzap | i only got a 2002 |
| 21:41.08 | podzap | still no t.38 |
| 21:41.11 | ManxPower | SexyKen, then you only need nat=yes and disable nat settings on the client |
| 21:41.13 | ManxPower | That is ALL |
| 21:41.18 | Dr-Linux | podzap: i have some problem with spa2100 |
| 21:41.27 | SexyKen | What do you mean disable nat setting son the client? |
| 21:41.40 | podzap | Dr-Linux: upgrade to the latest firmware, then try it again :-) |
| 21:41.57 | Dr-Linux | podzap: sometime it takes too long to after dial |
| 21:42.13 | *** join/#asterisk Un1x (i=seannn@CPE00163636ba58-CM0012257020c0.cpe.net.cable.rogers.com) |
| 21:42.14 | rob0 | Monkey13: Zap is zaptel (Digium) hardware. IAX is a network protocol like SIP but not as widely supported. No, your SIP phones probably do not support IAX. |
| 21:42.15 | podzap | new firmware might fix it |
| 21:42.24 | Un1x | heh.. |
| 21:42.48 | justinu | i like the spa2100s |
| 21:43.10 | podzap | i am thinking about getting a 2100 just for the t.38 support |
| 21:43.11 | Un1x | i like |
| 21:43.12 | Un1x | spa3000 |
| 21:43.12 | Dr-Linux | justinu: hey friend |
| 21:43.15 | Un1x | there better :) |
| 21:43.32 | podzap | afaik spa3000 doesn't support t.38, only the 2100 does |
| 21:43.33 | Dr-Linux | spa3000 is a gw |
| 21:43.56 | Un1x | dr-linux what makes you say that it doesn't differ much from spa2100 |
| 21:43.57 | Dr-Linux | podzap: what's t.38? |
| 21:44.22 | podzap | Dr-Linux: proper fax over voip |
| 21:44.44 | justinu | complicated fax over ip |
| 21:44.45 | Dr-Linux | Un1x: you can use fxo port in spa 3000 |
| 21:44.55 | Un1x | yea thats what is the only difference |
| 21:44.57 | justinu | ~t38 |
| 21:44.59 | jbot | i heard t38 is see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works, no, it's not ready yet, we'll let you know. a really lousy spec. |
| 21:45.10 | Dr-Linux | podzap: data is never good on voip |
| 21:45.14 | justinu | i wonder who thought it was a really lousy spec |
| 21:45.25 | justinu | yeah, it's called Voice over IP for a reason |
| 21:45.30 | justinu | not fax, or modem over IP |
| 21:45.35 | Un1x | lol |
| 21:45.38 | *** join/#asterisk MoutaPT (n=MoutaPT@a83-132-239-109.cpe.netcabo.pt) |
| 21:45.44 | Un1x | justinu doesn't asterisk support incoming fax? |
| 21:45.50 | *** join/#asterisk terrapen (n=cjs@166.70.183.108) |
| 21:45.58 | podzap | well, we can forgive the dumbasses for forgetting to include a very important use case... |
| 21:46.05 | Dr-Linux | Un1x: i does, but on spanDSP signals |
| 21:46.24 | Un1x | it |
| 21:46.25 | Un1x | not i |
| 21:46.35 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 21:46.42 | Dr-Linux | s/i/it |
| 21:47.00 | podzap | what did they think, that people would all of the sudden want to stop faxing? |
| 21:47.08 | Un1x | heh podzap |
| 21:47.12 | Un1x | it supports fax |
| 21:47.14 | Un1x | esential |
| 21:47.20 | Dr-Linux | justinu: any idea, why sipura 2100 takes too long sometime to send dialing request? |
| 21:47.23 | Un1x | fax does is convert data into voice kinda |
| 21:47.32 | Un1x | havent u heard a fax its bunch of static and sounds |
| 21:47.33 | justinu | Dr-Linux: does it depend on the length of the number you call? |
| 21:47.49 | podzap | gnite! |
| 21:48.05 | Dr-Linux | justinu: no, it's only 4 digit extension.. |
| 21:48.18 | MoutaPT | any one knows which is asterisk variable that gets MSNs instead of DID ? |
| 21:48.28 | Dr-Linux | justinu: also sometime i can't hear that the other end is ringing.. |
| 21:48.39 | justinu | Dr-Linux: not sure, you'd need to do some packet sniffing to find out |
| 21:49.20 | Dr-Linux | justinu: this happens with only other location SIP line, the local asterisk extension is just fine. |
| 21:49.39 | *** join/#asterisk |marv0997| (n=marv@65.165.87.198) |
| 21:49.51 | terrapen | so, is there a simpler way to ring a bunch of phones at once besides a very long Dial(.....) call? |
| 21:50.01 | Dr-Linux | justinu: also some time the light goes off for this sip line |
| 21:50.18 | nortex | terrapen, A queue |
| 21:50.36 | Dr-Linux | justinu: i'm just asking you, maybe if it's firewall issue. |
| 21:50.42 | terrapen | nortex, ah, that might work |
| 21:50.52 | terrapen | I'll need to make it so that it rings them all at once |
| 21:51.01 | Dr-Linux | justinu: we have VPN for the remote end |
| 21:51.03 | justinu | Dr-Linux: impossible to know without either enabling the logging on the sipura, or packet sniffing |
| 21:51.12 | nortex | terrapen, I use one at night with the ringall strategy for shop phones. |
| 21:51.20 | Dr-Linux | ooo |
| 21:51.23 | ManxPower | terrapen, there is no simplier way. there are many more complicated ways |
| 21:51.23 | terrapen | SuperDial macro may work too |
| 21:51.27 | ManxPower | I would call that a more complicated way |
| 21:51.36 | Dr-Linux | justinu: how can i check sipura's log...? |
| 21:51.51 | justinu | i think you need a syslog server setup, then you need to point the sipura2100 at that syslog server |
| 21:52.09 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 21:52.34 | Dr-Linux | justinu: i'll try first to give full access to sipura2100 |
| 21:52.59 | Dr-Linux | justinu: but strange problem is that, sometime it works sometime it doesn't :S |
| 21:53.09 | *** join/#asterisk Pazzo (n=thomas@host130-250-static.72-81-b.business.telecomitalia.it) |
| 21:54.10 | justinu | intermittant problems are very difficult to troubleshoot |
| 21:56.24 | Dr-Linux | ~dict intermittant |
| 21:56.51 | pdtmobile | ok, any pri studs in here, my braindead pri problem has come back today but I have logs this time! |
| 21:56.54 | pdtmobile | http://pastebin.ca/72485 |
| 21:57.06 | ManxPower | Well it is sunny and raining here. |
| 21:57.30 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 21:57.36 | Strom_C | pdtmobile: what's the issue? |
| 21:57.37 | pdtmobile | Outbound calls continue to work, inbound calls get congestion messages from the telco because our pbx tells the telco the channels the calls are coming in on are already in use |
| 21:57.58 | Strom_C | which end are outbound calls hunting from, and which end are inbound calls hunting from? |
| 21:58.16 | pdtmobile | inbound come in on the low end |
| 21:58.44 | pdtmobile | looks like outbound are doing the same |
| 21:58.49 | ManxPower | Well that answers half of the question |
| 21:58.56 | Strom_C | haha yes |
| 21:59.00 | pdtmobile | it's definitely not optimal |
| 21:59.01 | ManxPower | pdtmobile, make outbound start from the high end |
| 21:59.08 | terrapen | thanks, nortex, manxpower |
| 21:59.11 | pdtmobile | but! everything was working fine before going to 1.2.9.1, 1.2.6 |
| 21:59.12 | jsaunders | Anyone every come across SJPhone not answering incoming calls when you click Accept? Using latest asterisk and dial string like SIP/101&SIP/201&IAX2/301. |
| 21:59.15 | terrapen | this is not my usual asterisk installation |
| 21:59.27 | pdtmobile | (i will fix the stupid outbound trunk) |
| 21:59.38 | Strom_C | pdtmobile: it's still a bad idea to have inbound and outbound hunt from the same end of the PRI |
| 21:59.44 | pdtmobile | agreed |
| 21:59.44 | terrapen | building this for my father's bike stores and they want asterisk to act just like their crappy ancient AT&T Merlin system :) |
| 21:59.45 | ManxPower | pdtmobile, well use the older version |
| 21:59.46 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 21:59.49 | pdtmobile | but it should work itself out |
| 21:59.51 | *** part/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 21:59.56 | pdtmobile | ManxPower: i have exposed iax |
| 21:59.58 | Strom_C | pdtmobile: oh just fix the damned thing already |
| 21:59.59 | pdtmobile | not really gonna work |
| 22:00.14 | pdtmobile | Strom_C: I did :) |
| 22:00.15 | ManxPower | pdtmobile, Ah. |
| 22:00.28 | Strom_C | ok good - now come back if and when you encounter the same problem :) |
| 22:00.32 | ManxPower | Why not just use packet filtering to only allow connections from authorized places? |
| 22:00.40 | pdtmobile | ManxPower: because they move around |
| 22:00.52 | ManxPower | pdtmobile, it sucks to be you. |
| 22:01.00 | pdtmobile | yes I know |
| 22:01.30 | pdtmobile | but it doesn't change the fact that asterisk is going braindead on my pri |
| 22:01.50 | ManxPower | Client: There is no connectivity to the router I want you to configure. Can you talk me thru it over the phone. Me: I'll spend as many hours as you want me to. |
| 22:02.09 | ManxPower | Client then has a vision of the bill and decides to get connectivity to the router. |
| 22:02.12 | *** join/#asterisk saftsack (n=saftsack@p54A7DE7A.dip.t-dialin.net) |
| 22:02.25 | Strom_C | pdtmobile: so you're still having asterisk reject the inbound calls? |
| 22:02.36 | pdtmobile | oh, it goes away if I restart asterksi |
| 22:02.41 | pdtmobile | i already did that after I captured the debug |
| 22:02.48 | pdtmobile | only happens every couple of days |
| 22:03.07 | Strom_C | who is your telco and what kind of T1 card are you using |
| 22:03.11 | nortex | terrapen, I pulled one of those out a few weeks back, 72 ports. |
| 22:03.47 | pdtmobile | Expedius, TE110P |
| 22:04.50 | pdtmobile | sorry Xpedius |
| 22:04.54 | Strom_C | I can't find a company called "Expedius" but there is a telephone company called "XSpedius" - is that who you're referring ro? |
| 22:04.58 | pdtmobile | always get their name wrong |
| 22:05.07 | pdtmobile | hell, ya 3rd times the charm |
| 22:06.23 | Strom_C | what a stupid name for a company :) |
| 22:06.23 | pdtmobile | ya, horrible |
| 22:06.23 | Waverly360 | Maybe...but you'll probably remember it. |
| 22:06.23 | pdtmobile | i'm a customer and I can't remember it |
| 22:06.24 | Waverly360 | I didn't say you could spell it...just remember. |
| 22:07.39 | topping | hey guys, anyone using asterisk-jtapi? |
| 22:07.40 | nortex | Is there a way to disable the Polycom softkey for Buddies or My Stats? |
| 22:08.48 | ManxPower | nortex, see the polycom admin manual |
| 22:09.14 | mountainm2k | anybody recommend a good source for Polycom products? |
| 22:11.58 | pdtmobile | i swapped the hunt order, any ideas on things to look at if it happens again? |
| 22:16.35 | *** join/#asterisk ivanfm (n=ivanfm@c9068840.virtua.com.br) |
| 22:17.17 | *** part/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com) |
| 22:19.01 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 22:21.48 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 22:21.50 | MoutaPT | any one knows how to route calls based on MSNs instead of DID? |
| 22:21.54 | MoutaPT | BRI card |
| 22:23.08 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-146-200.dyn.sprint-hsd.net) |
| 22:25.48 | De_Mon | I'm trying to use VoIP from the bahamas and calls go out fine, but don't receive anything |
| 22:26.33 | c | bigging up the NAT traversal kru |
| 22:27.28 | *** join/#asterisk FoFiN (n=FoFiN@201.153.31.166) |
| 22:27.44 | FoFiN | hellow, how can i translate a call to another extension? |
| 22:29.25 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 22:31.02 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 22:31.21 | mrtwister | FoFiN: exten => Dial(Local/${EXTEN}@othercontext/n) |
| 22:31.42 | nortex | ManxPower, Can you tell me if the Polycom feature for call park can be configured to work with Asterisk? The Admin manula does not explain how to set what it does, just how to turn it on and off. |
| 22:31.56 | *** join/#asterisk pjchilds (i=pjchilds@pdpc/supporter/student/pjchilds) |
| 22:32.13 | *** join/#asterisk Sedorox (i=sedorox@smartserv/cna/Sedorox) |
| 22:32.52 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 22:33.10 | FoFiN | but i am in a active call |
| 22:33.21 | FoFiN | and i want that for example i can say ey wait a moment let me translate to X |
| 22:33.24 | FoFiN | to X Extension |
| 22:34.10 | nortex | FoFiN, Do you mean transfer to another extension? |
| 22:34.20 | FoFiN | yea |
| 22:34.28 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 22:34.33 | FoFiN | sorry my english is bad but i do what ever i can =) |
| 22:35.03 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 22:36.09 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 22:36.10 | FoFiN | and how can i transfer to another extension |
| 22:36.10 | FoFiN | ? |
| 22:36.42 | shmaltz | anybody seen any problems with queus when the members are Sip/sipaccount? |
| 22:38.58 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 22:39.11 | nortex | FoFiN, While in a call dial # and then the extension |
| 22:39.17 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 22:39.29 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 22:39.59 | FoFiN | tkz |
| 22:50.18 | CunningPike | EinsteinTaylor: Hey |
| 22:56.22 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 22:57.02 | rene- | where does monitor saves calls by default? |
| 22:57.55 | Bullseye_Network | rene-:/var/spool/asterisk/monitor |
| 22:58.01 | rene- | thanks thanks |
| 23:01.54 | *** join/#asterisk MoutaPT (n=MoutaPT@a83-132-239-109.cpe.netcabo.pt) |
| 23:04.14 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 23:04.15 | SpaceBass | hey folks |
| 23:04.21 | rene- | hey |
| 23:04.55 | SpaceBass | I was at a customer the other day that had a full blown cisco setup including webcams...if you called another use with a webcam, a video window opened up during the audio call |
| 23:05.22 | SpaceBass | since Apple's iChat is using sip for the video, is there a way to mimic that with ichat? |
| 23:05.42 | rene- | SpaceBass: Apple aint using sip for the video |
| 23:05.48 | rene- | Apple is not sip |
| 23:06.07 | SpaceBass | really? |
| 23:06.07 | rene- | you could use a video softphone that does sip like eyebeam |
| 23:06.09 | rene- | really |
| 23:06.17 | SpaceBass | pretty sure iChat was using SIP for the video, I know they are for the audio |
| 23:06.19 | rene- | eyebeam works with the apple camera |
| 23:06.27 | *** join/#asterisk rpm (i=russell@S010600111155e117.cg.shawcable.net) |
| 23:06.52 | SpaceBass | but if I use the softphone then I have to use it for the audio as well...as would the party being called |
| 23:06.56 | rene- | i think they can play with jabber now, so they can talk to other networks but even that way |
| 23:07.05 | rene- | you cant use the isight tru jabber in ichat |
| 23:07.08 | SpaceBass | what was so great about the cisco solution is that picking up the phone triggered the video |
| 23:07.10 | rene- | or you couldnt |
| 23:07.47 | SpaceBass | actually, im researching now...ichat uses SIP for the video too |
| 23:07.54 | rpm | anyone have experience working with centrex lines? im trying to get my fxo (tdm400p) to detect hang-ups faster and count rings, if a call comes in it will keep ringing for 5-10 seconds after the call has been hung-up |
| 23:07.58 | rene- | i am pretty sure iChat is closed but id love to be proved wrong |
| 23:08.13 | SpaceBass | now, it may be closed...im seeing some stuff about propritary headers, etc |
| 23:08.36 | Vorondil | hi all. quick question, i recently upgraded to the latest asterisk version on my freebsd 6.0-release machine. now i get "Ouch ... error while writing audio data: : Broken pipe" when starting asterisk =/ |
| 23:08.39 | Vorondil | any ideas? |
| 23:08.52 | *** join/#asterisk Meaw (n=debian@213.6.131.96) |
| 23:09.13 | SpaceBass | hummm.... never run asterisk on BSD, unless you count the OS X port |
| 23:09.34 | *** join/#asterisk dlynes_home (i=1000@S0106001217014b92.vc.shawcable.net) |
| 23:10.25 | Vorondil | SpaceBass: 1.2.7.1 ran like a charm before |
| 23:10.36 | rene- | asterisk runs fine on mac |
| 23:10.42 | dlynes_home | Vorondil: still does |
| 23:10.53 | SpaceBass | yeah |
| 23:10.53 | dlynes_home | Vorondil: i've got a couple of machines on 1.2.7.1 |
| 23:11.40 | Vorondil | dlynes_home: oh, i don't doubt it. ^_^ *i* have to be screwing it up somewhere, i'm sure. |
| 23:11.58 | dlynes_home | Vorondil: what seems to be the problem? |
| 23:12.16 | dlynes_home | Vorondil: I just got home...haven't seen the whole dialogue |
| 23:13.09 | Vorondil | dlynes_home: well, i did a 'portupgrade -R asterisk' to get up to 1.2.9.1 a little while ago |
| 23:13.16 | Vorondil | and now it barfs at runtime |
| 23:13.23 | dlynes_home | what does it barf out? |
| 23:13.37 | Vorondil | "Ouch ... error while writing audio data: : Broken pipe" |
| 23:13.38 | dlynes_home | i'm guessing you didn't upgrade asterisk-addons as well |
| 23:13.51 | dlynes_home | Vorondil: when you're trying to do what? |
| 23:14.01 | dlynes_home | Vorondil: moh? |
| 23:14.08 | Vorondil | just 'asterisk -d' |
| 23:14.22 | dlynes_home | Vorondil: try this instead: asterisk -vvvvvvvvvvvvvvvvg |
| 23:14.24 | rene- | Vorondil: you probably have some musiconhold process running? |
| 23:14.36 | dlynes_home | Vorondil: and then let's see what happens |
| 23:14.46 | rene- | do a ps aux | grep asterisk and kill everything named asterisk before starting the new version |
| 23:14.56 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 23:14.58 | kink0 | hello |
| 23:15.12 | dlynes_home | rene-: not to mention permanently deleting anything resembling mpg123 |
| 23:15.32 | kink0 | was in any way "channel_find_locked: Avoided deadlock" warning fixed ? |
| 23:15.46 | dlynes_home | kink0: that's not a bug |
| 23:15.58 | dlynes_home | kink0: unless you get it repeated over and over again |
| 23:16.08 | dlynes_home | kink0: if you just see it once in a while, that's by design |
| 23:16.18 | kink0 | dlynes_home, no ? what is ? I read it about old versions of Asterisk, but I am ussing 1.2.9.1 and sometimes I got that |
| 23:16.22 | sparkleytone | anyone have a grandstream gxp-2000? |
| 23:16.36 | dlynes_home | kink0: yeah...if there's a potential deadlock, you'll see that |
| 23:16.44 | dlynes_home | kink0: it's just telling you that it's avoiding it |
| 23:16.56 | dlynes_home | kink0: iow, it's doing as its told...making asterisk safe |
| 23:17.01 | Vorondil | wow, is pastebin slow today? |
| 23:17.07 | kink0 | dlynes_home, I got that every time I do show channels |
| 23:17.26 | dlynes_home | kink0: what kind of processor are you using? |
| 23:17.28 | kink0 | until Asterisk crashes or is restarted |
| 23:17.51 | kink0 | dlynes_home, is in a Dual xeon 2.8GHz/512 |
| 23:18.05 | dlynes_home | kink0: do you have hyperthreading enabled? |
| 23:18.14 | kink0 | dlynes_home, yes, I have it |
| 23:18.20 | dlynes_home | kink0: disable it |
| 23:18.26 | dlynes_home | kink0: it's not compatible with asterisk |
| 23:18.26 | Vorondil | rene-: did that |
| 23:18.59 | dlynes_home | kink0: besides...when you've got an SMP system, hyperthreading is kinda pointless |
| 23:19.09 | kink0 | dlynes_home, would be coding/decoding performance affected if I dissable hyperthreading in Xeon's ? I am ussing g729 from digium |
| 23:19.16 | dlynes_home | Vorondil: are you trying to use pastebin.ca? |
| 23:19.36 | dlynes_home | kink0: no...if anything, everything will work much more smooth |
| 23:19.46 | dlynes_home | kink0: like i said...hyperthreading is not compatible with asterisk |
| 23:19.52 | dlynes_home | kink0: it's been a known issue for some time now |
| 23:20.21 | Vorondil | dlynes_home: no, .com |
| 23:20.29 | Un1x | bah im bored |
| 23:20.30 | kink0 | ahh ok !! I will try so, yes I had read about, but I did not relations with actual problems , I will try your suggestions |
| 23:20.31 | dlynes_home | kink0: hyperthreading is not exactly a great thing, anyways...it's virtualizing a cpu into two virtual cpus |
| 23:20.50 | dlynes_home | kink0: so effectively on a dual cpu machine, you have four virtual cpus |
| 23:21.01 | dlynes_home | kink0: virtualizing anything is never a good idea, imho |
| 23:21.01 | [hC] | dlynes_home: david yat sin replied pretty quick, thank god. |
| 23:21.09 | dlynes_home | [hC]: yeah...he's pretty cool |
| 23:21.09 | kink0 | yes, I know, I agree |
| 23:21.16 | Vorondil | dlynes_home: screw this... http://rafb.net/paste/results/ol5tf688.html :D |
| 23:21.25 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 23:21.29 | [hC] | dlynes_home: apparently i need to attach the molex power adapter when using >4 fxo modules. didnt know that. |
| 23:21.33 | *** join/#asterisk rnovotny22 (n=rnovonty@198.57.19.126) |
| 23:21.39 | dlynes_home | [hC]: really? |
| 23:21.40 | Un1x | lmao |
| 23:21.41 | kink0 | but I was supossing to save/restore stack on CPU was a better that use the half of virtual CPU's...or so |
| 23:21.42 | [hC] | now, i must try to get hudlite to function. |
| 23:21.48 | [hC] | dlynes_home: yeah, that was his suggestion... |
| 23:22.02 | dlynes_home | [hC]: you mean that four conductor power that you need for the FXS ports? |
| 23:22.49 | dlynes_home | [hC]: i've heard people mention molex on here, but i never knew what it was |
| 23:23.16 | dlynes_home | [hC]: i'm guessing you mean the +5, -5, ground, ground connector, right? |
| 23:23.23 | rpm | i have two tdm400p's in an amd opteron dualcore 265. both with 4 fxo modules each.. periodically it completely crashes the machine with zaptel 1.2.6 and kernel 2.6.16.16.. can these devices not share an irq with anything else? |
| 23:23.47 | Un1x | rpm: you shouldn't have more then one card per machine |
| 23:23.52 | Un1x | read the website.... |
| 23:24.10 | [hC] | dlynes_home: yeah. the power adapter for fxs.. |
| 23:24.13 | Un1x | if you want that many fxo modules/fxs |
| 23:24.19 | Un1x | you should get, the TDM2400P |
| 23:24.26 | drray | get a channel bank |
| 23:24.51 | dlynes_home | Vorondil: hrm...i wonder... |
| 23:24.54 | dlynes_home | [chan_h323.so]Jun 26 19:17:59 WARNING[54365]: loader.c:325 __load_resource: Shared object "libldap-2.2.so.7" not found, required by "libh323_r.so.1.18.0" |
| 23:24.54 | dlynes_home | Jun 26 19:17:59 WARNING[54365]: loader.c:554 load_modules: Loading module chan_h323.so failed! |
| 23:24.54 | dlynes_home | Ouch ... error while writing audio data: : Broken pipe |
| 23:25.00 | rpm | i have a tdm2400p the card is too long for this 2U case. i wish these guys would just get a pri |
| 23:25.08 | dlynes_home | Vorondil: you think the libldap/chan_323 might be related? |
| 23:25.12 | dlynes_home | Vorondil: nah....couldn't be |
| 23:25.12 | [hC] | any of you guys get the hudlite software working standalone? (ie NOT in trixbox) |
| 23:25.54 | dlynes_home | rpm: stop making sense |
| 23:26.02 | dlynes_home | rpm: it's not allowed when you're dealing with customers |
| 23:26.14 | jsaunders | What's the ultimate iax2 ata? |
| 23:26.25 | dlynes_home | jsaunders: iaxy? the only iax2 ata? |
| 23:26.39 | jsaunders | iaxy you say, does not ring a bell. |
| 23:26.40 | Vorondil | dlynes_home: i suppose that's my next stab at it, but it didn't think not being able to load h323 would matter since i'm not using it at all |
| 23:26.51 | Un1x | rpm 2u is roughly equivalant to a, Mid size tower... |
| 23:26.53 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 23:26.57 | Un1x | and those fit perfectly nice in them |
| 23:27.04 | dlynes_home | jsaunders: probably www.digium.com/products/iaxy/ or something similar |
| 23:27.06 | ManxPower | Anyone here played with the M() option to Dial? |
| 23:27.11 | dlynes_home | jsaunders: it's that ata that digium sells |
| 23:27.15 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 23:27.34 | drray | iax powered ata |
| 23:27.36 | jsaunders | And... once again... go dylnes. |
| 23:27.46 | dlynes_home | ? |
| 23:28.04 | dlynes_home | Un1x: digium cards fit in 2U's now? |
| 23:28.11 | dlynes_home | Un1x: i thought they only fit in 3U's? |
| 23:28.14 | rnovotny22 | Config problem...getting busy tone on incoming sip call from BroadVoice. |
| 23:28.27 | jsaunders | You're always there to offer good advice, so... Go dylnes. Ya know, keep on keepin' on. |
| 23:28.29 | Un1x | err yea |
| 23:28.31 | kink0 | good nigh all |
| 23:28.32 | Un1x | im sorry mistake |
| 23:28.37 | kink0 | dlynes_home, thanks !!! |
| 23:28.37 | dlynes_home | jsaunders: ah...i thought you meant go away :p |
| 23:28.41 | Un1x | mid tower is some where in between 2u and 3u |
| 23:28.45 | jsaunders | Never |
| 23:28.46 | Un1x | forget exactly where :P |
| 23:28.50 | jsaunders | I am a kind soul. |
| 23:29.07 | jsaunders | Unless you were rude like a certain member of the null team. Heh. |
| 23:29.12 | dlynes_home | null team? |
| 23:29.20 | *** join/#asterisk w0rmzw3rth (n=knoppix@ool-18b8812e.dyn.optonline.net) |
| 23:29.21 | *** join/#asterisk tgrman (n=jcmoore@picard.ojc.nuvio.com) |
| 23:29.25 | w0rmzw3rth | I want to be able to have at least 20-30 phone lines on my Asterisk Box , so people can dial into my asterisk box and make phone calls from it , any idea on a cheap set up , with tier 1 |
| 23:29.45 | dlynes_home | w0rmzw3rth: two pri's? |
| 23:29.54 | dlynes_home | w0rmzw3rth: or even one pri? |
| 23:29.58 | Un1x | w0rmzw3rtg: where you getting that many PSTN lnes? |
| 23:30.04 | w0rmzw3rth | dlynes_home what do you mean |
| 23:30.19 | dlynes_home | w0rmzw3rth: cheaper to go with a pri, than going with 23 phone lines |
| 23:30.23 | jsaunders | pri |
| 23:30.34 | w0rmzw3rth | what is pri? |
| 23:30.35 | dlynes_home | w0rmzw3rth: go with two pri's, you'll have 46 channels |
| 23:30.36 | jsaunders | Which I've recently had great success w/. |
| 23:30.38 | jsaunders | T1 |
| 23:30.42 | jsaunders | If in North America. |
| 23:30.48 | w0rmzw3rth | where do I get pri from? |
| 23:30.51 | dlynes_home | w0rmzw3rth: primary rate ISDN T1 (E1 in Europe, and South America) |
| 23:30.53 | jsaunders | telco |
| 23:30.58 | dlynes_home | w0rmzw3rth: your ilec |
| 23:31.25 | w0rmzw3rth | I live in long island where could I find a company that does that? covad? |
| 23:31.47 | dlynes_home | w0rmzw3rth: also, if you're using sangoma pri cards, you can fractionalize your pri so that you can use whatever you're not using for phone lines to deliver 10Mbps Internet |
| 23:31.51 | jsaunders | I thought it was primary rate interface. ;) |
| 23:32.02 | dlynes_home | well....up to 10Mbps Internet |
| 23:32.10 | dlynes_home | jsaunders: i've heard both, actually |
| 23:32.23 | jsaunders | So have I. Just going off web defs. |
| 23:32.25 | w0rmzw3rth | niice |
| 23:32.34 | w0rmzw3rth | I just have to find a carrier in nyc |
| 23:32.46 | dlynes_home | jsaunders: it is an ISDN interface (just 23 channels, instead of 2) |
| 23:32.50 | w0rmzw3rth | do they assign a block of ip address? |
| 23:32.59 | dlynes_home | w0rmzw3rth: depends on your carrier |
| 23:33.14 | dlynes_home | w0rmzw3rth: but on mine, i'm getting 125 IP addresses |
| 23:33.19 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 23:33.30 | Dr-Linux | jsaunders: it's not wrong if you say "primary rate interface" |
| 23:33.30 | jsaunders | I'm quite aware of that Mr. Dylnes. Heh. |
| 23:33.39 | w0rmzw3rth | what kind of hardware would I need o hook the pri to asterisk box |
| 23:33.50 | Un1x | w0rmzw3rth: you want people to be able to dail and make calls correct, so where you getting all those Phone lines from? |
| 23:34.05 | jsaunders | Did you forget I was in here a few days ago inquiring why my b channels where constantly moving data, even whilst on hook? |
| 23:34.10 | w0rmzw3rth | and lets say I want a 5 line pbx box would I be able to purchase 5 lines from voicepulse.com and just assign them on my asterisk box? |
| 23:34.15 | Vorondil | dlynes_home: wtf: http://pastebin.ca/72563 |
| 23:34.23 | Dr-Linux | w0rmzw3rth: get TE110P digium card for 1 PRI line |
| 23:34.29 | dlynes_home | w0rmzw3rth: Digium te110p, te210p, te410p, te405p, or sangoma a101u, a102u |
| 23:34.57 | jsaunders | a104 |
| 23:35.06 | dlynes_home | yeah...or a104d |
| 23:35.07 | w0rmzw3rth | if I already have an existing internet connection? |
| 23:35.07 | jsaunders | :D |
| 23:35.20 | dlynes_home | w0rmzw3rth: ditch your internet connection :) |
| 23:35.25 | w0rmzw3rth | and lets say I want a 5 line pbx box would I be able to purchase 5 lines from voicepulse.com and just assign them on my asterisk box? |
| 23:35.26 | Dr-Linux | w0rmzw3rth: what you have T1 or E1 ? |
| 23:35.28 | dlynes_home | w0rmzw3rth: if you get two pri's, you won't need it anymore |
| 23:35.39 | dlynes_home | w0rmzw3rth: but you need sangoma hardware to be able to take advantage of that |
| 23:35.48 | w0rmzw3rth | via AIX? |
| 23:35.48 | mountainm2k | So, the standard advice is only one card per box -- can I put in a single-span T1 board and an 400-board? |
| 23:35.52 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 23:35.53 | dlynes_home | w0rmzw3rth: or go with an adtran850, some channel banks, and some other crap |
| 23:36.06 | drray | adit600 |
| 23:36.10 | dlynes_home | w0rmzw3rth: frankly, i think the sangoma card is cheaper |
| 23:36.26 | w0rmzw3rth | so I should get PRI and I don't need a tier 1 |
| 23:36.28 | Un1x | someone tell me where i can get some cheap PSTN lines, from in toronto.... |
| 23:36.38 | Un1x | instead of going with bell or rogers |
| 23:36.39 | Un1x | lmao |
| 23:36.42 | dlynes_home | Vorondil: i'm stumped....you need to ask that question on #freebsd |
| 23:36.51 | *** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen) |
| 23:36.55 | dlynes_home | Un1x: Telus? |
| 23:36.58 | Vorondil | dlynes_home: hmm, alrighty |
| 23:37.00 | dlynes_home | Un1x: Sprint? |
| 23:37.04 | Un1x | how much is telus dlynes_home |
| 23:37.06 | Un1x | rogers owns srint |
| 23:37.11 | dlynes_home | Un1x: erm...wait....sprint is rogers now :p |
| 23:37.13 | Un1x | *sprint they bought it rember |
| 23:37.35 | dlynes_home | Un1x: telus only charges for your first born |
| 23:37.43 | Un1x | lmao |
| 23:37.51 | dlynes_home | Un1x: and gives you ivr hell if you want to report a problem with your phone line |
| 23:38.09 | w0rmzw3rth | thnx |
| 23:38.12 | Un1x | comon dlynes im serious i need pstn lines |
| 23:38.30 | dlynes_home | look in the yellow pages under local phone lines |
| 23:38.45 | dlynes_home | you're asking someone in Vancouver what phone companies there are in toronto |
| 23:38.50 | dlynes_home | that makes too much sense |
| 23:39.24 | dlynes_home | is this for a business? |
| 23:39.32 | dlynes_home | if it is, you could try group telecom, too |
| 23:39.42 | dlynes_home | but i don't think they're available in T.O. |
| 23:39.46 | Un1x | nah personal usage for testing at the momemnt |
| 23:39.50 | Un1x | i se |
| 23:39.55 | dlynes_home | and besides |
| 23:39.57 | dlynes_home | they're owned by Bell |
| 23:40.08 | dlynes_home | Bell bought them last year |
| 23:40.13 | dlynes_home | oh |
| 23:40.17 | dlynes_home | actually you could try Allstream |
| 23:40.33 | file | Allstream is... AT&T? |
| 23:40.36 | dlynes_home | and i think you guys might have Aliant back east |
| 23:40.39 | dlynes_home | file: correct |
| 23:40.46 | file | Aliant is Atlantic Canada exclusively |
| 23:40.56 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 23:41.08 | dlynes_home | file: yeah...I got a nastygram from telus on behalf of telus, aliant and bell today |
| 23:41.17 | Un1x | lol |
| 23:41.23 | dlynes_home | file: they want us to start charging fees now, too :(( |
| 23:41.31 | dlynes_home | file: so they've filed a petition to the crtc |
| 23:41.33 | file | they sort of melded together |
| 23:41.44 | w0rmzw3rth | where can find a PRI service in Long island NY ? |
| 23:41.49 | CunningPike | dlynes_home: What sort of fees? |
| 23:42.01 | file | Telus Mobility uses the Aliant cellular network in Atlantic Canada, and Aliant is peered in Saint John with Bell for all internet traffic |
| 23:42.19 | dlynes_home | file: i guess they got jealous of the us, because the fcc just passed a resolution that all mobile carriers, pstn carriers and voip carriers must charge their customers the same user access fees |
| 23:42.22 | CunningPike | w0rmzw3rth: Your local telco(s).......... |
| 23:42.42 | dlynes_home | CunningPike: non-specific so far, but i'm guessing it'll be similar to what the fcc just passed |
| 23:42.43 | Un1x | does us fcc law concurr with canadian law? |
| 23:43.03 | dlynes_home | Un1x: no, but canadian law usually follows suit with a popular law, and vice-versa |
| 23:43.10 | Un1x | ahh |
| 23:43.15 | CunningPike | dlynes_home: So, 911 fee etc? |
| 23:43.25 | dlynes_home | CunningPike: nah...more like tariffs |
| 23:43.55 | dlynes_home | CunningPike: based on the amount of traffic you're sending over the wire |
| 23:44.30 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 23:44.38 | dlynes_home | so basically what it amounts to i think is that voip won't be any cheaper |
| 23:44.45 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 23:44.48 | CunningPike | dlynes_home: That'll be an interesting one - I wonder how they will define the difference between you and Skype for that purpose |
| 23:45.11 | dlynes_home | i would imagine with something like that, it'll cover skype, too |
| 23:45.18 | ManxPower | It seems that if I use the M() option to Dial ANY digit will exit out of the macro |
| 23:45.20 | dlynes_home | so skype will have to start paying tariffs as well |
| 23:45.36 | CunningPike | dlynes_home: Impossible - it's just packets on the Internet |
| 23:45.50 | PakiPenguin | Dr-Linux, bad bad :p |
| 23:46.09 | drray | packets that need to arrive within 500ms to be usefull |
| 23:46.13 | Dr-Linux | PakiPenguin: what happend dude? |
| 23:46.14 | Un1x | wait how does a pakistani person become a penguin,... |
| 23:46.22 | Un1x | isn't pakistan located in asia, sout asia to be exact |
| 23:46.28 | Un1x | where temperatures are around 40+ |
| 23:46.34 | dlynes_home | Un1x: ummmm |
| 23:46.37 | Un1x | meaning snow would be impossible |
| 23:46.38 | Un1x | :P |
| 23:46.42 | dlynes_home | Un1x: they get snow there, too |
| 23:46.49 | Un1x | wlel in mountain areas |
| 23:46.54 | Un1x | but, not in all around |
| 23:46.57 | Un1x | mountains meaning |
| 23:47.01 | Un1x | the himalayan area |
| 23:47.01 | CunningPike | drray: So they will slow down all Internet traffic unless you pay as an ITSP? |
| 23:47.02 | dlynes_home | Un1x: the Punjab gets snow in the winter |
| 23:47.10 | dlynes_home | Un1x: and they're not up in the mountains |
| 23:47.13 | Un1x | who says this? |
| 23:47.25 | drray | CunningPike, would you put that past them? |
| 23:47.26 | dlynes_home | Un1x: ummm...my boss? He's from there :) |
| 23:47.41 | CunningPike | drray: Not entirely...... |
| 23:47.51 | Un1x | dlynes_home: google says different :p |
| 23:47.56 | dlynes_home | CunningPike: heh...you should be used to Telus' antics by now :) |
| 23:48.08 | dlynes_home | Un1x: which punjab are you looking at? east or west? |
| 23:48.16 | Un1x | hmm the one in pakistan |
| 23:48.17 | Un1x | lmao |
| 23:48.23 | dlynes_home | that's west punjab |
| 23:48.26 | dlynes_home | my boss is from east punjab |
| 23:48.31 | dlynes_home | that's in India |
| 23:48.32 | CunningPike | drray: But they will then have to put up with lousy performance for other streaming media |
| 23:48.52 | Un1x | well accordin to my knowledge from google, it says east pakistan is bangladesh now... |
| 23:48.56 | Dr-Linux | dlynes_home: Punjab only 4% area get snow .. |
| 23:49.00 | Un1x | oh |
| 23:49.01 | drray | from their multicast server that content providers will happily pay to use |
| 23:49.02 | Un1x | anyway i dont care |
| 23:49.07 | dlynes_home | Un1x: that's east pakistan, not east punjab |
| 23:49.09 | Un1x | lmao i get enough snow to worry about :P |
| 23:49.12 | ManxPower | The ruling ONLY COVERS CALLS THAT CONNECT TO THE PSTN |
| 23:49.16 | Dr-Linux | NWFP gets 80% though |
| 23:49.18 | dlynes_home | Dr-Linux: still 4% :) |
| 23:49.20 | drray | so google only has to pay once to route traffic to comcast |
| 23:49.22 | ManxPower | Why is that so hard to understand? |
| 23:49.33 | dlynes_home | ManxPower: ah |
| 23:49.35 | Dr-Linux | dlynes_home: i'm from NWFP |
| 23:49.40 | Un1x | NWFP? |
| 23:49.51 | CunningPike | ManxPower: Ah - that makes more sense. You putting it in ALL CAPS MADE ALL THE DIFFERENCE. Thanks |
| 23:49.55 | dlynes_home | northwest fscking pakistan |
| 23:50.17 | Un1x | why is there a F in therE? |
| 23:50.20 | Dr-Linux | North West Frontier Provence = NWFP |
| 23:50.21 | Un1x | for fucking lol? |
| 23:50.23 | Un1x | ahh |
| 23:50.32 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 23:50.43 | dlynes_home | Un1x: your nick is unix, but you don't know what an fsck is? |
| 23:50.52 | Un1x | i do but i thought it ment something else |
| 23:50.56 | Un1x | with the phrase u where using |
| 23:51.33 | Un1x | fsck is the hdd check util if u constantly shut ur comp off improperly you have to run it |
| 23:51.36 | Un1x | well not have to |
| 23:51.37 | dlynes_home | Un1x: well, i'm not as rude as you |
| 23:51.39 | Un1x | but it's good idea to |
| 23:51.46 | CunningPike | fsck is a contraction of "For fuck's sakes", usually uttered by sysadmins when they lose sectors on their hard disks |
| 23:51.56 | Un1x | lol |
| 23:52.02 | orlock | heh |
| 23:52.09 | Un1x | i thought it was ffs :P |
| 23:52.19 | dlynes_home | Un1x: you mean fys? |
| 23:52.24 | Un1x | no FFS! |
| 23:52.25 | dlynes_home | erm gfys? |
| 23:52.26 | CunningPike | -bash: ffs: command not found |
| 23:52.28 | CunningPike | Nope |
| 23:52.29 | Un1x | lol |
| 23:52.33 | *** part/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 23:52.39 | dlynes_home | go fsck yourself :p |
| 23:52.40 | Dr-Linux | fsck, recover the lost sectors, and places them correctly |
| 23:52.58 | Dr-Linux | dlynes_home: twice i did :P |
| 23:52.58 | Un1x | he system utility fsck (for "file system check" or "file system consistency check") is a tool for checking the consistency of a file system in the Unix system and clones thereof. |
| 23:52.59 | orlock | heh |
| 23:53.01 | orlock | hopefully |
| 23:53.07 | orlock | or it just throws them into lost+found |
| 23:53.07 | Un1x | AND CLONES |
| 23:53.09 | Un1x | meaing Linux... |
| 23:53.10 | Un1x | :P |
| 23:53.11 | Un1x | lmao |
| 23:53.23 | dlynes_home | Un1x: dood |
| 23:53.35 | dlynes_home | Un1x: give it up |
| 23:53.43 | Un1x | bah l |
| 23:53.54 | CunningPike | As orlock says, fsck is actually a symlink to 'mv $FILE /lost+found |
| 23:54.31 | ManxPower | So nobody here has used the M() option to Dial? |
| 23:54.33 | orlock | then you get to play "guess what vital file i am" on the randomly numbered chunks you get left with |
| 23:54.46 | drray | if it was vital you'd have backed it up |
| 23:55.10 | CunningPike | drray: ;) |
| 23:55.33 | CunningPike | ManxPower: Not I |
| 23:55.50 | orlock | drray: ok, back up vmlinuz and see if it makes any difference after the system goes down hard and it cant be found :) |
| 23:55.56 | Meaw | does asterisk works with freeradius? |
| 23:56.08 | Meaw | for the accounting ..etc |
| 23:56.11 | dlynes_home | Meaw: yeah...i've seen a few people using it with freeradius |
| 23:56.29 | drray | I find it shocking that google can run an enterprise with cheap POS commodity boxes but some fat guy in his mothersbasement can't survive a fsck |
| 23:56.35 | Meaw | is there turotal howto ?' |
| 23:56.59 | dlynes_home | Meaw: probably....try looking for radius setup on the asterisk wiki |
| 23:57.04 | dlynes_home | ~wikis |
| 23:57.11 | jbot | it has been said that wikis is http://www.voip-info.org |
| 23:57.32 | CunningPike | ~thewiki |
| 23:57.33 | jbot | i heard thewiki is at http://www.voip-info.org/wiki-Asterisk |
| 23:57.34 | orlock | thats not my parents basement! |
| 23:57.36 | Meaw | i have freeradius installed with mysql pppoe-server |
| 23:57.36 | tzafrir_laptop | I figure you mean the ext fsck |
| 23:57.44 | Meaw | ok ty |
| 23:57.59 | tzafrir_laptop | reiserfs's one is actually even worse. never tried the xfs tool |
| 23:58.40 | *** join/#asterisk bkw__ (n=brian@asterisk/friend-and-developer/bkw) |
| 23:58.47 | *** join/#asterisk smackus (n=smackus@c-67-169-248-217.hsd1.ut.comcast.net) |
| 23:58.47 | orlock | i had reiser crap out on me bigtime |
| 23:59.05 | drray | you had your hardware underneath crap out on you bigtime |
| 23:59.09 | orlock | but thatwas years ago. i have had ext3 crap out badly recently as well |
| 23:59.14 | orlock | drray: shut the fuck up |
| 23:59.19 | orlock | the raid5 array was fine |
| 23:59.25 | orlock | the redundant psu's were fine. |
| 23:59.33 | orlock | the 3ware card said everything was fine. |
| 23:59.34 | orlock | :P |
| 23:59.48 | *** join/#asterisk trig_hm (i=jason@home.monkeypr0n.org) |
| 23:59.49 | Un1x | olock: no need to curse in here.. |