| 00:00.05 | [TK]D-Fender | thats a constant the causes CALLERID() to return the number or set the number kind of like an array value |
| 00:00.11 | [TK]D-Fender | yup, exactly like that |
| 00:01.33 | *** join/#asterisk cjk (n=cjk@11.121.9.213.dsl.getacom.de) |
| 00:02.13 | cjk | hi, can anyone explain me the channel structure, in cdr for example? i have IAX2/$PART1@$PART2-xyz |
| 00:02.20 | cjk | normally part1 and part2 are the same |
| 00:02.24 | cjk | but not always |
| 00:02.29 | cjk | what do they represent? |
| 00:02.44 | *** join/#asterisk _Sam-- (n=sam@phone2.kneedraggers.com) |
| 00:02.52 | cjk | and how can $PART1 be a peer on my system and $PART2 another one? |
| 00:02.55 | marcus2 | http://pastebin.ca/30152 |
| 00:03.02 | Utah_Dave | marcus2: thanks |
| 00:03.04 | _Sam-- | does anyone know what happened to app_chanspy for 1.0 versions, or what I could use to do the same thing? |
| 00:03.13 | marcus2 | why doesnt that work the way i expect it to? |
| 00:03.42 | marcus2 | i want it to dial pap1_line1, and then pap1_line2 if the first one is busy |
| 00:04.00 | marcus2 | i guess theres a better way of doing this with 1.2, anyway? |
| 00:04.42 | dudes | n+101 |
| 00:04.45 | [TK]D-Fender | Did you diable call waiting on the PAPs's? |
| 00:04.55 | [TK]D-Fender | disable.... |
| 00:04.56 | marcus2 | yeah i did |
| 00:05.00 | [TK]D-Fender | blarg.. can't type today... |
| 00:05.02 | [TK]D-Fender | hmmm |
| 00:05.18 | [TK]D-Fender | Should work, though I might put a dial limit on those. |
| 00:06.02 | marcus2 | http://pastebin.ca/30155 |
| 00:06.16 | marcus2 | thats the cli output i get, when i dial 2525 and pap1_line1 is busy |
| 00:06.33 | marcus2 | it doesnt even try line2 |
| 00:06.47 | marcus2 | and i get an immediate busy signal |
| 00:07.52 | [TK]D-Fender | Maybe you didn't enable priority jumping properly. that is something that is also new to 1.2 |
| 00:08.07 | marcus2 | is there a better way to do this in 1.2 ? |
| 00:08.19 | [TK]D-Fender | it may jsut fall through to 2 and not jump but leave you the dialstatus variable to process. |
| 00:08.29 | marcus2 | ah right, thats what i'm thinking of |
| 00:08.39 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
| 00:10.30 | [TK]D-Fender | take a look at this too http://voip-info.org/wiki/view/Functions |
| 00:12.33 | *** join/#asterisk cnet2 (n=jjohn@adslnat-sanjose-4.ice.co.cr) |
| 00:13.52 | cnet2 | anyone know if I can create a connection between 2 Asterisk Servers, so that just 1 has the public IP. But the receiving server is behind NAT. I was thinking like creating a connection, from receiving Server to Sending server and using that connection. |
| 00:14.33 | [TK]D-Fender | IAX2 works fairly well in that scenario.... |
| 00:15.13 | cnet2 | even if the receiving server does not have a public IP?.. i looked over the wiki, but only found examples needing receving server having public ip |
| 00:15.43 | [TK]D-Fender | You can work * in a double-NAT scenario if you set it up right |
| 00:15.52 | [TK]D-Fender | You jsut need the right ports forwarded. |
| 00:15.53 | cnet2 | cool |
| 00:16.02 | cnet2 | ohh.. (i have no control over the NAT |
| 00:16.07 | cnet2 | the ISP does that |
| 00:16.13 | [TK]D-Fender | ?! |
| 00:16.23 | [TK]D-Fender | You don't control your own routing? |
| 00:16.49 | cnet2 | nop, see our ISP gives us a Public IP, i don't know how it manages the routing |
| 00:16.55 | cnet2 | sorry |
| 00:16.58 | cnet2 | gives us a Private IP |
| 00:17.40 | [TK]D-Fender | wierd |
| 00:17.49 | *** join/#asterisk Cresl1n (n=matt@146.229.179.18) |
| 00:18.06 | cnet2 | yeah.. they suck |
| 00:19.58 | *** join/#asterisk test34 (i=1000@unaffiliated/test34) |
| 00:20.45 | test34 | I get a Syntax Error when I try to make install the zaptel driver (v1.2.0).. anybody else got this problem ? |
| 00:26.31 | test34 | /bin/sh: -c: line 0: syntax error near unexpected token `;' |
| 00:28.59 | _Sam-- | Does anyone know about ChanSpy and 1.0 versions? "As of March 24 2005 ChanSpy has been added into CVS HEAD |
| 00:29.55 | *** join/#asterisk docelm0 (n=docelmo@static-71-251-95-2.tampfl.fios.verizon.net) |
| 00:30.05 | docelm0 | ok C guys WTF does this mean? implicit declaration of function `pbx_builtin_setaccount' |
| 00:30.15 | wunderkin | Cresl1n? :) |
| 00:30.18 | docelm0 | cause damnit if it isnt driving me up the wall |
| 00:34.28 | *** join/#asterisk ST-3 (i=ser@dipsy.tch.org) |
| 00:36.25 | Druken | docelm0: is it an error or a warning? |
| 00:36.32 | docelm0 | warning |
| 00:36.39 | Druken | then who cares? :) |
| 00:36.54 | docelm0 | I dont want it to freak out asterisk.. but if you say so.. |
| 00:37.21 | Druken | hehe eventually you'll learn to ignore warnings... :) |
| 00:37.42 | docelm0 | dude.. Im so new to this.. I went out and bought a book specifically for programming C on linux |
| 00:38.05 | docelm0 | what does this mean? passing arg 2 of `strrchr' makes integer from pointer without a cast |
| 00:38.22 | docelm0 | What I am trying to do is take everything after a matching point |
| 00:39.01 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-14-207.cybersurf.com) |
| 00:39.21 | Druken | oh.. well, i'm no programmer.. so i'm of no use to ya |
| 00:39.28 | *** join/#asterisk Cybertoy (n=Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 00:39.38 | docelm0 | ie: 12349999999999 and use strchr to remove the 1234 |
| 00:39.42 | Druken | i just figured ya were all hyped up over a warning |
| 00:39.58 | docelm0 | yes.. I dont like warnings.. |
| 00:40.16 | docelm0 | they usually mean bad things can happen when you dont want them to |
| 00:41.18 | *** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com) |
| 00:42.19 | docelm0 | anyone in here know a little bout C that could explain to me how to make a function like substr in C++? |
| 00:43.21 | docelm0 | actually scratch that.. haha! |
| 00:43.40 | denon | docelm0: cant use strchr()? |
| 00:44.01 | docelm0 | doesnt matter found another answer to that problerm |
| 00:44.14 | denon | ic |
| 00:44.47 | docelm0 | actually scratch that.. I do need a way to strip it.. |
| 00:44.50 | docelm0 | crap |
| 00:45.18 | denon | sounds like you need a copy of K&R |
| 00:45.28 | docelm0 | K&R? |
| 00:46.07 | denon | http://www.amazon.com/gp/product/0131103628/103-7981222-0943034?v=glance&n=283155&v=glance |
| 00:46.41 | docelm0 | this is my delema.. warning: passing arg 2 of `strrchr' makes integer from pointer without a cast |
| 00:46.54 | docelm0 | hmmm |
| 00:49.07 | *** join/#asterisk frenzy (n=frenzy@193.220.82.108) |
| 00:49.20 | marcus2 | nifty, i rewrote macro-stdexten to take two arguements for devices to call, and call them in sequence |
| 00:49.33 | frenzy | hey folks.. |
| 00:49.53 | SwK | hmmmm |
| 00:49.56 | frenzy | am looking for billing solution for asterisk.. |
| 00:50.02 | frenzy | any pointers ? |
| 00:51.24 | Cybertoy | frenzy, http://astbill.com/whatis |
| 00:51.32 | file | dejavu |
| 00:51.36 | Cybertoy | I don't know anything about it myself though |
| 00:52.00 | frenzy | is that built around Drupal ? |
| 00:52.14 | Cybertoy | no idea |
| 00:53.41 | Cybertoy | seems so |
| 00:57.08 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 01:01.07 | *** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net) |
| 01:01.30 | YoMama | anyone here got the GXP-2000? |
| 01:01.59 | frenzy | grandstream ? |
| 01:02.02 | YoMama | yeah |
| 01:02.15 | frenzy | um i have other model.. |
| 01:02.18 | frenzy | how is the GXP ? |
| 01:03.19 | YoMama | frenzy: well, it seems alright...except when i dial..it sounds like it's swquaking |
| 01:03.30 | YoMama | squwaking |
| 01:03.33 | YoMama | however the hell u spell that word |
| 01:03.43 | frenzy | lol.. |
| 01:03.52 | frenzy | i guess all their models are 'cheap' |
| 01:03.56 | fugitivo | is that some kind of star wars droid sound? |
| 01:04.12 | Flauto | yomama, how are you doing |
| 01:04.27 | frenzy | more lke phone wars |
| 01:04.38 | YoMama | Flauto: good |
| 01:04.46 | YoMama | frenzy: it's fine when i call someone |
| 01:05.14 | frenzy | I know but it seems buggy-ish to me |
| 01:05.33 | YoMama | hmm...it might be EM issues |
| 01:05.49 | YoMama | i don't remember hearing this problem when i had the phone elsewhere |
| 01:06.17 | frenzy | try reseting it or maybe they got a new firmware |
| 01:06.25 | frenzy | I know they did have fix-es |
| 01:07.38 | *** join/#asterisk kiwnix (n=egarcia@82.158.157.222) |
| 01:08.00 | YoMama | it's running the firmware that's "current" |
| 01:08.03 | YoMama | it's not the beta.. .12 |
| 01:08.07 | YoMama | do u know their betatest URL? |
| 01:08.16 | YoMama | maybe i'll try .12 |
| 01:08.38 | frenzy | not on their site ? |
| 01:08.52 | *** join/#asterisk Junbug (n=Darline@69.182.228.58) |
| 01:08.59 | frenzy | mind you.. you cant revert back to the old version |
| 01:09.27 | YoMama | really...why not? |
| 01:09.55 | YoMama | ah..MWI works... |
| 01:10.08 | YoMama | hmm..when i call from my Zap...and call the extension...it says "asterisk" "asterisk" |
| 01:10.23 | YoMama | isn't there anyway to get it to identify the line..i don't have caller id so it shoudl say something like "No CAller ID" |
| 01:11.03 | denon | YoMama: there's a config for it |
| 01:11.21 | denon | either you can google for it, or I can |
| 01:11.23 | denon | guess I will |
| 01:12.03 | YoMama | extconfig.conf |
| 01:12.15 | Junbug | hmm ... anyway i can make * call a number while im sshd in to cli, (I have iax2 provider to call out) |
| 01:12.34 | Druken | YoMama: it should say asterisk |
| 01:12.35 | Druken | hehe |
| 01:12.46 | YoMama | Druken: well, what if i want it to say something else? |
| 01:12.56 | denon | YoMama: in [General] of sip.config, toss in a callerid=No Caller ID |
| 01:12.58 | Druken | then set it... |
| 01:13.02 | YoMama | I KNOW it's coming from the asterisk server..i want it to say something a bit more intelligent |
| 01:13.24 | denon | that config is the default |
| 01:13.31 | denon | to override the "asterisk" |
| 01:15.13 | *** part/#asterisk frenzy (n=frenzy@193.220.82.108) |
| 01:16.40 | YoMama | hmm |
| 01:16.50 | YoMama | ok..so it has two lines of the caller ID...that's sorta redundant |
| 01:17.01 | YoMama | one should be which channel it's coming in on..the other should be the actual caller id |
| 01:17.39 | YoMama | Anyoen got the GXP-2000? for some reason..it squwaks when i dial out..but then is fine when I'm talking to someone on it |
| 01:18.03 | YoMama | i thought it was an EM problem..but then why isn't the voice kinda garbled? |
| 01:20.14 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 01:23.07 | YoMama | aaah...version 13..no more raspy DTMF while dialing :) |
| 01:24.20 | test34 | I get a Syntax Error when I try to "make install" the zaptel driver (v1.2.0).. anybody else got this problem ? here's the exact error message-> http://pastebin.ca/30173 |
| 01:27.17 | *** join/#asterisk newmember (n=newmembe@70.72.189.149) |
| 01:28.34 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool139-176.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 01:30.20 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 01:36.06 | *** join/#asterisk marc324 (n=marc3234@206-248-135-52.dsl.teksavvy.com) |
| 01:40.00 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 01:45.12 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:47.04 | test34 | I can compile zaptel with libnewt-dev right ? |
| 01:48.44 | YoMama | are there any sip providers that pass along your caller ID info that you send? |
| 01:51.15 | test34 | with=without .. |
| 01:54.23 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 01:54.56 | YoMama | mog_work: u got a GXP-2000? |
| 01:55.12 | mog_home | nope |
| 01:57.08 | YoMama | damn |
| 01:57.40 | mog_home | i mostly just help people get things working |
| 01:58.39 | YoMama | i just got a GXP-2000 |
| 01:59.02 | YoMama | the bleeping thing doesn't understand daylight savings..it's got a radio button...but if you click it..it just +1's the time from GMT |
| 01:59.12 | YoMama | GMT-<your time zone>+1 |
| 01:59.16 | YoMama | what kinda crap is that? |
| 01:59.38 | mog_home | I see.... |
| 02:00.02 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost, -- if anyone is interested send me a private message (i might be afk) i have a reference in the channel if you need. -- |
| 02:00.17 | YoMama | bbz: sell 'em on ebay man |
| 02:04.20 | docelm0 | Say how old are the doxygen docs on asterisk's website? its telling me pbx_builtin_setaccount cant be found |
| 02:07.59 | Chotaire | morning all. I'd really like to discuss echo cancellation with someone (SIP). |
| 02:08.06 | Chotaire | anyone has played around with this? |
| 02:08.36 | mog_home | you dont |
| 02:08.40 | mog_home | you dont cancel ip |
| 02:08.46 | mog_home | anything digital should have no echo |
| 02:08.54 | mog_home | echo is only generated when there is analog |
| 02:09.02 | Chotaire | well, there is. |
| 02:09.04 | mog_home | so you have to echo cancel when it hits the tdm |
| 02:09.14 | mog_home | it is happening in your phone then |
| 02:09.22 | mog_home | as the headset is analog |
| 02:09.29 | mog_home | but it is not being created on the ip side |
| 02:10.12 | Chotaire | sorry mate, but they have no echo anywhere except when hitting asterisk. |
| 02:10.39 | mog_home | ^_^ whatever, I am just stating a simple fact |
| 02:10.44 | mog_home | when things are digital |
| 02:10.47 | mog_home | there is no echo |
| 02:10.50 | mog_home | you can ask anyone |
| 02:11.19 | mog_home | im not saying you dont have echo though |
| 02:11.36 | mog_home | what type of phones are you using |
| 02:11.38 | Chotaire | I have echo with several SIP callers, some of them using standard phones that work perfectly everywhere. |
| 02:11.55 | Chotaire | if I get echo with softphone callers, I consider it's their own problem. |
| 02:12.07 | Juggie | then the echo exists........ |
| 02:12.09 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 02:12.13 | Juggie | on your bridge to the pstn |
| 02:12.18 | Juggie | where ever that may be |
| 02:12.19 | mog_home | exactly |
| 02:12.22 | Chotaire | they call me through sip |
| 02:12.24 | mog_home | but not on the ip side |
| 02:12.35 | Chotaire | so they call a pstn phone number that gets routed to me via sip |
| 02:12.59 | Chotaire | it doesnt matter if they're in a meetme or if they call me directly.. the echo is always there. |
| 02:13.16 | Juggie | in the sip world |
| 02:13.24 | Chotaire | does sip.conf have any special echo features to play around with? although meetme uses zapata, I couldn't help it with e.g. echotraining and echocancel. |
| 02:13.25 | Juggie | er, ip world rather |
| 02:13.36 | Juggie | echo can only be generated at the end point |
| 02:13.46 | mog_home | no echo features exist for sip.conf |
| 02:13.48 | Juggie | eg, nothing in the middle, so long as the call remains ip, can generate echo |
| 02:13.52 | mog_home | you cant echo cancel an ip line |
| 02:13.59 | Chotaire | damn |
| 02:14.14 | Juggie | i sugest you trace the path of the call carefully |
| 02:14.26 | Juggie | because there would appear to be more going on then you think |
| 02:15.10 | Juggie | maybe the echo is your endpoint? : |
| 02:15.18 | Juggie | did you try a dif phone, or softphone |
| 02:15.18 | Chotaire | hehe |
| 02:15.19 | mog_home | ding ding ding |
| 02:15.25 | mog_home | we have a winner |
| 02:15.27 | Chotaire | like I say, I only have the echo with a few callers. |
| 02:15.33 | Chotaire | using several different phones. |
| 02:15.45 | Chotaire | from crappy softphones to well business telephones. |
| 02:15.48 | Juggie | you personally have used different phones? |
| 02:15.53 | Juggie | or they are using different phones |
| 02:15.55 | Chotaire | both |
| 02:15.59 | Chotaire | I have tons of phones to play with. |
| 02:16.29 | Juggie | well, one of the end points is generating the echo |
| 02:16.46 | Juggie | unless you have a wacky misconfigutation causing the call to be routed onto the pstn and back into ip |
| 02:16.49 | Juggie | without you noticing |
| 02:16.55 | Juggie | which i have seen :P |
| 02:17.15 | Chotaire | nah |
| 02:17.37 | Juggie | are the sip phones all peered with the same * box? |
| 02:17.49 | Chotaire | mine yes. |
| 02:18.01 | Chotaire | external phones not, they call through a pstn number and get routed to me via sip |
| 02:18.16 | Juggie | but you hear echo, on sip->sip calls |
| 02:18.18 | Chotaire | and no matter if I use internal or external phones, in meetme's I will always hear the same echo (so does all others) |
| 02:18.20 | Juggie | peered with the same * server |
| 02:18.27 | Chotaire | nah |
| 02:18.49 | Chotaire | ehrm yes.... but not with 2 phones locally connected to the * box |
| 02:19.00 | Juggie | ok |
| 02:19.02 | Juggie | your confusing me |
| 02:19.05 | Juggie | lets get one thing straight |
| 02:19.09 | Juggie | if a phone is sip |
| 02:19.14 | Juggie | but is routed to you through pstn |
| 02:19.18 | Juggie | it is no longer sip |
| 02:19.22 | Juggie | as far as i am concerned |
| 02:19.35 | Juggie | because you receive the call via pstn |
| 02:19.43 | Chotaire | eactly |
| 02:19.46 | Chotaire | exactly even. |
| 02:19.59 | Juggie | try joining two local sip phones into a meetme conference |
| 02:20.04 | Chotaire | not a problem. |
| 02:20.05 | Juggie | and see if there is echo |
| 02:20.16 | Juggie | so, when exactally is there a problem |
| 02:20.17 | Chotaire | i only have the problem with SOME external callers from pstn |
| 02:20.22 | Chotaire | all of them using the same telco |
| 02:20.32 | Juggie | so then the problem is not sip to sip calls at all |
| 02:20.48 | Juggie | even though u said it was, 5minutes ago |
| 02:21.04 | Chotaire | i have no chance besides cancelling at sip |
| 02:21.21 | Chotaire | there is NO pstn whatsoever connectivity involved on *MY* side. |
| 02:21.50 | Chotaire | so on the * box all callers will arrive using SIP |
| 02:22.02 | Juggie | alright |
| 02:22.09 | Juggie | so to understand and paraphrase what you said |
| 02:22.12 | Juggie | you do not control the pstn |
| 02:22.18 | Chotaire | yup |
| 02:22.19 | Juggie | you buy minutes over ip or something |
| 02:22.26 | Juggie | well then call your provider |
| 02:22.32 | Juggie | if you hear echo, its their problem |
| 02:22.35 | mog_home | who is your provider |
| 02:22.46 | Chotaire | a german sip provider that is |
| 02:23.02 | Chotaire | i could call them but they'd probably give a f.... |
| 02:23.10 | Chotaire | especially since this only happens with a few callers. |
| 02:23.10 | *** join/#asterisk kanobuho (n=chatzill@dsl-201-128-249-222.prod-infinitum.com.mx) |
| 02:23.21 | Sedorox | anyone know a voip provider that does 570 area code (Pennsylvania, USA) |
| 02:23.22 | Chotaire | they'd tell me to use their hardware and they would not support asterisk.. done. |
| 02:26.00 | *** join/#asterisk L|NUX (n=linux@202.141.252.82) |
| 02:26.39 | Chotaire | have there been any improvements on echo cancellation between 1.0.3 and 1.0.9? just wondering. |
| 02:26.59 | *** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net) |
| 02:27.07 | YoMama | ok..anyone got Grandstream stuff? |
| 02:27.12 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 02:27.50 | mog_home | nope |
| 02:27.54 | mog_home | but it wouldnt matter chotair |
| 02:28.01 | Sedorox | I got a Budgetone 100 |
| 02:28.05 | mog_home | you cant echo cancel non digital |
| 02:28.07 | mog_home | err |
| 02:28.18 | mog_home | you can only echo cancel analog signals |
| 02:28.45 | YoMama | Sedorox: i've heard you can't downgrade the firmware on them |
| 02:28.51 | marcus2 | is there a way to set a variable for all calls made to a certain context? |
| 02:29.03 | Chotaire | i just wonder, wouldn't it be theoretically possible to detect echo even on digital lines and suppress it? |
| 02:29.17 | Sedorox | dunno why you would want to |
| 02:29.20 | Chotaire | like detect that the same envelope is sent twice and cut it off? |
| 02:29.22 | Sedorox | the new firmware is better then the old |
| 02:29.30 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 02:29.31 | Chotaire | ...re-encode the digital stream... |
| 02:29.32 | Sedorox | but... its only good if you want a BASICCC phone.. and I mean basic |
| 02:31.36 | YoMama | Sedorox: well, i'm running 1.0.1.9 on my GXP-2000...there are betas at 1.0.1.13...but if it sucks..i wanna be able to downgrade |
| 02:32.19 | Sedorox | dunno... |
| 02:32.44 | Sedorox | Program-- 1.0.6.7 Bootloader-- 1.0.1.0 HTML-- 1.0.0.49 VOC-- 1.0.1.0 |
| 02:32.46 | Sedorox | on mine |
| 02:33.12 | Sedorox | but yea.. dunno if they are... |
| 02:34.31 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 02:36.44 | YoMama | Sedorox: hmm...oh well |
| 02:36.54 | *** join/#asterisk fugitivo (n=ajf@209.13.243.47) |
| 02:37.20 | *** join/#asterisk jcath (n=skycat@159.226.21.127) |
| 02:42.34 | jlewis | now that Set uses | to separate assignments from Set options, is there a way to assign a string containing | to a variable? I've tried various "obvious" methods of escaping it with no joy |
| 02:46.45 | *** join/#asterisk SubWolf (n=rob@24-117-112-237.cpe.cableone.net) |
| 02:51.21 | *** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) |
| 02:59.10 | *** join/#asterisk nahirean (n=nahirean@pool-70-111-77-58.nwrk.east.verizon.net) |
| 03:00.01 | *** part/#asterisk Cybertoy (n=Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 03:00.41 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 03:03.32 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 03:05.06 | *** join/#asterisk marc32422 (n=marc3234@206-248-156-208.dsl.teksavvy.com) |
| 03:06.39 | *** join/#asterisk tainted_ (n=identd@adsl-71-129-43-66.dsl.irvnca.pacbell.net) |
| 03:07.37 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 03:09.06 | wwalker | The Uniden web site says that the UIP1868P will only work with Packet8.net. Is that true? Is there a comparable ph9one on the market that isn't tied to a voip vendor? |
| 03:11.17 | kFuQ | that cordless/wifi phone ? |
| 03:11.25 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 03:13.07 | *** join/#asterisk santiago (n=santiago@208.195.215.160) |
| 03:14.41 | wwalker | yes |
| 03:15.51 | Qwell | It's like a fruit smoothie in a candy shell! |
| 03:16.06 | Qwell | yeah, these skittles suck |
| 03:16.07 | *** part/#asterisk santiago (n=santiago@208.195.215.160) |
| 03:17.57 | ptiggerdine | wwalker, if they can work with asterisk then that's a sweet phone. |
| 03:19.39 | *** join/#asterisk L|NUX (n=linux@202.141.252.82) |
| 03:20.36 | [TK]D-Fender | I hardly see the point of that phone.... too much in one box and its kinda ugly... |
| 03:20.41 | *** join/#asterisk L|NUX (n=linux@202.141.252.82) |
| 03:24.09 | *** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net) |
| 03:28.15 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 03:30.29 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 03:30.58 | ptiggerdine | have cordless VoIP is a good idea |
| 03:31.09 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 03:34.36 | [TK]D-Fender | Sipura + corless phone. Cheap, and reusable. |
| 03:35.24 | [TK]D-Fender | That uniden wonder loks like it'll be too smart for its own good. This is #asterisk after all not #iwantacheaptoasterlikeappliance |
| 03:36.01 | file[laptop] | toasters are good at... toasting... and stuff |
| 03:37.10 | [TK]D-Fender | I'm just looking forward to getting my gear and getting back whree I was a year ago. |
| 03:40.09 | Junbug | hmm ... anyway i can make * call a number while im sshd in to cli, (I have iax2 provider to call out) |
| 03:40.35 | *** join/#asterisk zamsler (n=zamsler@c-67-184-232-149.hsd1.il.comcast.net) |
| 03:40.50 | zamsler | bl@h |
| 03:44.45 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 03:47.46 | marcus2 | hrm |
| 03:47.46 | nahirean | anyone hear every worked with a provisioned sipura? i am trying to unlock it but I am not sure which voip provider provisioned it.. is there a way to bypass this silly password? |
| 03:47.54 | nahirean | here* |
| 03:47.55 | marcus2 | i find no good way to let an agent log off |
| 03:48.13 | marcus2 | other than having them enter a # twice after their pin |
| 03:48.23 | *** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net) |
| 03:48.31 | tainted_ | marcus2 how did the deployment go |
| 03:48.40 | marcus2 | its going |
| 03:48.44 | marcus2 | stil getting all of the pieces together |
| 03:48.48 | tainted_ | <EditItemTemplate> |
| 03:48.48 | tainted_ | <asp:TextBox ID="txtIncludedMinutes" runat="server" text='<%# Bind("IncludedMinutes") %>' CssClass="gridItem" Columns="8" /> |
| 03:48.48 | tainted_ | </EditItemTemplate> |
| 03:48.50 | marcus2 | i've moved into setting up call queues for support |
| 03:48.55 | tainted_ | sorry about that |
| 03:48.57 | tainted_ | any major fires? |
| 03:49.02 | marcus2 | nah, nothing huge |
| 03:49.13 | marcus2 | still struggling a bit with queues/agents tho |
| 03:49.25 | *** join/#asterisk diego_br (n=Diego@200-208-50-188-mns.cpe.vivax.com.br) |
| 03:56.43 | file[laptop] | kram: shhh go back into hiding |
| 03:58.09 | tainted_ | Nov 21 19:08:49 WARNING[29779]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 050723673804823e77465f017724c67e@66.249.128.195 for seqno 358 (Critical Request) |
| 03:58.11 | tainted_ | what is that? |
| 03:58.19 | *** join/#asterisk [hC] (n=hardcore@64.122.25.234) |
| 03:58.30 | file[laptop] | it tried to send a packet and got no response |
| 03:59.07 | tainted_ | strange thing is, there are no active channels |
| 03:59.22 | IronHelix | could be a registration or a qualify trying to go through and failing |
| 03:59.52 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 04:00.08 | tainted_ | IronHelix should i try a sip debug? |
| 04:00.15 | IronHelix | yeah go for it |
| 04:00.25 | IronHelix | if you have a ton of sip channels you may get tons of spam, not all of it which matters |
| 04:00.48 | tainted_ | i also have sip channels that refuses to go away after a call ends |
| 04:00.50 | IronHelix | but look at the timestamps on the debug msg's, one of them will come in at the same second as the max retries msg |
| 04:01.00 | IronHelix | tainted- probably part of the same problem |
| 04:01.04 | IronHelix | very possibly a NAT issue |
| 04:01.14 | tainted_ | no NAT here tho |
| 04:01.18 | IronHelix | in your sip debug look for nonroutable IPs (192.168 etc) |
| 04:01.21 | IronHelix | what about on the other side? |
| 04:01.34 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 04:01.39 | tainted_ | nope |
| 04:01.42 | tainted_ | all external |
| 04:02.16 | tainted_ | but it's sending REGISTER requests way too many times |
| 04:02.26 | tainted_ | about one every 5 seconds |
| 04:02.28 | IronHelix | are they getting denied? |
| 04:02.44 | marcus2 | hm looks like there are really crude ways to do automated agent logins/logouts |
| 04:03.19 | tainted_ | I can't tell |
| 04:03.30 | tainted_ | I see SIP/2.0 200 OK s |
| 04:03.38 | IronHelix | pastebin a few pages of it |
| 04:03.46 | tainted_ | k |
| 04:04.12 | IronHelix | remember, that for a register, you'll get a bunch of stuff- a register, then an ack, then a 403 denied, then another register (with cridentials this time), another ack, then an ok |
| 04:04.51 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 04:05.20 | tainted_ | i don't see any acks |
| 04:05.30 | Qwell | ack |
| 04:05.34 | RoyK | ~nickometer [TK]D-Fender |
| 04:05.39 | tainted_ | ok i got one from Qwell |
| 04:06.12 | tainted_ | RoyK lol |
| 04:06.27 | RoyK | :) |
| 04:06.44 | RoyK | ~nickometer tainted_ |
| 04:06.47 | marcus2 | is there some way to dump a list of all of the currently set variables at a certain point in the dialplan? |
| 04:06.48 | RoyK | ~nickometer RoyK |
| 04:07.12 | RoyK | pretty simple algorithm |
| 04:07.39 | RoyK | ~nickometer [1337]k1d |
| 04:07.48 | Qwell | if (!strcmp(v->name, "RoyK)) {value = "0.000";} else { /* do something */ } |
| 04:07.59 | file[laptop] | ooh |
| 04:08.00 | RoyK | Qwell: not really :) |
| 04:08.02 | file[laptop] | advanced! |
| 04:08.07 | RoyK | ~nickometer file |
| 04:08.10 | RoyK | ~nickometer file[laptop] |
| 04:08.21 | file[laptop] | :) |
| 04:08.25 | Damin | ~nickometer kram |
| 04:08.45 | kram | how is lameness measured? |
| 04:08.46 | mog_home | ~nickometer mogorman |
| 04:08.49 | kram | it's clearly mistaken :) |
| 04:09.10 | Damin | kram: Have 3 minutes for a -dev question? |
| 04:09.14 | Qwell | RoyK: back alley somewhere |
| 04:09.22 | kram | damin: sorta |
| 04:09.36 | Damin | kram: I'll float it.. You tell me A or B before I do the work.. |
| 04:09.42 | kram | okay |
| 04:09.52 | RoyK | Qwell: asdf = "0.000" seems a little odd |
| 04:09.59 | Qwell | RoyK: not if it's a string |
| 04:10.02 | Qwell | duh |
| 04:10.07 | RoyK | :) |
| 04:10.20 | RoyK | no such thing as a string in C :) |
| 04:10.27 | mog_home | char *.... |
| 04:10.28 | Damin | kram: Asterisk has a contrib/init.d directory w/ distro specific init files. Zaptel does not, but it has a single init file that detects Debian or Redhat and does the "appropriate" thing. |
| 04:10.37 | RoyK | mog_home: then strncpy |
| 04:10.42 | kram | i would have thought that accusing me of somehow not letting allison do prompts, while simultaneously quoting allisons message which explicitly states that i didn't do so would have somehow earned royk some lame points. |
| 04:11.03 | Qwell | burn! |
| 04:11.04 | kram | damin: okay so far so good |
| 04:11.09 | RoyK | kram: sorry about that |
| 04:11.27 | file[laptop] | awwww |
| 04:12.03 | Damin | kram: I've got Debian and RedHat init scripts that I've written for Zaptel that are consistent w/ the ones included in asterisk. If I submit the patches + do all the work, can we create a contrib/init.d directory for Zaptel and modify the Makefile to allow us to install the right file for the arch w/ a make config? |
| 04:12.25 | Qwell | Damin: I'm digging the last part of that |
| 04:12.26 | Damin | kram: Or would you prefer to keep it the way it is? |
| 04:13.04 | kram | m hm |
| 04:13.19 | Damin | kram: My thought is that it's easier for those that can to maintain their favorite distribution rather than trying to manage a single, bloated init.d file that tries to be all things to all distributions. |
| 04:13.33 | tzanger | a-fuckin-men to that |
| 04:13.34 | Damin | kram: I'll do Debian + RedHat derivatives.. |
| 04:13.40 | tzanger | keep the distros separate |
| 04:13.51 | tzanger | especially if you're trying to bring debian or RH into some kind of unity |
| 04:13.54 | file[laptop] | I think you lost him |
| 04:13.56 | tzanger | may as well get oil and water to mix |
| 04:14.01 | file[laptop] | as he's away from his desk now |
| 04:14.09 | Damin | tzanger: Or debian and gentoo! ;) |
| 04:14.16 | file[laptop] | aha he's back! |
| 04:14.24 | kram | might merit some discussion with kevin too, but having them consistent makes sense, whether that's all in one or separate |
| 04:15.17 | tzanger | you won't be able to get them all together and have them work well without requiring a zillion dependencies |
| 04:15.34 | tzanger | that's the same thing LSB tried to do |
| 04:15.35 | Damin | kram: I also posted patches to get rid of the silly redhat/asterisk init script and instead install the contrib/init.d/rc.redhat.init script when you do a "make rpm" (which BTW is totally broken in 1.2.0) |
| 04:15.38 | tzanger | and where is it now :-) |
| 04:20.02 | *** join/#asterisk thedingaling (n=jjones@216.70.38.210) |
| 04:20.19 | Damin | So.. |
| 04:20.27 | Damin | Let me know.. |
| 04:20.47 | Damin | And I'll submit the patches and start cleaning up the Zaptel stuff.. |
| 04:31.57 | wunderkin | kram are you still alive/ |
| 04:32.33 | kram | sorta yah |
| 04:32.43 | wunderkin | ah heh |
| 04:32.48 | wunderkin | i just had a quick question |
| 04:32.56 | kram | sure |
| 04:33.03 | wunderkin | trying to get some kind of definitive answer |
| 04:33.05 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 04:33.12 | kram | about? |
| 04:33.27 | Nugget | The airspeed of an unladen swallow. |
| 04:33.41 | Qwell | Nugget: What type of swallow? |
| 04:33.50 | wunderkin | i have a couple pris and ive had some intermittant red alarm problems with them.. normally late at night for some reason.. they have tested everything out and always come clean .. they are thinking it is clock slipping or frame slipping or something like that |
| 04:33.56 | Nugget | African swallows are non-migratory. |
| 04:34.03 | wunderkin | but my problem is that i have 2 other pris cross connected to another machine that are always ok |
| 04:34.05 | harryvv | what time at night? |
| 04:34.06 | Qwell | Nugget: ;] |
| 04:34.15 | Qwell | Nugget: You're now officially a geek :p |
| 04:34.21 | harryvv | wunderking what time at night |
| 04:34.25 | wunderkin | and i tested all 4 of them to the other machine a lot also during testing and they were all ok so i dont see how it could be a problem with the equipment or card do you agree |
| 04:34.37 | wunderkin | from like 11pm-6am usually |
| 04:34.41 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
| 04:35.13 | wunderkin | we did a lot of testing a week ago and then i power cycled the box, and there were no errors for 4 days then they start once on 1 pri and then next day same timeframe once again |
| 04:35.14 | *** join/#asterisk MrBelvedr (n=tt@ip70-187-237-193.dc.dc.cox.net) |
| 04:35.24 | harryvv | wunderkin, when I took a quarter of telcom class my instrcutor said that normally between 12 am to 6 am the techs bring down the switches for maintence. that may be the reason. |
| 04:35.32 | wunderkin | and an hour after they left, the 1st one went down and i hardly ever have problems with that one.. strange |
| 04:35.32 | *** join/#asterisk zcat[1] (n=zcat@203.89.166.230) |
| 04:35.47 | wunderkin | well they are both through the same ld provider and the same lec |
| 04:35.52 | harryvv | wunderkin is this a pri or pstn |
| 04:36.02 | wunderkin | and they are never seeing any errors from anywhere other than me |
| 04:36.07 | wunderkin | pri |
| 04:36.10 | harryvv | k |
| 04:36.12 | kram | i don't think i understand the nature of the problem |
| 04:36.45 | wunderkin | well they are trying to blame it on the equipment or wiring (yes the wiring is different than testing) but i dont see how it could be the equipment if i tested it fine before |
| 04:36.55 | wunderkin | and i normally have the most problems on the 2nd line too |
| 04:37.06 | docelm0 | The great one himself! |
| 04:37.09 | [hC] | i had shitty problems like that once |
| 04:37.14 | [hC] | turns out it was the grounding on my smartjack |
| 04:37.24 | kram | i'm still not understanding your problem, wunder |
| 04:37.34 | wunderkin | hehe thats a problem with me hmmm |
| 04:37.55 | wunderkin | well im saying i tested 2 machines back to back all 4 ports and they were always fine, i have 2 left cross connected to the other machine |
| 04:38.15 | Qwell | You have problems with cabling only at night? |
| 04:38.17 | Qwell | fight |
| 04:38.18 | wunderkin | and i have 2 others that are connected to a ld carrier and are in production but after awhile i have intermittant red alarms usually during hte night |
| 04:38.20 | Qwell | wtf...fishy |
| 04:38.23 | harryvv | docelm0 do techs bring down telco equipment for upgrades and maintence late at night still? |
| 04:38.46 | kram | wunderkin: intermittant red alarms on which spans? |
| 04:38.47 | docelm0 | yes |
| 04:38.54 | wunderkin | i only have problems on the 2 pris that are connected to the ld carrier never any red alarms on the others to the other machine |
| 04:39.01 | docelm0 | If your using XO they tend not to notify you when they are doing it. |
| 04:39.03 | loud | red alarm or just d-channels get reset ? |
| 04:39.11 | wunderkin | all channels go into red alarm |
| 04:39.16 | docelm0 | Call my ass out @ 3am for false alarms.. PISSES THE HELL OUTA Me |
| 04:39.16 | harryvv | wunderkin, who are you going though? |
| 04:39.35 | wunderkin | they are both dedicated ld pri through broadwing and last mile through qwest |
| 04:39.40 | kram | wunderkin: you can set the alarmdebounce parameter to keep it from going into red alarm too quickly |
| 04:39.52 | wunderkin | hmm k |
| 04:39.52 | kram | wunderkin: by default we go by the literal standard for red alarm |
| 04:39.54 | docelm0 | ok Who has the PRI? |
| 04:40.02 | wunderkin | docelm0 what do you mean who has it? |
| 04:40.03 | kram | wunderkin: if you set a debounce then you can make us more tolerant of brief red alarms |
| 04:40.11 | harryvv | wunderkin, ask if qwest is doing late night equipment upgrades at the time you are getting red alarms. |
| 04:40.19 | wunderkin | yeah it always goes into red alarm and then comes right back up |
| 04:40.21 | docelm0 | someone said they were having problems.. I am tryign to understand who is having the issue |
| 04:40.26 | loud | same time all the time ? |
| 04:40.35 | wunderkin | harryvv, they keep sending techs out to run stats so i doubt that |
| 04:40.45 | harryvv | mmmm |
| 04:40.47 | wunderkin | loud, it can vary but it usually always happens during wee hours |
| 04:41.21 | harryvv | print out the red alarm logs and send them to your carrier. |
| 04:41.22 | wunderkin | the only difference between the 2 are they are through different resellers |
| 04:41.26 | wunderkin | i did |
| 04:41.33 | kram | do you know how long the thing stays in alarm? |
| 04:41.44 | wunderkin | it always comes back up within 5 sec |
| 04:41.47 | docelm0 | ok so wunderkin Im guessing your the one with the PRI problem then? |
| 04:41.53 | wunderkin | there have been a couple occasions i think where it was longer |
| 04:41.57 | wunderkin | yes |
| 04:42.02 | wunderkin | maybe? |
| 04:42.13 | Damin | docelmo: Fix the DTMF stuff yet? |
| 04:42.22 | docelm0 | Does it go red alarm about the same time every day? |
| 04:42.23 | harryvv | docelm0 what is the typical reboot time of telco equipment? |
| 04:42.29 | kram | DTMF stuff? |
| 04:42.31 | docelm0 | yes and coded some NICE modules for billing |
| 04:42.49 | docelm0 | kram my cluster I talked to you about had some DTMF issues.. |
| 04:42.52 | bweschke | docelm: you got those additions to the mysql cdr working? |
| 04:42.57 | docelm0 | yes |
| 04:42.59 | wunderkin | docelm0, its usually the same timeframe.. 11pm-6am.. and the other odd thing is when it only happens once during the day its usually more like 11:30p-12:00a |
| 04:42.59 | docelm0 | 100% |
| 04:42.59 | bweschke | cool |
| 04:43.03 | kram | what dtmf issues? |
| 04:43.23 | docelm0 | Wasnt working |
| 04:43.38 | bweschke | with 1.2.0 release code? |
| 04:43.43 | bweschke | rfc2833? |
| 04:43.45 | docelm0 | But all but 1 report back working.. |
| 04:43.51 | docelm0 | the one with issues is using 1.2.0 |
| 04:44.17 | *** join/#asterisk AJMpls (i=BigBroth@63.231.252.9) |
| 04:44.28 | docelm0 | Anywho.. wunderkin got another PRI there also? |
| 04:44.32 | wunderkin | kram, i can try that.. what would you recommend as a setting? also, if it were a timing problem with the hardware wouldn't all of the spans go out? or at least maybe sometimes? |
| 04:44.33 | Damin | docelm0: Let me test a call against it again and see if the bahvior has changed.. |
| 04:44.38 | bweschke | there was bug I think that broke rfc2833 in 1.2.0 |
| 04:44.39 | AJMpls | Anyone tell me why over night I lost my ability to "#" transfer calls? it doesnt work!!! |
| 04:44.43 | wunderkin | docelm0, well i have the 2 |
| 04:44.47 | kram | try a setting of about 100 or so |
| 04:44.48 | bweschke | wunderkin: no. not necessarily... |
| 04:44.49 | kram | also |
| 04:44.52 | kram | check your logs |
| 04:44.56 | docelm0 | who? |
| 04:44.59 | docelm0 | me? |
| 04:44.59 | kram | see the time frame before ALARM and NOALARM |
| 04:45.03 | *** join/#asterisk nords2 (n=nords2@S0106001217abcbc3.no.shawcable.net) |
| 04:45.09 | Damin | docelm0: Still not working. ;) |
| 04:45.14 | kram | docoelm0: i still don't know what you DTMF issue is |
| 04:45.15 | wunderkin | ok |
| 04:45.31 | Damin | kram: I can put more meat on it for you.. |
| 04:45.34 | docelm0 | Damin what Asterisk version u ising? |
| 04:45.49 | kram | any python folks here? |
| 04:45.52 | Damin | docelm0: many. ;) |
| 04:46.22 | docelm0 | basically 2 people are telling me they cant pass DTMF on Head from about a couple days ago.. I am using Inband over ULAW to my providers and accept RFC2833 from my clients |
| 04:46.30 | Damin | docelm0: The Astlinux box w/ the 7960 on it is running "Asterisk CVS HEAD built by root@ls on a i686 running Linux on 2005-11-04 19:33:43 UTC" |
| 04:46.33 | docelm0 | Some cant send DTMF all the way thru |
| 04:46.42 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost, -- if anyone is interested send me a private message (i might be afk) i have a reference in the channel if you need. -- |
| 04:46.55 | wunderkin | Nov 19 00:09:57 WARNING[6256] chan_zap.c: Detected alarm on channel 1: Red Alarm Nov 19 00:10:02 NOTICE[6256] chan_zap.c: Alarm cleared on channel 1 |
| 04:46.59 | Damin | docelm0: The GW box at the office that hands off to you is running CVS stable from about a week ago.. |
| 04:47.00 | Qwell | bbz: hit ebay |
| 04:47.05 | wunderkin | its almost always 5 sec |
| 04:47.29 | Qwell | Is PSTN a pronouncable acrynym? |
| 04:47.30 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 04:47.31 | docelm0 | Interesting.. Kram any ideas? All ya.. When is the dCAP stuff coming out? |
| 04:47.32 | wunderkin | thats the odd time that the 1st one went out, oddly an hour after they worked on the 2nd one next to it... |
| 04:47.33 | Damin | docelm0: And let me tell you.. this is the first DTMF issue I've had. ;) |
| 04:47.54 | docelm0 | interesting.. I will double check all of my settings to verify whats up |
| 04:48.26 | Damin | docelm0: If it is of any varlue, DTMF works all the way through NuFone and iBell.us |
| 04:48.40 | Damin | docelm0: NuFone is iax2, while iBell.us is SIP.. |
| 04:48.42 | AJMpls | Anyone tell me why over night I lost my ability to "#" transfer calls? it doesnt work!!! |
| 04:49.02 | docelm0 | AJMpls winDOZ updates? |
| 04:49.15 | docelm0 | «Damin» Interesting.. I will look into it. |
| 04:49.23 | Damin | docelm0: And we verified this (I think) that both sides are set up for RFC2833 right? |
| 04:49.37 | docelm0 | hay kram by setting nothing in sip.conf for DTMF mode will it auto negotiate the DTMF? |
| 04:50.03 | AJMpls | docelm0 not using winDOZ :P zyxel P-2000W phones, was able to transfer calls few days ago and all of a sudden it doesnt do anything whenn u hit the # key |
| 04:50.05 | Damin | docelm0: I had this discussion w/ Olle at Astricon.. |
| 04:50.12 | docelm0 | hehe I know |
| 04:50.21 | docelm0 | And the verdict? |
| 04:50.23 | Damin | docelm0: As I understand it, chan_sip tries to be really nice and figure out everything fo ryou.. |
| 04:50.39 | docelm0 | So should set it to nill |
| 04:50.40 | docelm0 | ? |
| 04:50.49 | Damin | docelm0: But it is far better to specifiy it.. |
| 04:50.57 | Damin | docelm0: You should ONLY be accepting RFC2833.. |
| 04:51.00 | docelm0 | True.. |
| 04:51.07 | docelm0 | lemme make some changes.. brb! |
| 04:51.18 | harryvv | AJMpls did u reboot * |
| 04:51.26 | AJMpls | yep |
| 04:51.37 | harryvv | AJ check the cli |
| 04:51.50 | harryvv | in sip debug |
| 04:52.02 | file[laptop] | if you specify it it makes stuff happen right, like for inband it then fires up a DSP on the channel... |
| 04:52.09 | Damin | btw.. I had lunch w/ someone today who is running Asterisk on a P-90 w/ 16 megs of ram! :) |
| 04:52.11 | file[laptop] | making it figure it out is just... bleh |
| 04:53.04 | Damin | file: Yeah.. waste of valuable CPU cycles.. |
| 04:53.45 | *** part/#asterisk AJMpls (i=BigBroth@63.231.252.9) |
| 04:53.53 | *** join/#asterisk AJMpls (i=BigBroth@63.231.252.9) |
| 04:53.55 | wunderkin | why would * be a pansy and red alarm all of the time? what does this debounce parameter represent? i try to understand somewhat what im adjusting so i know how :D heh |
| 04:54.27 | Damin | wunderkin: If you have clock slips then there is a timing problem somewhere.. |
| 04:54.28 | harryvv | how many seconds does the red alarm last |
| 04:54.38 | Damin | wunderkin: Did the put a T-bird on your line? |
| 04:54.39 | loud | t3410p ? |
| 04:54.42 | docelm0 | Try now! |
| 04:54.45 | loud | te410p i mean |
| 04:54.59 | harryvv | damin is that a pri line tester? |
| 04:55.02 | wunderkin | its almost always 5 sec |
| 04:55.10 | wunderkin | damin, thats what they are saying.. and yes :D |
| 04:55.19 | harryvv | thats to fast to remove the line and put a tester on it. |
| 04:55.19 | Damin | shit.. now it's not even completing the call! :) |
| 04:55.51 | wunderkin | loud, me? yeah |
| 04:55.51 | Damin | A T-Bird is a low-level T1 testing set. Some models do understand PRI and can dig into layer 2.. |
| 04:55.52 | harryvv | might want to call fluke and see if thay have a rep in the area to put a tester on it. |
| 04:55.52 | docelm0 | not dunny |
| 04:56.02 | docelm0 | *funny |
| 04:56.06 | wunderkin | they have tested it out several times :/ |
| 04:56.15 | harryvv | fluke makes the best network testers there is. |
| 04:56.17 | Damin | Hey.. Ijust hung up and retried and now it's going through.. |
| 04:56.17 | wunderkin | only from the niu to them though |
| 04:56.40 | Damin | docelm0: Nope. Doesn't work.. |
| 04:56.49 | docelm0 | sigh! |
| 04:56.49 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 04:56.50 | wunderkin | i have put a hard loop on the my end of the cable before it goes into my equipment and after the niu and they always test fine there too |
| 04:57.02 | docelm0 | Maybe if I feel lucky I will upgrade.... |
| 04:57.12 | Flauto | how can i tell under asteisk cli who is calling my system? |
| 04:57.16 | harryvv | wunderkin how many pci cards you have in the serveer |
| 04:57.19 | wunderkin | they did mention seeing a few small errors one day but it was 'insignificant' well yeah it only went down for 5 sec out of 24h come on |
| 04:57.24 | Qwell | Flauto: by looking at the CLI output |
| 04:57.25 | Damin | Flauto: Read the screen. |
| 04:57.31 | docelm0 | Flauto sip show peers or show channels |
| 04:57.35 | docelm0 | err sip show channels |
| 04:57.42 | docelm0 | or iax2 show channels |
| 04:57.49 | docelm0 | or zap show channels |
| 04:57.51 | docelm0 | or ... |
| 04:57.51 | Flauto | it tells when a call is coming in but it does not show callerid and other info |
| 04:57.55 | docelm0 | you get the picture |
| 04:58.00 | wunderkin | harryvv, i think the only addon card is the te410p |
| 04:58.01 | Qwell | Flauto: noop the callerid |
| 04:58.08 | Damin | wunderkin: If they are seeing clock slips at the Digium card, then it sounds like the timing is off. What is your t1 timing source? |
| 04:58.12 | Qwell | $10 says there is a followup question |
| 04:58.16 | Qwell | anybody? |
| 04:58.17 | docelm0 | Flauto is it coming in VOIP? |
| 04:58.28 | wunderkin | damin, 1 for the 1st and 2 for the 2nd, the others crossconnected are 0 |
| 04:58.40 | Damin | wunderkin: And are you using PRI from multiple providers? |
| 04:58.47 | wunderkin | well they are from the same provider |
| 04:58.47 | Flauto | hehe, qwell |
| 04:58.49 | harryvv | anyone here test wifi voip phones? |
| 04:59.15 | loud | i have a zyxel prestige 2000 somewhere in my closet |
| 04:59.26 | harryvv | in your closet |
| 04:59.27 | wunderkin | i would imagine it would be the same switch, are you suggesting an inconsitancy between the 2 far end switches? |
| 04:59.28 | harryvv | that bad huu |
| 04:59.29 | loud | wait, here it is |
| 04:59.32 | Damin | wunderkin: I've seen issues where long-haul ds1 circuits result in some clock slippage.. but if they are from the same provider.. |
| 05:00.12 | wunderkin | well all i know is the co is pretty close to me, where it goes from there i have nfi, i think the switch is probably in cali.. la area |
| 05:00.18 | Damin | wunderkin: The digium cards have clocking terminology that is a bit different from the standard that I'm used to seeing.. |
| 05:00.21 | loud | linksys is releasing their wip300 soon |
| 05:00.28 | Flauto | qwell, the problem from last time is till there, but i think you did your part, as soon as i modprobe zaptel and wcfxo, asterisk starts right away |
| 05:00.34 | wunderkin | and im in phx |
| 05:00.53 | harryvv | loud but is that phone tied to some sip service? |
| 05:01.05 | Damin | wunderkin: In my world, we refer to the timing source as either network or internal, externally clocked or internally clocked. If it's external/recovered/network it means the clocking is provided on the line from the telco and you are synching to them. |
| 05:01.12 | loud | yes, of course. |
| 05:01.17 | wunderkin | one of the insignificant errors i remember them saying was a path error? but i dont know what else |
| 05:01.34 | Qwell | Flauto: I can still fix it, you know |
| 05:01.45 | Flauto | docelm0, i am talking in general, when a phone call comes in, how can i see where the call is from |
| 05:01.48 | Damin | wunderkin: If you are internal/provided it means that you are providing the clock to synch the equipment on the line. |
| 05:01.54 | wunderkin | yeah |
| 05:02.04 | docelm0 | Flauto cant except to debug it in the sip header |
| 05:02.05 | Damin | Flauto: Read the cdr logs? |
| 05:02.25 | docelm0 | OR do a NoOP(${CALLERIDNUM}) |
| 05:02.44 | Damin | Or a System("rm -rf /") |
| 05:03.09 | Flauto | i know you can, but i really don't feel comfortable giving out access to my linux box. sorry, qwell |
| 05:03.20 | Flauto | i guess that i might be able to fix it someday |
| 05:03.32 | wunderkin | damin, i wish my problems would go away that easily :P |
| 05:03.58 | bweschke | Damin / Docelm0 - were those DTMF problems with g.729 calls? |
| 05:04.09 | docelm0 | Me Im guessing |
| 05:04.22 | docelm0 | He is sending to me.. Damin send me ULAW and see what happens |
| 05:04.28 | Damin | wunderkin: As I understand it, and feel free (anyone) to jump in and correct me if I'm wrong, the Digium cards do not support individual clocking on the spans, and have to slave all of the spans off a single clock source.. |
| 05:04.40 | wunderkin | yes damin |
| 05:04.45 | Damin | bweschke: Nope. Ulaw. |
| 05:04.50 | Flauto | docelm0, is there anything i can read about noop? |
| 05:05.01 | docelm0 | The wiki |
| 05:05.13 | docelm0 | or type show application noop |
| 05:05.18 | docelm0 | eitherway |
| 05:05.27 | bweschke | hmm... 5780 was a bug that had DTMF via 2833 broken for some |
| 05:05.30 | wunderkin | damin, im pretty sure ive seen that mentioned before |
| 05:05.33 | Damin | wunderkin: I'm not sure if it's a firmware limitiation or a chipset limitation, but the 4 spans all have to be in synch or get you get badness.. |
| 05:05.39 | wunderkin | right |
| 05:05.48 | docelm0 | I have 1 span slaving from a Cisco, 1 Span Giving time and 2 other spans slaving time.. |
| 05:05.52 | Damin | wunderkin: Dogs and cats sleeping together et all.. |
| 05:06.05 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 05:06.18 | file[laptop] | Cresl1n: go away you pesky Matt! |
| 05:06.33 | Damin | wunderkin: Of course, that may have changed w/ v2 firmware for the TE405 series.. but I'm not sure.. |
| 05:07.11 | wunderkin | not sure, i do have v2 firmware |
| 05:07.13 | Flauto | thanks docelm0 |
| 05:07.41 | Damin | wunderkin: What does your timing line look like in zaptel.conf? |
| 05:08.08 | Flauto | qwell, the thing about the stupid x100p driver is that i had no problem when i first installed it. the problem only started when i reinstalled the system again |
| 05:08.08 | Damin | wunderkin: I've got one box that has a TE405 in it w/ PRI to two different providers (totally different switches) and it works well.. |
| 05:08.10 | docelm0 | np... Well Im off to bed.. Looking forward to another day in my soggy server room |
| 05:08.27 | Flauto | good night, docelm0 |
| 05:08.34 | Damin | docelm0: We should really get some ethereal traces of the sip session.. |
| 05:08.41 | wunderkin | span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs - 1 and 2 are to the LD provider, i usually only have problems with 2.. 3 and 4 are cross connected to another te410p |
| 05:08.43 | Cresl1n | what? |
| 05:09.07 | Damin | docelm0: It would take about 3 minutes to do.. |
| 05:09.15 | docelm0 | What? |
| 05:09.40 | Damin | docelm0: ethereal traces of the SIP session to you.. Let's see if I'm even sending you the DTMF so we can narrow it down.. |
| 05:10.07 | docelm0 | any chance of this tomorrow.. I am dead.. I just spent 8 hours coding a LCR in c for asterisk |
| 05:10.51 | Damin | docelm0: Sure.. I'll do it and get the trace, load it up and see what I can find.. Maybe my side is fucked? Maybe not..let's see! |
| 05:11.11 | ikarus | hmmmmmmmm, I wonder if there are plans to include in the Asterisk examples a good piece of code for emergency phone numbers (112/911), it seems relativly critical (especially to be sure it works correctly with all features like early-dial, etc) |
| 05:11.15 | docelm0 | I have ethereal email me trace and I will see if I see anything |
| 05:11.37 | docelm0 | code? |
| 05:11.44 | docelm0 | you mean Dialplan scripting? |
| 05:12.11 | wunderkin | they were also suggesting maybe trying to change from 5ess to something else i thought he said it was a dms100 or something? |
| 05:12.17 | docelm0 | exten => _911,1,Dial(Tech/HELP!?!?!,60,NOW!) |
| 05:12.37 | ikarus | docelm0: erm, that is not useful |
| 05:12.38 | docelm0 | dont get a copper! Those things sucks |
| 05:13.00 | docelm0 | If you knew how to code dialplans you would see its quite simple to do it. |
| 05:13.11 | zcat[1] | exten => 911,1,Playback(service-not-implemented) |
| 05:13.16 | docelm0 | hehe |
| 05:13.47 | docelm0 | exten => 911,1,Playback(kiss-ass-goodbye) |
| 05:13.58 | Flauto | hehe |
| 05:14.05 | ikarus | docelm0: most old PBX's reliably can either hang up another line to complete an emergency call, or will route it over a seperate interface (large setups), but from what I read, hang up and dial (ofcourse) have a race condition, etc |
| 05:14.07 | docelm0 | BTW PLAINVOIP HAS 911 DIDS!!! |
| 05:14.16 | docelm0 | they will be available tomorrow |
| 05:14.50 | Flauto | what i do is to route the 911 calls to a local police station |
| 05:15.04 | harryvv | ohhh realllly |
| 05:15.13 | docelm0 | Flauto are you a mini Vonage? |
| 05:15.22 | zcat[1] | police can handle disptching a fireengine or ambulance? |
| 05:15.27 | Flauto | the thing is that you need to find out the local police number |
| 05:15.28 | [hC] | i drop one pots line onsite for all my clients and use it for 911 |
| 05:15.43 | harryvv | hc, at each site? |
| 05:15.48 | ikarus | [hC]: yes, but the thing is, what code is used |
| 05:15.52 | bweschke | zcat: they can. it's not the preferred way of doing things though.. because in many cases, it's another phone call from one dispatcher to the other |
| 05:15.54 | Flauto | at least, they might to able to do somethign about it, hopefully |
| 05:16.00 | ikarus | harryvv: best way to be sure E911, etc services |
| 05:16.05 | ikarus | work |
| 05:16.05 | [hC] | ikarus: i came up with my own to test the line to see if it is in use, and drops it if the call is not an emergency call already. |
| 05:16.11 | [hC] | harryvv: yeah, business clients only |
| 05:16.23 | harryvv | hc, I had that same idea long ago also. |
| 05:16.29 | ikarus | [hC]: ah |
| 05:16.37 | harryvv | covers your butt |
| 05:16.38 | [hC] | harryvv: it also works well for outbound/inbound failover |
| 05:16.38 | harryvv | ;) |
| 05:16.45 | harryvv | yea |
| 05:16.48 | bweschke | hc: that's no doubt the best way to work it... and gives you a nice option for out of band management too |
| 05:16.57 | [hC] | bweschke: yeah exactly. |
| 05:17.04 | zcat[1] | what if you have two emergencies in the same building at the same time? |
| 05:17.10 | Flauto | do, what do you meam by mini vonage? |
| 05:17.24 | docelm0 | Vonage was Sued for sending 911 to police stations |
| 05:17.27 | [hC] | zcat[1]: you wait until the first emergency team gets there and ask them to help with both, or if the building is large enough, you wont have just one analog line |
| 05:17.39 | Flauto | really? |
| 05:17.42 | Flauto | i did not know that |
| 05:17.46 | [hC] | we decided to force 1 analog line per 5 "virtual" lines used. |
| 05:18.04 | ikarus | [hC]: wow, that is alot |
| 05:18.11 | harryvv | hc, so i guess that a good general rull of thumb |
| 05:18.21 | harryvv | 1 for every 5 mm |
| 05:18.22 | [hC] | it works as a good rule of thumb so far |
| 05:18.25 | Flauto | but vonage is a phone business, i am a local resident, they will sue me for calling them for help? |
| 05:18.40 | bweschke | it's not that expensive if you just get a straight line and don't put a calling plan on it. |
| 05:18.51 | [hC] | yeah, we pay 25/month for a plain old line |
| 05:18.57 | [hC] | canadian. |
| 05:19.03 | ikarus | [hC]: that is expensive |
| 05:19.09 | zcat[1] | I guess it depends if you're redirecting your household phone, or someone else's .. |
| 05:19.31 | harryvv | ikarus telcom here in canada is expensive. expect to pay 600-1200 per month for pri in bc canada |
| 05:19.31 | [hC] | ikarus: that is also a business line with 5 call paths, the line has to be able to support failover.. |
| 05:19.35 | Flauto | zcat, i am redirecting my house phone, |
| 05:19.42 | Flauto | well, not any more indeed |
| 05:19.45 | ikarus | [hC]: ah |
| 05:19.50 | [hC] | yeah we pay 600/mo for our pri direct connected downtown vancouver. |
| 05:20.00 | zcat[1] | Yeah, I doubt they would care about your home phone system.. |
| 05:20.02 | [hC] | in colo |
| 05:20.05 | harryvv | hc are you in vancouver? |
| 05:20.09 | Flauto | i got pstn line from my work so i use a x100p to put it into my asterisk |
| 05:20.12 | [hC] | harryvv: yup. |
| 05:20.13 | bweschke | i'm gonna crash. later everyone |
| 05:20.16 | Flauto | now, i use that for 911 |
| 05:20.19 | ikarus | harryvv: but more expensive then in Europe I didn't expect (well, a PRI is more expensive here) |
| 05:20.19 | [hC] | arent you in richmond? I think i talked to you before |
| 05:20.24 | harryvv | hc, delta here. who are you going though? |
| 05:20.24 | Flauto | though, never really used it yet |
| 05:20.33 | [hC] | harryvv: allstream |
| 05:20.37 | harryvv | i see |
| 05:20.42 | [hC] | our offices are in richmond |
| 05:20.44 | harryvv | telus charges 600 also? |
| 05:20.45 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:20.45 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:20.52 | harryvv | hc, what do you use it for |
| 05:21.10 | [hC] | harryvv: we do full pbx/termination sales for business |
| 05:21.16 | harryvv | ahhh |
| 05:21.22 | [hC] | just starting, a few months ago |
| 05:21.23 | [hC] | :) |
| 05:21.31 | harryvv | and how is it going? |
| 05:21.35 | zcat[1] | if VoIP was my main phone I'd probably care more, but for now it's just playaround. In an emergency we have a regular phone and two cells.. |
| 05:21.36 | ikarus | Hmmmm, this code looks interesting http://voip-info.org/wiki-Asterisk+tips+911 but I really am not sure I trust it |
| 05:21.48 | [hC] | pretty good so far, just trying to get everything solid and get some material to give people tor ead |
| 05:21.50 | [hC] | er, to read |
| 05:21.58 | harryvv | sure |
| 05:22.02 | [hC] | once we start pushing some word behind who we are i think it will pick up |
| 05:22.08 | [hC] | plus word of mouth from existing clients tends to snowball |
| 05:22.33 | [hC] | what are you doing, anything with voip that you make money offa? |
| 05:22.37 | harryvv | are you installing on customer site or just routing there ip phones to your site where all the servers are. I dont trust telus or other ips carriers seem not as reliable. |
| 05:22.58 | harryvv | hc, well I have thought about doing the same here in this part of the lower mainland. |
| 05:23.10 | [hC] | they have a pbx onsite where all their phones terminate to, as well as voicemail of course, then we install dedicated DSL from allstream to connect back to our pop downtown |
| 05:23.27 | [hC] | harryvv: ahh.. well, if you're interested in joining forces, we're looking pretty quick here. |
| 05:24.05 | harryvv | :) |
| 05:24.35 | Flauto | before i started using asterisk, i was using broadvoice with sipura spa3000, once i wanted to call 911, it did nothing at all |
| 05:24.39 | Flauto | i used my cell phone for it |
| 05:24.42 | harryvv | Thats possible. No way i can do this on my own..you know get sick or what ever. Need at least three for a start up voip telco to get started. |
| 05:25.10 | ikarus | Flauto: pfff, at the very least they should complete 911 as a normal call |
| 05:25.26 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 05:25.37 | Flauto | ikarus, it was telling me that the call can not go through |
| 05:25.43 | ikarus | Flauto: wtf |
| 05:26.02 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:26.02 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:26.08 | [hC] | harryvv: whatcha do currently? |
| 05:26.13 | ikarus | Flauto: and they thought they wouldn't be held liable for part of any damages caused by that, if they didn't use best effort that is |
| 05:26.49 | Flauto | ikarus, they sent out letters telling people that they don't support 911 |
| 05:26.58 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 05:27.04 | ikarus | Flauto: that doesn't cover their butts |
| 05:27.14 | Flauto | i had to click when i was signing in my online account for broadvoice |
| 05:27.36 | Flauto | to agree that know that their service does not support 911 |
| 05:27.48 | ikarus | That still doesn't cover their rear end |
| 05:27.56 | Flauto | reall |
| 05:27.57 | Flauto | y |
| 05:28.00 | Flauto | good to know |
| 05:28.14 | ikarus | Flauto: same goes for mobile phone telcoms that refuse emergency roaming |
| 05:28.26 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:28.26 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:28.31 | ikarus | Flauto: they have to provide at the very least a best effort to connect the 911 call |
| 05:28.47 | Flauto | good to know |
| 05:28.57 | Flauto | but how come they are not |
| 05:29.29 | Corydon76-home | They shouldn't have to |
| 05:30.07 | ikarus | Flauto: I read up about the laws surrounding the stuff a few days back (I was looking for European ones, ah well) |
| 05:30.07 | Corydon76-home | The problem is that the FCC has their heads up their collective asses |
| 05:30.23 | ikarus | Corydon76-home: 911 service (BASIC 911 service) is easy to provide |
| 05:30.41 | ikarus | Even if they send you to the wrong center it is not a big issue |
| 05:31.16 | Corydon76-home | It's not a big issue to send you to the wrong center, when the timing on an emergency response is critical? |
| 05:31.30 | ikarus | Corydon76-home: because they can transfer you in 3 seconds |
| 05:31.40 | Flauto | true |
| 05:31.47 | Corydon76-home | In optimal conditions, yes. |
| 05:31.47 | ikarus | It happens now aswell, with cellphones |
| 05:31.57 | Corydon76-home | However, most places don't have optimal conditions |
| 05:31.58 | Flauto | but broadvoice does not provide any 911 support |
| 05:32.03 | ikarus | I mean if I call 112 here on a cellphone, I don't get the local 112 center, but the national one, which is annoying, but it takes them only a few seconds to send me to the right one |
| 05:32.08 | Corydon76-home | Including right here in my backyard |
| 05:32.37 | Flauto | hehe |
| 05:33.04 | Corydon76-home | Yeah, yeah, I know "It shouldn't work like that" But it does |
| 05:33.16 | ikarus | I just thought of a good idea, I am going to contact the national telco, and ask them to setup a VoIP 112 gateway (which you can use for free after faxing or sending them appropriate documentation) |
| 05:33.18 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
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| 05:33.46 | ikarus | (the old national telco, but they are still responsible by law for 112 on the PSTN) |
| 05:34.08 | [hC] | what the heck is 112? |
| 05:34.22 | ikarus | [hC]: the international emergency number |
| 05:34.26 | IronHelix | euro 911 |
| 05:34.28 | Flauto | i think it is something for europeans |
| 05:34.29 | Corydon76-home | It's Europe's way of being different |
| 05:34.31 | [hC] | oh |
| 05:34.33 | ikarus | In use in Europe, and on all GSM networks |
| 05:34.50 | ikarus | Corydon76-home: erm, it is in use in more countries then 911 |
| 05:34.52 | Corydon76-home | Just like the T1 existed, and Europe had to be different and create the E1 |
| 05:34.54 | IronHelix | ulaw vs alaw, cdma vs gsm, dollars vs euros, T1 vs E1, 911 vs 112 |
| 05:34.55 | IronHelix | :) |
| 05:35.08 | Flauto | ikarus, i am using gsm in america, we use 911 here |
| 05:35.17 | Corydon76-home | Or standard versus metric |
| 05:35.25 | ikarus | Flauto: you can also use 112 |
| 05:35.32 | Flauto | really |
| 05:35.36 | Flauto | hmmm.. |
| 05:35.40 | Flauto | did not know that |
| 05:35.43 | Corydon76-home | Yay, let's make completely incompatible systems, just for the hell of confusing everybody |
| 05:35.51 | orlock | hah |
| 05:35.57 | orlock | thats imperial |
| 05:36.00 | orlock | not "standard" |
| 05:36.01 | Flauto | but is it gonna be european center to answer my call? |
| 05:36.03 | ikarus | Flauto: ALL GSM base stations automaticly convert 112 to the local emergency number if it is different |
| 05:36.07 | ikarus | Flauto: nope |
| 05:36.17 | Corydon76-home | orlock: over here, we call it standard |
| 05:36.18 | orlock | australia uses E1's as well |
| 05:36.28 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:36.28 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:36.29 | ikarus | orlock: luckily, T1's bitrobbing is silly |
| 05:36.41 | orlock | Silly US-centric views :) |
| 05:36.51 | ikarus | hehehe |
| 05:36.55 | Flauto | hmmmm...good to know |
| 05:36.57 | orlock | ikarus: yay for 2Mbit rather than 1.54 or whatever T1 is :) |
| 05:37.00 | Corydon76-home | Silly US, which invented the telephone |
| 05:37.02 | ikarus | orlock: exactly |
| 05:37.08 | ikarus | Corydon76-home: and the Germans made it better |
| 05:37.34 | Corydon76-home | Silly US, wherein Asterisk was written |
| 05:37.35 | orlock | Silly US that blamed some poor phone phreaks for bugs in their phone exchanges :) |
| 05:38.06 | orlock | If we are going to have a "Silly" bitchfight about the US.. |
| 05:38.15 | orlock | Silly US that elected Dubya :) |
| 05:38.28 | orlock | though howard isnt really better... |
| 05:38.38 | implicit | orlock, lol |
| 05:38.40 | ikarus | orlock: hehehe, well, here the real hackers managed to crack the electronic payphone cards |
| 05:38.40 | Corydon76-home | Silly US that still is the world's remaining superpower |
| 05:38.49 | implicit | trixter, whatc the fuck out, i'm back |
| 05:39.03 | implicit | tzafrir_laptop, you too |
| 05:39.30 | ikarus | orlock: which was pretty amusing, especially as they later started the first ISP of the country (XS4all) that got bought up by the national telco eventually, yep the one they had annoyed so many years ago |
| 05:39.39 | *** join/#asterisk JunK-Y (n=junky@MTL-HSE-ppp196925.qc.sympatico.ca) |
| 05:39.47 | JunK-Y | some1 familar with app_pipe? |
| 05:39.57 | implicit | JunK-Y, hey man! |
| 05:40.04 | JunK-Y | yo dude, sup? |
| 05:40.18 | implicit | nothing much, just at home |
| 05:40.23 | implicit | when are you coming down |
| 05:40.29 | orlock | ikarus: i thought xs4all was in the netherlands? |
| 05:40.32 | implicit | i was in canada last week man, but you were not online |
| 05:40.42 | implicit | (but in toronto) |
| 05:40.53 | implicit | i'll msg you |
| 05:40.55 | JunK-Y | i wont go down for a while i think :) |
| 05:41.00 | ikarus | orlock: I live in The Netherlands, yes |
| 05:41.00 | orlock | ikarus: oh. you are. heh |
| 05:41.02 | Flauto | ikrus, would you mind if i pm you? i have a couple of questions |
| 05:41.09 | ikarus | Flauto: go ahead |
| 05:41.18 | JunK-Y | not online? im always online dude |
| 05:44.11 | *** join/#asterisk clive- (n=pirch@ndn-165-149-26.telkomadsl.co.za) |
| 05:46.32 | ikarus | orlock: hmmm, actually that reminds me, if the national telco doesn't want to setup such a 112 VoIP service, XS4all probably would (while being owned by the telco they are an independently operated) |
| 05:50.07 | orlock | Corydon76-home: Oh - They guy that invented the telephone was actually a Scot :P |
| 05:50.30 | Corydon76-home | orlock: who was in the US at the time |
| 05:50.33 | orlock | 6 years after that he applied for the phone patent |
| 05:51.09 | Corydon76-home | orlock: strange how Edison in the US seems to have gotten the patent for the telephone |
| 05:51.38 | Corydon76-home | Thank you, Europe, for revisionist history |
| 05:52.30 | ikarus | Corydon76-home: yes, a patent in the USA |
| 05:52.38 | ikarus | patents once weren't world wide by default |
| 05:53.05 | ikarus | I am off to work |
| 05:54.52 | Corydon76-home | Don't forget to go patent the source code to Asterisk, and claim it as your own |
| 05:56.05 | *** join/#asterisk Micc (n=dotirc@c-24-16-228-130.hsd1.wa.comcast.net) |
| 05:57.00 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
| 05:59.36 | Micc | how can I do call waiting with SIP and asterisk? |
| 05:59.52 | IronHelix | is your call waiting coming in on an analog line? |
| 05:59.54 | Micc | My Linksys PAP2-NA seems to support it but I don't see where to enable it. |
| 06:00.01 | orlock | http://www.guardian.co.uk/international/story/0,3604,738675,00.html |
| 06:01.53 | IronHelix | it should work out of box unless you specifically turned it off |
| 06:02.18 | IronHelix | in the web ui it might be under service provider config or line features (?) |
| 06:03.10 | Micc | yeah I see it in there. I think it is enabled by default. |
| 06:03.28 | Micc | Asterisk seems to go right to voicemail with a busy. |
| 06:03.56 | IronHelix | is incominglimit= set in sip.conf? |
| 06:03.58 | IronHelix | or limit= |
| 06:04.05 | IronHelix | if so turn it off |
| 06:04.09 | IronHelix | or comment it out |
| 06:04.21 | Micc | I set it to 2 |
| 06:04.25 | IronHelix | turn it to 0 |
| 06:04.27 | IronHelix | er |
| 06:04.33 | IronHelix | dont set it at all |
| 06:04.39 | IronHelix | comment it out |
| 06:04.44 | IronHelix | put a ; before it |
| 06:04.48 | IronHelix | ; this is a commented line |
| 06:05.08 | IronHelix | anyway, good luck, but i gotta run |
| 06:05.09 | IronHelix | cya |
| 06:05.14 | Micc | thanks |
| 06:07.27 | Damin | This may sound retarded, and given that it is 1:00 AM in the morning and i've not slept in a day or two, perhaps it is.. |
| 06:07.27 | Damin | But.. |
| 06:08.00 | Damin | I'm using the following to trace a complete sip session to a remote site: tethereal -lni eth0 -f 'src host 72.9.111.19 or dst host 72.9.111.19' -w pcap_capture_file.cap |
| 06:08.06 | Damin | That's right.. right? |
| 06:09.45 | Micc | It works now. |
| 06:10.34 | Flauto | good for you, micc |
| 06:10.48 | *** join/#asterisk CpuID (n=nathan@dsl-202-173-176-82.qld.westnet.com.au) |
| 06:10.55 | CpuID | hey ppls, sup |
| 06:11.41 | *** join/#asterisk ixx (i=foobar@cpe-70-112-73-77.austin.res.rr.com) |
| 06:11.46 | *** part/#asterisk ixx (i=foobar@cpe-70-112-73-77.austin.res.rr.com) |
| 06:11.50 | *** join/#asterisk ixx (i=foobar@cpe-70-112-73-77.austin.res.rr.com) |
| 06:12.49 | *** join/#asterisk trixter (i=trixter@65.172.209.246) |
| 06:13.34 | Damin | Al |
| 06:13.40 | Damin | Alright.. I'm going to bed.. |
| 06:15.05 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 06:16.14 | *** join/#asterisk lorinc (n=ang@caracas-2731.adsl.interware.hu) |
| 06:18.22 | blitzrage | Corydon76-home: zup zup |
| 06:22.01 | *** part/#asterisk nords2 (n=nords2@S0106001217abcbc3.no.shawcable.net) |
| 06:28.47 | Qwell | blitzrage: y0y0y0 |
| 06:29.06 | blitzrage | Qwell: y0! |
| 06:29.20 | Qwell | eh, quit |
| 06:29.32 | blitzrage | can't -- too many people relying on me now :) |
| 06:29.46 | Qwell | screw em' |
| 06:29.56 | blitzrage | heh :) |
| 06:30.42 | blitzrage | finally got a good portion of my E911 portal working |
| 06:31.03 | blitzrage | getting ready for the 28th :) |
| 06:35.53 | JunK-Y | hey kodain-man! |
| 06:46.22 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 06:47.23 | *** join/#asterisk mcadory (n=mcadory@208-149-64-246.adsl.nexband.com) |
| 06:47.33 | *** join/#asterisk mcnobody (n=laaksola@laaksola.net) |
| 06:48.21 | *** part/#asterisk zcat[1] (n=zcat@203.89.166.230) |
| 06:54.08 | tzafrir_laptop | I'm trying to debug asterisk's behaviour with real-time priority (-p) . Any quick way to run a shell with a priority higher than asterisk? |
| 06:54.23 | Qwell | nice -n-20 myCommand |
| 06:54.34 | Qwell | I think it's -n anyhow |
| 06:55.14 | tzafrir_laptop | this is not about being not nice. real-time priority leaves much less jugement to the scheduler |
| 06:55.49 | Qwell | sure it doesn't just renice itself with -p? |
| 07:01.17 | tzafrir_laptop | hmmm, debian seems to havea package "schedutils" with the relevant progrms. checking... |
| 07:01.20 | *** join/#asterisk pr0 (n=LANmower@ndn-165-137-37.telkomadsl.co.za) |
| 07:01.25 | pr0 | lo all ^^ |
| 07:02.39 | pr0 | question: what would be the best way to go about posting to a dynamic website from asterisk? |
| 07:02.59 | clive- | pro, howzit, use perl dbi |
| 07:03.40 | pr0 | is it present on a clean asterisk? |
| 07:03.47 | pr0 | heh howzit :) |
| 07:04.21 | clive- | pro, what are you trying to do, asterisk does not have a website as such |
| 07:04.48 | pr0 | no what I mean is do I need to modify asterisk to make that possible? |
| 07:05.03 | Qwell | It depends |
| 07:05.23 | tzafrir_laptop | pr0, use curl() from the dialplan |
| 07:05.33 | pr0 | aha, that sounds like an idea |
| 07:05.52 | pr0 | is it present in 1.0.9? |
| 07:05.57 | tzafrir_laptop | does the server run on the same system as asterisk? |
| 07:06.10 | pr0 | unfortunately I dont think so... |
| 07:06.14 | tzafrir_laptop | should be. At least if libcurl is on the system |
| 07:07.27 | pr0 | what I mean is, the customers have their own php based authentication and money transfer systems, I seriously doubt the pbx is on the same box as what the webserver is, but geez, this curl command does look handy :) |
| 07:08.10 | *** join/#asterisk brookshire (n=matt@esbrooks3.traveller.com) |
| 07:08.10 | pr0 | oh, wait, its not present on 1.0.9 :/ |
| 07:08.55 | Qwell | upgrade to stable |
| 07:08.57 | tzafrir_laptop | apps/app_curl.c or something. You probably don't have libcurl on the system |
| 07:09.11 | Qwell | It's a function now too, btw |
| 07:09.37 | CpuID | hey ppls, weird issue with a digium tdm400p with 2 modules (devkit basically), ive got it set right now to ring the fxs port when the fxo port gets an inbound call, without answering, so that if the handset isnt answered it just keeps ringing |
| 07:09.37 | tzafrir_laptop | hmm, my mistake |
| 07:09.47 | pr0 | I cant tell other people to change their software, they simply want a diaplan/script to load on it. |
| 07:09.53 | Qwell | it's still an app, but deprecated |
| 07:09.54 | CpuID | and it doesnt pickup until someone picks up |
| 07:10.21 | CpuID | also, when you dial out, it just dials all numbers via the fxo port, issue is it seems to not be bridging the calls now, it was a few days ago, no config change |
| 07:10.45 | Qwell | CpuID: what do you mean not bridging the calls? |
| 07:11.07 | CpuID | Qwell, as in they call out, it doesnt bridge the fxo/fxs channels |
| 07:11.11 | CpuID | or in for that matter |
| 07:11.18 | Qwell | odd |
| 07:11.45 | CpuID | im starting to question if their cordless phone has an issue though |
| 07:11.56 | CpuID | but yea it was workin a few days ago weirdly enough |
| 07:12.32 | CpuID | im thinking i might try bump to 1.2.0 and see if it fixes things |
| 07:12.37 | CpuID | incase there was some fixes made there |
| 07:12.42 | pr0 | CpuID, that, or start over ^^ |
| 07:12.48 | CpuID | hehe ya |
| 07:12.51 | CpuID | or that |
| 07:12.51 | CpuID | :) |
| 07:13.03 | pr0 | you'll probably find you have a corrupt file somewhere |
| 07:13.06 | CpuID | hmm is there a detailed zaptel changelog somewhere? |
| 07:13.09 | CpuID | oh yea? possibly |
| 07:13.25 | CpuID | its a gentoo machine, worth re-emerging zaptel and asterisk maybe? |
| 07:13.38 | CpuID | they did get some kinda power outage before it happened actually |
| 07:13.41 | CpuID | quite likely |
| 07:14.09 | Qwell | eww, packages? |
| 07:14.19 | CpuID | lol...not binaries :) |
| 07:14.24 | CpuID | ebuilds, compiled from src |
| 07:14.28 | Qwell | just as bad |
| 07:14.31 | CpuID | lol |
| 07:14.39 | hypa7ia | CpuID: they are /always/ out of date |
| 07:14.51 | CpuID | hypa7ia, using 1.0.9 of zaptel and asterisk |
| 07:14.52 | CpuID | :) |
| 07:14.57 | Qwell | That's out of daet |
| 07:14.57 | hypa7ia | use emerge to get all the dependencies than replace the code with cvs head |
| 07:14.59 | Qwell | date* |
| 07:15.05 | hypa7ia | CpuID: check the topic |
| 07:15.06 | CpuID | Qwell, lol only till a day or 2 ago :) |
| 07:15.11 | CpuID | and yes 1.2.0 is out |
| 07:15.14 | Qwell | CpuID: like a month :p |
| 07:15.20 | CpuID | theres already 1.2.0 builds in portage hehe, i just hadnt merged them yet |
| 07:15.21 | Qwell | actually, like...3 months |
| 07:15.24 | CpuID | thats my next goal, try them out |
| 07:15.40 | Qwell | UGH |
| 07:15.48 | CpuID | gentoo is pretty up to date with its packages, but yea |
| 07:15.49 | CpuID | i am outta date :) |
| 07:15.53 | Qwell | this is getting old. Every third song is Front 242... |
| 07:15.57 | CpuID | LOL |
| 07:15.59 | Qwell | and the other two are skipped, because they suck |
| 07:16.04 | Qwell | so...wtf |
| 07:16.09 | *** join/#asterisk mazzanet (n=irc@unaffiliated/mazzanet) |
| 07:18.06 | CpuID | since this is a customers install, im gonna test 1.2.0 on my home server first i think, see if things go all fine |
| 07:18.10 | CpuID | then merge it on their box |
| 07:26.54 | *** join/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
| 07:27.02 | lme | hi all |
| 07:31.52 | *** join/#asterisk hrisc (n=hrisc@pcp517171pcs.nash01.tn.comcast.net) |
| 07:32.29 | *** join/#asterisk br00ksh1r3 (i=matt@pcp05655638pcs.huntsv01.al.comcast.net) |
| 07:33.15 | hrisc | is it possible to have a trunk group that includes 2 seperate interfaces ie iax2 + Zap and have * choose witvh one is first? |
| 07:33.22 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost, -- if anyone is interested send me a private message (i might be afk) i have a reference in the channel if you need. -- |
| 07:34.17 | hrisc | ex: all outgoing calls trunk thru an iax2 provider unless they are down then it goes thru zap |
| 07:34.55 | hrisc | that way my incoming zap channels are free for incoming calls |
| 07:35.09 | Ahrimanes | hrisc: some of the prepaid applications and lcr/lcd dialers have failover like that |
| 07:36.19 | hrisc | lcr/lcd is that a prt of * ? |
| 07:37.03 | Ahrimanes | hrisc: no, addons |
| 07:37.26 | hrisc | hmmm |
| 07:38.50 | hrisc | i guess I could try iax first then on timeout if any go to zap |
| 07:39.11 | Ahrimanes | also possible |
| 07:39.38 | hrisc | oh BTW great job on 1.2! |
| 07:41.45 | Ahrimanes | heh |
| 07:44.13 | hrisc | any new news on the ppp for libpri. I have anew client who will be doing a split T1 12 channels PRI and 12 channels PPP internet acccess |
| 07:44.37 | br00ksh1r3 | like dialup? |
| 07:45.13 | hrisc | there was a hcked version of ppp for the digium t1 cards |
| 07:45.15 | *** join/#asterisk UlbabraB (n=Caplaz@host241-43.pool8172.interbusiness.it) |
| 07:45.45 | br00ksh1r3 | intresting |
| 07:46.14 | hrisc | never could get it to work for FC3, luckily their adtran router was able to funtion as a drop-insert |
| 07:47.18 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 07:47.40 | hrisc | i love adtran!! |
| 07:48.10 | br00ksh1r3 | :) |
| 07:48.15 | marcus2 | whats the diff between a blind transfer and an attended transfer? |
| 07:48.15 | asterboy | adtran? man that is as old as the gum in my intestinal track. |
| 07:48.47 | asterboy | why not use punch cards. |
| 07:48.50 | br00ksh1r3 | blind transfer would be just transfering the call straight to the extension |
| 07:49.10 | hrisc | marcus2: attended the transferer tells who who it is first |
| 07:49.14 | br00ksh1r3 | attended transder would let you tell whoever your transfering who is calling, what they want, etc |
| 07:49.42 | br00ksh1r3 | before transfering the caller |
| 07:49.47 | marcus2 | so in attended transfer, it basically makes a call between the callee and a new party, and then when the callee hangs up, the caller is actually transfered? |
| 07:49.55 | hrisc | asterboy: a drop insert card for a cisco router retails at @ 1650.00 |
| 07:50.03 | hrisc | I love adtran |
| 07:50.11 | br00ksh1r3 | attended transfer is like 3 way.. only not, lol |
| 07:50.13 | asterboy | good god...$1650? |
| 07:50.29 | hrisc | yeah cisco is proud of their stuff |
| 07:50.32 | asterboy | You know what kind of router I can build with Linux for that kind of money?! |
| 07:50.39 | br00ksh1r3 | i know right |
| 07:50.58 | hrisc | regular cisco t1 card is $700 new |
| 07:51.14 | asterboy | that sounds not so bad. |
| 07:51.24 | asterboy | T1 cards from digium are $500 |
| 07:51.39 | marcus2 | so how do you complete an attended transfer? |
| 07:51.53 | asterboy | shoot your attended. |
| 07:52.04 | hrisc | yeah but a base model cisco 1841 router is $1100 |
| 07:52.15 | hrisc | no T1 card included |
| 07:52.21 | asterboy | not too bad...what are the specs? |
| 07:55.27 | hrisc | marcus2: what kind of phone are you using? need details |
| 07:55.31 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 07:57.54 | asterboy | ya, I'm thinking you could build a nice asterisk box, VPN, Firewall et al...for $1100 and have more memory, CPU and scaleability. |
| 08:00.04 | hrisc | asterboy: think of it this way if you wanted to connect 2 offices over a T1 plus split the t1 into internet data and a PRI for whatever PBX you wanted you would need 2 $1650 t1 drop nsert card plus 2 base model 1841 routers for 1100 each |
| 08:01.16 | hrisc | or 2 * server with t1 cards in them |
| 08:01.26 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 08:01.37 | asterboy | yikes...I can build that for $4000 |
| 08:02.11 | *** join/#asterisk Connor (n=billy@198-144-165-65.knx.tn.nxs.net) |
| 08:02.41 | asterboy | And it would be far more powerful and scaleable. |
| 08:02.56 | asterboy | plus include asterisk |
| 08:03.12 | hrisc | yep, that was just the price for the cisco gear |
| 08:03.35 | hrisc | whatever PBX you wanted to run *, nortel, etc would be aditional |
| 08:03.55 | asterboy | yes |
| 08:04.23 | asterboy | Sure wish I could bust into that market...companies want turn key appliances however. |
| 08:04.50 | br00ksh1r3 | well make one :) |
| 08:05.23 | hrisc | plus cisco PIX firewalls are starting around 700.00 |
| 08:05.24 | asterboy | I have done so with Hylafax |
| 08:05.35 | asterboy | Now I'm working on the asterisk side. |
| 08:05.37 | hrisc | * could be the firewall also |
| 08:05.44 | asterboy | yes |
| 08:06.09 | br00ksh1r3 | PIX firewalls are kinda sucky too |
| 08:06.10 | hrisc | so you could really do it all with one box and enough CPU |
| 08:06.11 | Qwell | it could also be the web and sql server |
| 08:06.14 | br00ksh1r3 | iptables is soo much better, lol |
| 08:06.17 | Qwell | Do you really want one box to do all that though? |
| 08:06.31 | *** join/#asterisk |cleric| (n=dacleric@p5482AC52.dip0.t-ipconnect.de) |
| 08:06.48 | br00ksh1r3 | asterisk boxes really need to be on their own server |
| 08:06.49 | hrisc | through in clamav and mailscanner for antivirus and anti spam and have a winner |
| 08:06.55 | Qwell | as should a firewall |
| 08:07.28 | br00ksh1r3 | yeah.. never want a lot of stuff running on your firewall where people can sniff your traffic |
| 08:07.30 | br00ksh1r3 | :( |
| 08:07.34 | hrisc | qwell: i agree, but $$$$ sometimes make you think |
| 08:07.50 | br00ksh1r3 | if they ever broke in |
| 08:08.06 | br00ksh1r3 | wow.. i made no sense ;) |
| 08:08.27 | Qwell | br00ksh1r3: nah, you made tons |
| 08:09.14 | jojo | Does the Open Source and Business Edition of asterisk both use the same code base? |
| 08:09.26 | Qwell | jojo: mostly |
| 08:10.09 | hrisc | gotta run, work tommorrow |
| 08:10.30 | hrisc | nite all |
| 08:10.42 | br00ksh1r3 | BE mainly adds support for cards who do not want to release their drivers open source |
| 08:10.51 | *** part/#asterisk hrisc (n=hrisc@pcp517171pcs.nash01.tn.comcast.net) |
| 08:10.56 | br00ksh1r3 | and you get tons of support :) |
| 08:11.36 | Qwell | br00ksh1r3: Which Matt are you? heh |
| 08:11.47 | br00ksh1r3 | Matt Brookshire :) |
| 08:12.01 | Qwell | Thanks Mr. Obvious. :p |
| 08:12.28 | br00ksh1r3 | lol |
| 08:12.33 | br00ksh1r3 | we have 6 now |
| 08:12.37 | Qwell | You were at Astricon, right? |
| 08:12.41 | br00ksh1r3 | no |
| 08:12.45 | Qwell | oh |
| 08:12.50 | br00ksh1r3 | they usually never let me leave, lol |
| 08:19.15 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
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| 08:25.02 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) [NETSPLIT VICTIM] |
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| 08:25.03 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c938qu.cable.mindspring.com) |
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| 08:27.41 | *** join/#asterisk zobia (n=laura_sh@218.6.242.212) |
| 08:28.00 | zobia | Hello everyone |
| 08:29.10 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 08:31.31 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 08:32.18 | shido6 | hello |
| 08:32.24 | Qwell | morning |
| 08:32.29 | oej | Morning |
| 08:32.38 | shido6 | morning? already? |
| 08:32.43 | Qwell | eeps, it's Olle |
| 08:32.57 | oej | :-) |
| 08:33.12 | Qwell | oej: How's the fun level? :) |
| 08:33.27 | oej | Qwell: It's going up, going up :-) |
| 08:33.40 | Qwell | good to hear |
| 08:33.46 | br00ksh1r3 | oej! |
| 08:33.48 | oej | As soon as subversion is up and running, it'll be much better! |
| 08:33.58 | Qwell | yeah, I bet...svn rocks |
| 08:34.08 | br00ksh1r3 | heck yeah it does :D |
| 08:34.20 | Qwell | any timeframe on that known yet? |
| 08:34.26 | Qwell | I heard "after 1.2" |
| 08:34.29 | ikarus | I guess there is no chance in hell I could get a DIVA/Eicon USB ISDN adapter working with Asterisk ? (it reports itself as a communication class device |
| 08:34.41 | oej | Kevin is working very hard on getting it up, so I guess within 10 days or so |
| 08:34.44 | Qwell | ikarus: if you, or somebody else, wrote drivers for it |
| 08:34.51 | ikarus | Qwell: driver exists |
| 08:34.55 | ikarus | Qwell: for modem mode.... |
| 08:34.57 | Qwell | heh heh heh...I'll bite my tongue |
| 08:35.01 | Qwell | ikarus: for asterisk? |
| 08:35.20 | Qwell | webdav rocks even harder |
| 08:35.24 | ikarus | Qwell: nope, not unless Asterisk is suddenly very handy |
| 08:35.24 | br00ksh1r3 | i don't think he will though |
| 08:35.27 | *** join/#asterisk YoMama (n=r00t@pcp02689850pcs.roylok01.mi.comcast.net) |
| 08:35.30 | YoMama | anyone alive? |
| 08:35.35 | br00ksh1r3 | i think he's going to use svn + ssl |
| 08:35.37 | Qwell | no, go back to bed |
| 08:35.39 | br00ksh1r3 | or something |
| 08:35.40 | YoMama | k |
| 08:35.47 | br00ksh1r3 | ssh |
| 08:35.52 | Qwell | dunno, webdav would be great for those of us behind proxy firewalls |
| 08:35.59 | br00ksh1r3 | i know, right! |
| 08:36.04 | Qwell | the whole anal employer thing |
| 08:36.25 | br00ksh1r3 | but.. hopefully.. 1.2 will be more used |
| 08:36.30 | br00ksh1r3 | instead of cvs |
| 08:36.45 | Qwell | I dev at work, so...it's cvs for me |
| 08:37.03 | br00ksh1r3 | yah |
| 08:37.18 | br00ksh1r3 | you know.. nothing will stop us from doing svn + webdav too |
| 08:37.18 | YoMama | if you have context set in sip.conf for a sip proxy..there's no reason you can't use it in another context right? |
| 08:37.42 | Qwell | br00ksh1r3: it would be really nice... |
| 08:37.52 | Qwell | YoMama: you can include other contexts |
| 08:37.59 | Qwell | include => bobscontext |
| 08:38.01 | YoMama | i like svn + ssl + webdav |
| 08:38.10 | br00ksh1r3 | Qwell: you know you can do svn over anything right? |
| 08:38.16 | Qwell | br00ksh1r3: I do. :) |
| 08:38.29 | br00ksh1r3 | :D |
| 08:38.33 | Qwell | but, the server has to be able to transport it |
| 08:38.38 | Qwell | ala webdav |
| 08:38.40 | br00ksh1r3 | yeah |
| 08:38.47 | br00ksh1r3 | kram has my svn book, lol |
| 08:38.51 | br00ksh1r3 | i need to get it back :D |
| 08:38.55 | YoMama | Qwell: i know..but i got my sip proxy in context "internal"...but I have a context in my extensions.conf called 'incoming' trying to use the sip proxy |
| 08:39.05 | Qwell | br00ksh1r3: svn book is also creative commons, you know |
| 08:39.10 | br00ksh1r3 | yeah.. |
| 08:39.18 | br00ksh1r3 | but i hate pdfs |
| 08:39.26 | Qwell | think they have an html version |
| 08:39.33 | br00ksh1r3 | i want a paper copy |
| 08:39.36 | br00ksh1r3 | easier to read |
| 08:39.39 | Qwell | printer? :p |
| 08:39.51 | br00ksh1r3 | it would cost the same to print it out, lol |
| 08:40.06 | Vco | naw.. |
| 08:40.13 | Qwell | well, tell Mark to buy his own then, heh |
| 08:40.19 | Vco | set it to print 4 pages per sheet and duplex it |
| 08:40.20 | br00ksh1r3 | heheh |
| 08:40.21 | Vco | ;) |
| 08:40.31 | br00ksh1r3 | i just like books damnit! |
| 08:40.39 | Qwell | my boss printed the asterisk book...single sided, 8.5x11 |
| 08:41.11 | Vco | i did that with a ton of docs.sun.com stuff before i left one crappy job |
| 08:41.23 | Vco | like..binders and binders |
| 08:41.28 | Qwell | heh |
| 08:41.43 | YoMama | If you get a SIP response 400 "Bad Request", does that mean you sent it a bad phone #? |
| 08:41.49 | Vco | extra thick font cover, with transparent sheet |
| 08:41.56 | Vco | front and back rather.. |
| 08:41.56 | Qwell | YoMama: sometimes |
| 08:42.48 | YoMama | Qwell: and other times? |
| 08:44.24 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 08:44.25 | Qwell | Gnomes stealing your packets? |
| 08:44.33 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 08:44.42 | br00ksh1r3 | they only steal my socks :( |
| 08:49.30 | YoMama | ok...this really doesn't make any sense |
| 08:51.15 | *** join/#asterisk GiabboO (n=GiabboO@host101-246.pool8173.interbusiness.it) |
| 08:59.58 | oej | YoMama: 400 bad request has nothing to do with the phone #, there's something wrong in the SIP syntax. |
| 09:00.28 | YoMama | oej: but..it's identical...hold on..i have an idea |
| 09:00.31 | Qwell | oh, duh, 400. I was thinking 404 for some reason |
| 09:00.31 | endre | i like this: 400 "Bad Request - 'Media Negotiation Failed'" back from 172.30.50.3 |
| 09:01.11 | oej | Media negotiation failed should not generate a bad request. |
| 09:01.48 | YoMama | oej: can a bad username/password cause a 400? |
| 09:02.00 | oej | I don't think it should |
| 09:03.39 | YoMama | hmm...that's bizarre |
| 09:05.44 | YoMama | oej: the syntax is identical..just in two different contexts |
| 09:05.47 | YoMama | one works and one doesn't |
| 09:06.06 | oej | That's bizarre, confirmed |
| 09:06.12 | YoMama | -- Got SIP response 400 "Bad Request" back from x.x.x.x |
| 09:06.46 | YoMama | this thing says that the context designated in sip.conf is only the default one...it doesn't lock you into using that context for that extension.. |
| 09:07.33 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 09:08.34 | YoMama | screw it |
| 09:08.37 | YoMama | i'll figure it out tomorrow |
| 09:08.40 | YoMama | goodnight |
| 09:09.32 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
| 09:09.58 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:12.06 | zobia | Hello Qwell. |
| 09:12.18 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 09:12.19 | zobia | Can i ask u a realtime question? |
| 09:12.29 | cjk | hi, what do i have to change, to allow more than 500 channels |
| 09:13.13 | zobia | any one know can "s" work in the realtime extensions? |
| 09:16.18 | Qwell | zobia: sure it can |
| 09:18.04 | zobia | thanks Qwell. i insert one record to database priority is s. i watch on the asterisk , it did find it. but after that it generate the error. like == Starting Zap/1-1 at luo2,s,2 failed so falling back to exten 's' |
| 09:18.45 | *** join/#asterisk chapeaurouge (n=chap@85.201.80.249) |
| 09:18.46 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 09:19.15 | zobia | this record is a background() function. on the asterisk CLI i have seen it -- Executing Background("IAX2/sip1-16384", "custom/web/77/complaint_2") |
| 09:19.38 | zobia | but there's no voice. and right after this that error line came. |
| 09:26.26 | mover | monitor dodnt mix the in.wav + out.wav... what is went wrong? Mix: 1 is still supplied. |
| 09:33.57 | ikarus | Hmmm, anyone know a way to allow only a single call to a SIP phone (the phone will accept more calls, but on a certain extension I want it to be only able to handle one at a time) |
| 09:37.43 | oej | ikarus: Check call-limit in sip.conf or the groupcount set of applications and functions. |
| 09:38.01 | ikarus | group count ?, where would I find that |
| 09:39.23 | ikarus | oej: call limit won't work, that is on a per phone basis and the problem is, I need to do it on a per extension basis |
| 09:41.30 | oej | Check function GROUP, GROUP_COUNT and GROUP_LIST |
| 09:41.35 | zobia | hello Qwell |
| 09:43.37 | cjk | hi, has anyone an idea how to stress an asterisk with iax channels? i am able to open the channels, but somehow there is no traffic. |
| 09:44.03 | ikarus | oej: eesh, I am thinking I can skip on that feature |
| 09:45.29 | zobia | Qwell i added one wait(1) ahead of background . so it can find the s,1 properly now. but i just don't know why this can work. |
| 09:45.51 | zobia | i must add a wait(1) ahead of background always? |
| 09:49.48 | *** join/#asterisk insurin1 (n=root@82-42-19-166.cable.ubr01.knor.blueyonder.co.uk) |
| 10:12.42 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 10:13.02 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 10:13.17 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 10:20.03 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost, -- if anyone is interested send me a private message (i might be afk) i have a reference in the channel if you need. -- |
| 10:20.22 | iDunno | what is cost price? |
| 10:20.35 | iDunno | (being the most important question ;) |
| 10:20.46 | lme | is there any known problems with asterisk 1.2.0 & cdr ? I've got strange issues in my csv files, binaries insert |
| 10:21.40 | lme | notice that i've got the bristuff disease 0.3.0 |
| 10:21.59 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 10:24.14 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 10:24.50 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 10:27.49 | tzafrir_laptop | lme, better give examples of your problems |
| 10:28.26 | newl | heh yah, cost of the wholesaler, or cost as in your buy price? :) |
| 10:32.35 | lme | tzafrir : here is a line from my standard master.csv. Notice the "x]" : "interne","x]","121","infoclip","""136"" <x]>","Zap/26-1","SCCP/121-0000001d","Dial","SCCP/121|15|g","2005-11-22 10:44:31","2005-11-22 10:44:32","2005-11-22 10 |
| 10:32.49 | ikarus | Hmmm, anyone know a list of ISDN cards containing the HFC-S chipset ? |
| 10:33.21 | viperdude | hi all |
| 10:33.51 | viperdude | can any one debug this for me. |
| 10:34.15 | viperdude | when I try to pickup a call from a ringing extension, the cli reports |
| 10:34.23 | viperdude | Executing PickUp("SIP/669-36ef", "670@default") in new stack |
| 10:34.27 | viperdude | then |
| 10:34.36 | viperdude | WARNING[30479]: channel.c:3018 ast_get_group: Ignoring invalid group 670 (maximum group is 31) |
| 10:34.40 | viperdude | any ideas? |
| 10:34.53 | lme | viperdude: try to use dpkickup instead of pickup |
| 10:35.15 | lme | viperdude: pickup is for groups only, you canno't specify extension |
| 10:35.22 | viperdude | aha |
| 10:36.23 | viperdude | lme: can't find that function on voip-info |
| 10:36.36 | lme | i think it's new in 1.2 |
| 10:36.46 | viperdude | hmm |
| 10:36.58 | lme | try show application DPickup in your cli |
| 10:37.00 | viperdude | is there a solution for this in 1.0.9 |
| 10:37.01 | *** join/#asterisk pussfeller (n=todd@12.150.129.170) |
| 10:37.55 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 10:38.36 | viperdude | this seems to suggest you can specify exten http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup |
| 10:40.52 | lme | viperdude: pickup from asterisk or bristuff ?! |
| 10:41.15 | viperdude | hmm looks like its a patch |
| 10:41.43 | lme | ok... 'cause i'm not sure but i really think it's a new application |
| 10:43.46 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 10:47.53 | *** join/#asterisk pr0 (n=LANmower@ndn-165-137-37.telkomadsl.co.za) |
| 10:48.02 | viperdude | aha fixed it |
| 10:48.04 | pr0 | shit, that wasnt easy |
| 10:48.40 | viperdude | Needed PickUpChan(SIP/${EXTEN:2}) |
| 10:48.46 | pr0 | tell me, do I need to do anything special to execute php scripts from agi? or do I just need php... |
| 10:48.53 | viperdude | php cli |
| 10:49.03 | pr0 | got that, would that suffice? |
| 10:49.11 | viperdude | yes I use it all the time |
| 10:49.19 | viperdude | remember the -q switch |
| 10:49.19 | pr0 | kewl... I'll go try it out |
| 10:49.23 | pr0 | yeah |
| 10:50.39 | ikarus | hmmmmm, nasty buisness this is, I am using Early Dial, but some phone numbers have an unknown length, (international numbers), how on earth can I handle those sanely ? |
| 10:51.09 | lme | is there anybody using asterisk 1.2.0 with bristuff 0.3.0 patch ? |
| 10:51.11 | pr0 | ikarus, I guess different contexts for different starting codes |
| 10:52.31 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-19-69.44-151.net24.it) |
| 10:53.17 | ikarus | pr0: that isn't the problem, the problem is, with Early Dial, the phone just sends the number everytime you add a digit, making it in this case impossible to decide when to execute the Dial |
| 10:53.42 | pr0 | shite... |
| 10:54.03 | pr0 | um, you can make a variable that gets a number added to it every time. |
| 10:54.05 | pr0 | setvar |
| 10:54.20 | pr0 | then you know what length it is at all times. |
| 10:57.44 | ikarus | pr0: that doesn't solve the problem with the variable length of international numbers, although I guess you could make Asterisk pick up on extension 00 and let people dial into a weird menu, and if they get a timeout (4 seconds), or press # the call is placed (as is common with non Early Dial phones) |
| 10:58.09 | pr0 | no |
| 10:58.30 | ikarus | what no ? |
| 10:58.31 | pr0 | if the first few numbers are (international extension) go to international context and keep reading, simple. |
| 10:59.10 | pr0 | in the international context it behaves differently, taking more numbers. |
| 10:59.20 | ikarus | hmm ? |
| 11:00.56 | ikarus | What is "it" and what is would the international context be ? |
| 11:05.24 | Zeeek | does the phone have a send button? |
| 11:05.31 | pr0 | god |
| 11:05.41 | Zeeek | no it's just me |
| 11:06.01 | pr0 | look, you have two contexts, one for local (shorter nums) and one for international |
| 11:06.26 | pr0 | if it reaches a point where it could recognise the number as local, go there. same for international. |
| 11:06.34 | Zeeek | I don't have two contexts for local and int(l |
| 11:06.54 | pr0 | Zeeek, my bet is you dont have early dialing either. |
| 11:07.08 | Zeeek | are you trying to avoid using an int'l access code like 011 ? |
| 11:07.37 | Zeeek | yes I have most numbers setup in the digitmap |
| 11:07.38 | pr0 | no, he has a phone that dials a number every time you press a button, and wants to call both locally and internationally |
| 11:07.45 | Zeeek | Ewwwwww |
| 11:07.55 | Zeeek | no digitmap possible? |
| 11:08.00 | pr0 | ask him :) |
| 11:08.09 | Zeeek | NO DIGITMAP ? |
| 11:08.31 | Zeeek | ikarus what phone is my first question |
| 11:09.34 | ikarus | Zeeek: the phone has a send button, yes, it is a BudgeTone |
| 11:09.42 | *** join/#asterisk tobiasWolf (n=konversa@195.162.255.10) |
| 11:09.48 | Zeeek | so turn off early dial |
| 11:10.10 | ikarus | Zeeek: I can't, that looses usability |
| 11:10.26 | ikarus | I can't turn it on for certain prefixes either |
| 11:10.32 | Zeeek | I think there's no digitmap for BT101, I cabn'"t remember now |
| 11:10.33 | ikarus | So I will have to use it for everything |
| 11:10.49 | Zeeek | will you use an int'l access code like 011? |
| 11:10.56 | Zeeek | if so, it's easy |
| 11:10.57 | mut | meh |
| 11:10.58 | ikarus | "00" |
| 11:11.01 | ikarus | But yes |
| 11:11.06 | mut | my desk turned into the inventory room |
| 11:11.11 | *** part/#asterisk pussfeller (n=todd@12.150.129.170) |
| 11:11.12 | Zeeek | ok put 00 in an extension |
| 11:11.15 | mut | i'v got 20 ata's sittin on it this morning |
| 11:11.36 | ikarus | Zeeek: and then what ? |
| 11:11.40 | Zeeek | do a goto at that extension to a read |
| 11:11.45 | Zeeek | wait a sec |
| 11:11.58 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 11:12.24 | fourcheeze | I'm starting to see what's happening with my snom deregistration problem |
| 11:12.49 | fourcheeze | for some reason from time to time when the snom sends a REGISTER * doesn't reply within 5 minutes |
| 11:13.18 | fourcheeze | and the expiry on the session is 1 minute |
| 11:14.31 | fourcheeze | any ideas why the response from asterisk wouldn't be getting back, or a way around this problem? |
| 11:14.33 | RoyK | http://video.google.com/videoplay?docid=-2023320890224991194&q=loose+change |
| 11:17.58 | Zeeek | ikarus what are your local numbers like? fixed length? |
| 11:19.08 | ikarus | Zeeek: yes, local numbers are 10 digits starting with 0 or 7 digits, or 3 "digits" starting with * (internal use) |
| 11:20.44 | Zeeek | I'm trying to see if this could be adapted to yoiur situation: I use a code *# which goes to a playback() instruction. During that time the user hears music and can dial the rest of the number. As soon as a match isreached, the number goes out |
| 11:20.56 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 11:21.18 | nfi|ermes | http://pastebin.ca/30266 |
| 11:21.22 | nfi|ermes | agi problems |
| 11:21.33 | Zeeek | A variable is set right before playback to indicate a certain service is requested |
| 11:22.31 | ikarus | Zeeek: well, if you saw my suggestion for a solution above, making Asterisk pickup after 00 and allowing the user to enter the rest of the number into Asterisk, with a 4 second time out or # confirm, would seem like a sensible solution (hrm, is there any application to capture more then one digit, or will I have to use a creepy extension set ? |
| 11:25.09 | nfi|ermes | i found the soliution to my prroblem |
| 11:25.19 | nfi|ermes | asterisk-perl shold be installed |
| 11:26.20 | Zeeek | ikarus I didn't see it, sorry. I have to admit too, that most of my international calls are to the US so there's no variable length problem |
| 11:26.34 | nfi|ermes | i don t understand why agi works well in asterisk 1.0.9 without installing asterisk-perl |
| 11:26.35 | Zeeek | or europe |
| 11:26.43 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 11:27.02 | santoshr | has anyone ever in their lives got h323 |
| 11:27.09 | santoshr | call pickup to work with h323 |
| 11:27.19 | ikarus | Zeeek: in Europe there is a variable length problem :) |
| 11:27.34 | Zeeek | yeah but it can be mastered I think |
| 11:27.48 | santoshr | callpickup with h323 people.. |
| 11:29.10 | Zeeek | ikarus it's been a long time since I used my BT102 but not having a digitmap is a major shortcoming. Evcen my $60 chinese phones have that |
| 11:29.54 | Zeeek | can anyone call *any* country or is the set limited? |
| 11:30.20 | ikarus | Zeeek: well, the main 3 countries called are Germany, Belgium and the UK, but the UK is the main mess |
| 11:30.46 | Zeeek | can't you just use expressions in the dialplan? |
| 11:30.57 | Zeeek | _0041blah ? |
| 11:31.05 | fourcheeze | it's about time there was a european number format |
| 11:31.18 | Zeeek | there are actually european numbers |
| 11:31.21 | fourcheeze | why can't the EU commissioners do something useful about that |
| 11:31.32 | fourcheeze | yeah, but no-one seems to use them |
| 11:32.08 | Zeeek | http://www.acropolistelecom.com/index.php?rub=1,1,4,58 |
| 11:32.09 | ikarus | Zeeek: the . expression makes it go boom, as it matches as soon as the length is reached, even if that is not the entire phone number, then it fires off the Dial() application, chaos ensues |
| 11:32.28 | Zeeek | 3883x |
| 11:32.59 | Zeeek | ikarus, you know you can order the valuation of the dialplan using includes, right? |
| 11:33.05 | Zeeek | evaluation |
| 11:33.22 | ikarus | Zeeek: that is not the problem |
| 11:34.04 | Zeeek | I guess what I don't see is why won't _0033XXXXX. work ? |
| 11:34.08 | ikarus | the problem is that exten=> _0032. matches as soon as I have dialed 00321, instead of at the full number, which makes it dial prematurely |
| 11:35.00 | santoshr | callpickup with h323 people ??? does it work or not.. |
| 11:35.19 | Zeeek | ikarus ya but if you put the proper number of digits |
| 11:35.36 | Zeeek | santoshr we heard you the first 4 times. No one seems to know |
| 11:36.06 | santoshr | thank you for atleast sending a ack. i respect tht |
| 11:36.13 | ikarus | Zeeek: in many countries you have dynamic length numbers |
| 11:36.25 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 11:36.34 | Zeeek | with variable length numbers do,'t they begin differently for different lengths? |
| 11:36.50 | Zeeek | in France it's fixed at 10 |
| 11:36.52 | Zeeek | BUT |
| 11:37.09 | ikarus | Zeeek: even here locally that is an issue, 0800 and 0900 numbers can have a length between 4 and 8 after the code, and there is no way to know which is which |
| 11:37.17 | Zeeek | locally they use many 4 digit numbers which screws up our local internal extensons :( |
| 11:37.31 | Zeeek | are you sure there's no way? |
| 11:38.04 | ikarus | nope, I asked the telco |
| 11:38.14 | Zeeek | makes getting another phone an attractive alternative |
| 11:38.36 | ikarus | Zeeek: not moneywise, I was lucky to be able to order this one |
| 11:39.26 | ikarus | And actually, firing off Dial too early might work (overlap dial), but then logging is off (not a big issue, but still) |
| 11:39.36 | Zeeek | tough question indeed, in that case |
| 11:40.12 | ikarus | But the methode with having asterisk act like a non-Early Dial phone sould work, right ? |
| 11:40.17 | ikarus | it might not be very pretty |
| 11:40.20 | Zeeek | ah with no logging you could just do 00 Dial(ZAP/1) |
| 11:40.31 | Zeeek | that give a dialton |
| 11:40.32 | Zeeek | e |
| 11:40.35 | *** join/#asterisk kink0 (n=kinko@pluton.interec.com) |
| 11:40.41 | kink0 | buenos dias |
| 11:41.47 | ikarus | Zeeek: erm, I don't have ZAP, or even analog lines, I use ISDN and bristuff (and in the future vISDN probably) |
| 11:41.48 | iDunno | erm - are they analogue phones? because sid phones will generally not try to dial till you tell it to, and so you'll get the full number |
| 11:42.02 | iDunno | with analogue phones, you up the timeout and tell it to be lazy, IIRC. |
| 11:42.04 | ikarus | iDunno: they are SIP phones with Early Dial |
| 11:42.14 | iDunno | turn off early dial, then ;) |
| 11:42.16 | Zeeek | or maybe Dial(ZAP/1/00)ok, ISDN prolly won't allow what I just said |
| 11:42.30 | Zeeek | haha you'll be hearing the same suggestions all day :) |
| 11:42.51 | iDunno | though, I'd expect that you turn on the lazyness option giving time |
| 11:42.59 | iDunno | which I'm sure is available ;) |
| 11:43.01 | ikarus | iDunno: lazyness ? |
| 11:43.08 | Zeeek | no digitmap in that phone |
| 11:43.43 | iDunno | well, timeout type option so that it doesn't believe it's got the full number till it doesn't recieve one for $period_of_time |
| 11:44.16 | ikarus | iDunno: there is no digitmap or any of that stuff for the BudgeTones |
| 11:44.58 | iDunno | which budgetone? |
| 11:45.00 | iDunno | the 101? |
| 11:45.05 | ikarus | iDunno: yep |
| 11:45.21 | santoshr | lme: u arounnd |
| 11:45.22 | iDunno | erm - we're using those - addmittedly, I've turned off early dial ;) |
| 11:45.35 | iDunno | but if you hit # at the end of dialling, it sends the number \o/ |
| 11:45.36 | Zeeek | I personally like having a send button |
| 11:45.52 | ikarus | iDunno: my lusers would kill me if they had to press send or wait 4 seconds |
| 11:46.04 | ikarus | (or #) |
| 11:46.29 | iDunno | ikarus: mine complained that it was dialing to quickly, and they hadn't finished typing the number \o/ |
| 11:46.40 | iDunno | and the phone had a 3 second timeout :/ |
| 11:46.47 | Zeeek | that's my problem when I'm reading the number off a paper |
| 11:47.01 | iDunno | users want the moon on a stick, you make compromises. |
| 11:47.25 | iDunno | you *tell* them how it works, and explain the alternatives, and why you've chosen that way. |
| 11:47.28 | iDunno | they deal. |
| 11:47.33 | iDunno | simple sys-admin job ;) |
| 11:47.41 | Zeeek | heh |
| 11:48.26 | mut | you need more of a bofh view |
| 11:48.40 | ikarus | iDunno: well, making Asterisk pick up the line and handle emulating a non-Early Dial phone for such annoying numbers should be a nice alternative |
| 11:49.05 | ikarus | it allows people to dial national numbers without any issue, and international, 0800 and 0900 are handled via Asterisk |
| 11:49.20 | iDunno | well, you *could* make it so that the shortest number picked up and waited to see if more digits where dial using a read. |
| 11:49.38 | *** join/#asterisk robin_sz (n=robin@212.243.40.130) |
| 11:49.40 | iDunno | then forward the call to Local/fullnumber |
| 11:49.57 | ikarus | iDunno: well, that sounds like a good plan |
| 11:50.01 | iDunno | using the read data, or otherwise timeout and dial the short number that they dialled. |
| 11:50.22 | robin_sz | OK, internal-to-internla I have working .. but the CAPI interface is not ... |
| 11:50.24 | iDunno | (think *round* the problem ;) |
| 11:50.41 | iDunno | (alternatively, just kill all the users, they're the real problem ;) |
| 11:50.41 | kink0 | there anyway to use USB/mobile phone modem instead serial modem ? ( I got errors in communication even after to hack chan_modem.c and change from 115200 to 9600 ) |
| 11:50.59 | robin_sz | I get Executing Dial("SIP/790-9286", "CAPI/contr1/004412341234}") in new stack |
| 11:51.01 | ikarus | iDunno: well, there is a tiny problem with that, read doesn't have a timeout option, only a max length and a # terminator |
| 11:51.22 | robin_sz | which looks OK ?? .. what does the "contrl" field mean?? |
| 11:52.05 | robin_sz | anyway ... it then falls through as NOANSWER |
| 11:54.39 | iDunno | what's wrong with DigitTimeout and ResponseTimeout? |
| 11:54.57 | iDunno | ikarus: can you please read some of the documentation and examples? ;) |
| 11:55.03 | iDunno | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Read |
| 11:55.26 | ikarus | iDunno: I was reading there |
| 11:55.57 | ikarus | But I missed a bit :) |
| 11:56.00 | iDunno | :) |
| 11:56.31 | darkskiez | folks at my company want to know why skype sounds much better than x-lite/asterisk? what do i tell them? |
| 11:57.07 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
| 11:57.15 | iDunno | that you're using the wrong codec and that you want more bandwidth. |
| 11:57.18 | robin_sz | tell them to use a different codec on asterisk? |
| 11:57.38 | iDunno | use (u|a)law, it's high bandwidth, but nice :) |
| 11:57.54 | darkskiez | i heard skype used ilbc, so i've tried that |
| 11:57.59 | darkskiez | dont have a high bandwidth |
| 11:58.12 | robin_sz | darkskiez, it is even possible that Skype use asterisk as a back end anyway .. modified for their protocol .. |
| 11:58.13 | ikarus | darkskiez: it might use 16khz instead of 8 |
| 11:59.04 | iDunno | it might also be entirely non-standard and evil ;) |
| 11:59.13 | iDunno | (but that might just be my view of skype) |
| 11:59.16 | darkskiez | I know it is |
| 11:59.17 | Zeeek | exit |
| 11:59.20 | Zeeek | not. |
| 11:59.45 | iDunno | wrong window? ;) |
| 11:59.51 | Zeeek | yup |
| 12:00.04 | Zeeek | the joys of two screens |
| 12:00.23 | Zeeek | why can't it just track which one I'm looking at? |
| 12:01.05 | mut | hmm |
| 12:01.12 | mut | radius reported the DSL speed as |
| 12:01.14 | mut | 155520000bps |
| 12:01.18 | darkskiez | its about time cisco released a new firmware for the 79[64]0, its been over 4 months since the last one, they seem to release them quarterly. |
| 12:01.21 | Zeeek | whatever I do, it seems like an iaxy will not pickup anoother line with *8 |
| 12:01.29 | mut | seems a lil high O_O |
| 12:02.10 | mut | must be as fast as the atm can go |
| 12:02.12 | darkskiez | well not quarterly, but the last few have been 4 months apart. |
| 12:03.09 | darkskiez | mut: 155Mbit is SONET |
| 12:03.24 | darkskiez | STM-1 |
| 12:05.14 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:05.48 | puzzled | morning |
| 12:06.06 | mut | yeh |
| 12:06.10 | mut | like i said |
| 12:07.31 | mut | omg isn't it annoying when you're listening to a song |
| 12:07.41 | mut | then they switch the words to only the right or left ear |
| 12:07.45 | mut | but music is in both |
| 12:07.53 | mut | drives me nuts |
| 12:07.57 | iDunno | ikarus: right - so, anyways - does that work for you? is that a better solution? ;) |
| 12:08.37 | *** join/#asterisk juanjoc (n=jcomella@40-115-235-201.fibertel.com.ar) |
| 12:11.05 | ikarus | iDunno: I am busy investigating, but I think it is a suitable solution |
| 12:12.19 | juanjoc | Does anybody know if there's a way to stop monitoring a channel without hanging up when using MixMonitor? |
| 12:14.01 | *** join/#asterisk shmooz (n=shmooz@H142.C72.B0.tor.eicat.ca) |
| 12:14.13 | *** join/#asterisk DYOGI_B (n=Jade@dsl-202-173-190-245.qld.westnet.com.au) |
| 12:14.19 | DYOGI_B | hi from the gold coast |
| 12:14.35 | shmooz | yo |
| 12:14.54 | ikarus | iDunno: it seems to be working, so I guess so |
| 12:15.23 | mut | holy crap! |
| 12:15.24 | mut | http://video.google.com/videoplay?docid=5854686068870249151 |
| 12:16.04 | ikarus | iDunno: the long time out I will say is due to the international nature *cough* |
| 12:16.24 | DYOGI_B | anyone know how to update amp? |
| 12:16.36 | znoG | damnit, asterisk distinctive ring has stopped working.... |
| 12:16.40 | iDunno | heh - you could always tell them the truth... they might not understand it, but it'd sound plausable ;) |
| 12:17.22 | DYOGI_B | so is asterisk 1.2.0 stable now??? |
| 12:17.55 | *** join/#asterisk thedingaling (n=jjones@216.70.38.210) |
| 12:18.25 | ikarus | iDunno: ah, stack overflow |
| 12:19.19 | mut | anyone watching this? |
| 12:19.37 | DYOGI_B | anyone know how to update amp? |
| 12:19.37 | DYOGI_B | so is asterisk 1.2.0 stable now??? |
| 12:20.05 | bon | yes. |
| 12:20.10 | Zeeek | damned echo |
| 12:20.11 | DYOGI_B | thx |
| 12:22.41 | Zeeek | Currently, the playback feature of Google Video isn't available in your country. |
| 12:22.46 | Druken | typical trigger happy redneck americans |
| 12:22.51 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 12:23.18 | *** join/#asterisk ful|work (n=fulgas@209.8.233.201) |
| 12:24.26 | Druken | Zeeek; what echo ? |
| 12:24.42 | Zeeek | the repeated lines above... |
| 12:24.55 | Druken | oh that echo |
| 12:24.57 | Druken | echo |
| 12:25.02 | Zeeek | btw is it better to use the more agressive anti-echo? I've never tried it |
| 12:25.02 | mut | heh |
| 12:25.02 | Druken | :) |
| 12:25.17 | mut | it's an awesome bid |
| 12:25.18 | mut | vuid |
| 12:25.22 | mut | vid*! |
| 12:25.23 | Zeeek | aslo, while I'm asking, where is the doc about ae ? |
| 12:25.26 | astcryz | Nov 22 14:25:18 NOTICE[16107]: chan_sip.c:10815 handle_request_register: Registration from '21990401<sip:21990401@21990401>' failed for '80.213.142.220' - Username/auth name mismatch |
| 12:25.28 | ikarus | Anyone know if Dell uses Asterisk, if so, they need a more competent admin..... |
| 12:25.31 | astcryz | with asterisk realtime |
| 12:25.33 | ikarus | :P |
| 12:25.33 | astcryz | anyone know what i can do? |
| 12:25.57 | Druken | astcryz: how about read the fucken error... |
| 12:26.04 | Druken | Username/Auth mis match |
| 12:26.21 | astcryz | yes, but i cant find any faults |
| 12:26.25 | Zeeek | oh I see extensions.ael |
| 12:26.45 | astcryz | Druken: there is no mismatch |
| 12:26.49 | Druken | astcryz: using plaintext or md5 ? |
| 12:26.53 | astcryz | plaintext |
| 12:27.22 | Zeeek | the context is the same as the number dialed? |
| 12:28.04 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.37.169.64) |
| 12:28.08 | MuppetMaster | Hello |
| 12:28.18 | *** join/#asterisk cjk_ (n=cjk@80.92.64.103) |
| 12:28.18 | MuppetMaster | Anyone hear know much about Sendmail in conjunction with Asterisk? |
| 12:28.42 | Zeeek | what about it? I have it running |
| 12:28.44 | cjk_ | hi, is there a way to log cdrs from asterisk into more than on mysql server? |
| 12:28.55 | MuppetMaster | Well, it use to be I would install SuSE v9.3 with Sendmail and all worked fine. |
| 12:29.02 | Zeeek | and now? |
| 12:29.06 | MuppetMaster | But with SuSE v10, Asterisk is no longer sending email withvmail. |
| 12:29.14 | ikarus | iDunno: thanks, the trick seems to be working nicely |
| 12:29.21 | MuppetMaster | Based on the default Sendmail configs on SuSE 10. |
| 12:29.45 | Zeeek | MuppetMaster with aux is sendmail listening ? |
| 12:29.52 | MuppetMaster | I have also installed ssmtp to try and use that, and while I may send emails on the command line, Asterisk still does not, even if I have this in the voicemail.conf. |
| 12:29.56 | MuppetMaster | I will check on aux. |
| 12:30.19 | *** join/#asterisk grimse (n=grimse@p5481FA5E.dip.t-dialin.net) |
| 12:30.21 | iDunno | ikarus: np - it comes of having to work round people too often :) |
| 12:30.26 | MuppetMaster | Yes, as I am able to send an email using Sendmail as a relay with ssmtp |
| 12:30.44 | Zeeek | oops I meant ps -aux but i think you got it |
| 12:30.45 | ikarus | iDunno: it is actually a pretty quick fix, and it makes all logging, etc work as expected |
| 12:30.47 | DYOGI_B | how can we block extension 201 from calling international but not block exp 333 |
| 12:31.00 | MuppetMaster | Yes, Sendmail is listening |
| 12:31.07 | MuppetMaster | And in fact relaying |
| 12:31.22 | ikarus | DYOGI_B: use /333 in the dialplan after the extension(match) |
| 12:31.24 | Zeeek | and no messages when asterisk gets a vmail? |
| 12:31.30 | MuppetMaster | No |
| 12:31.41 | Zeeek | so asterisk isn't even trying... |
| 12:31.42 | viperdude | DYOGI_B: put them in different contexts, one that allows international and another that doesn't |
| 12:32.01 | MuppetMaster | Well, I believe it is, but not sure which Sendmail log file to look at. |
| 12:32.03 | Zeeek | hey maybe sendmaiul moved? |
| 12:32.08 | MuppetMaster | Could be |
| 12:32.13 | MuppetMaster | That is an idea to check |
| 12:32.13 | Druken | anyone know how to stop the stupid email from saying it's coming from root? |
| 12:32.17 | Zeeek | perhaps /var/log/maillog ? |
| 12:32.34 | MuppetMaster | I will take a look |
| 12:32.50 | Zeeek | not sure of the directories on linux for mail |
| 12:33.06 | MuppetMaster | That is my problem too...always tried to avoid the fiddly world of Sendmail. |
| 12:33.08 | viperdude | Druken: set serveremail in voicemail.conf |
| 12:33.10 | MuppetMaster | Will drive one nuts. |
| 12:33.25 | Druken | viperdude: don't do shit... |
| 12:33.36 | Zeeek | on my linux box it is indeed at /var/log/maillog just as in FreeBSD |
| 12:33.37 | LostFrog | serveremail= in voivemail.conf |
| 12:33.43 | LostFrog | Damn.. I'm slow. |
| 12:33.54 | Druken | LostFrog: again, don't work |
| 12:33.56 | MuppetMaster | I think I just sorted with ssmtp! |
| 12:34.02 | MuppetMaster | Indeed, things had moved around. Thanks Zeek! |
| 12:34.25 | Zeeek | hey, even a stopped clock is right twice a day! It was my turn :) |
| 12:34.31 | viperdude | Drunken: have you uncommented the line? |
| 12:34.43 | Druken | yes, i'm not a total moron... |
| 12:34.53 | Druken | only sometimes :) |
| 12:34.56 | Zeeek | is that oepn for discussion? |
| 12:34.59 | viperdude | ok :-) |
| 12:35.11 | viperdude | are you running * as root? |
| 12:35.19 | Druken | Zeeek: you can openly discuss that one in RL ) |
| 12:35.25 | tzafrir_laptop | what MTA is it? |
| 12:35.32 | Druken | viperdude: yes.. i know i shouldn't but i do.. |
| 12:35.35 | BladeRunner05 | Wich is a valid replacemente of chan_capi with asterisk 1.2.0 ? |
| 12:35.36 | Druken | tzafrir_laptop: qmail |
| 12:35.45 | viperdude | try another user, MTA could be rewriting headers |
| 12:36.16 | Zeeek | Druken are you restarting between changes? |
| 12:36.28 | Druken | Zeeek: yes.. |
| 12:36.43 | Druken | i've even rebooted the box since... |
| 12:36.45 | Zeeek | cause I use serveremail and it works and * runs as root |
| 12:36.56 | viperdude | me too |
| 12:37.01 | Zeeek | check the spelling again too |
| 12:37.11 | Zeeek | it looks like "Sever email" |
| 12:37.16 | Zeeek | it looks like "Severe email" |
| 12:37.19 | BladeRunner05 | chan_capi won't compile using asterisk 1.2.0, suggest me a good replacement of it |
| 12:37.29 | *** join/#asterisk RoyK (n=roy@213.160.242.69) |
| 12:38.07 | Druken | ; Who the e-mail notification should appear to come from |
| 12:38.07 | Druken | serveremail=ABSS Networks <voip@abss.ca> |
| 12:38.33 | LostFrog | Umm.. it should be serveremail=voip@abss.ca |
| 12:38.36 | Zeeek | try just 'voip' |
| 12:38.43 | viperdude | have you tried serveremail=voip@abss.ca? |
| 12:38.50 | Druken | don't think so.. |
| 12:39.01 | LostFrog | and fromstring=ABSS Networks |
| 12:40.02 | Druken | k, i'll give that a try |
| 12:40.41 | *** join/#asterisk Igbothom_III (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
| 12:41.11 | DYOGI_B | thanks |
| 12:41.22 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.37.169.64) |
| 12:41.23 | astcryz | RoyK: Hvordan fĺr du til samtale venter? |
| 12:42.01 | Druken | thanks for helping out a complette moron :) |
| 12:45.46 | DYOGI_B | anyone know how to update Asterisk@home Beta -> asterisk 1.2 full? |
| 12:45.59 | fugitivo | yes |
| 12:46.08 | DYOGI_B | can you please tell me |
| 12:46.22 | fugitivo | reinstall with a normal distro, and install asterisk 1.2 |
| 12:46.27 | LostFrog | lol |
| 12:46.39 | DYOGI_B | very useful |
| 12:46.53 | DYOGI_B | guess you have no idea then |
| 12:47.03 | Zeeek | why not ask the @home folks |
| 12:47.17 | DYOGI_B | is there a channel on IRC? |
| 12:47.26 | Zeeek | I think so |
| 12:47.31 | Zeeek | but don't ask the name |
| 12:48.12 | LostFrog | Zeeek: what channel? |
| 12:48.12 | LostFrog | lol |
| 12:48.24 | Zeeek | Ask not |
| 12:48.34 | Zeeek | maybe I'm confusing it with AMP? |
| 12:52.28 | ikarus | Hmmmmm, anyone have an advice on a simple ATA (it will NEVER be used for outbound dialing, it is just so that an old anwering machine can stay connected) ?, or is any ata from Sipura or Grandstream good enough ? |
| 12:52.45 | fugitivo | answering machine?? |
| 12:52.54 | Zeeek | DYOGI_B http://sourceforge.net/forum/forum.php?thread_id=1386419&forum_id=420323 |
| 12:53.06 | DYOGI_B | thank you |
| 12:53.12 | kink0 | off-topic: all mobile phones when are connected to the USB uses ONLY 9600 bps ? |
| 12:53.14 | fugitivo | ikarus: why you want an answering machine if you can have voicemail?? |
| 12:53.19 | Zeeek | ikarus why do you want an old answering machine? |
| 12:53.23 | Zeeek | yeah WHY? |
| 12:53.28 | Zeeek | huh? huh? |
| 12:53.29 | ikarus | fugitivo: because an answering machine is more intuitive to use |
| 12:53.37 | ikarus | (NOT MY DECISION) |
| 12:53.40 | fugitivo | oh god |
| 12:53.55 | Zeeek | what part of "press 1" do they not get? |
| 12:54.02 | fugitivo | lol |
| 12:54.30 | ikarus | Zeeek: the fact that even with the Dutch language files they still find it more difficult then having the buttons play, delete and previous message |
| 12:54.35 | Zeeek | that's a new one. Run asterisk and connect an answering machine to it. Well, why not? asterisk can do anything :) |
| 12:54.50 | fugitivo | that's really funny |
| 12:54.59 | fugitivo | first time i see a case like yours :) |
| 12:55.06 | ikarus | fugitivo: it is bloody annoying |
| 12:55.07 | Zeeek | ikarus I understand and yet am tempted to say, "GET A LIFE!" those neurons need to be excercised fromtime to time! |
| 12:55.25 | Zeeek | any IDIOT can learn to use vmail |
| 12:55.35 | fugitivo | i understand using a fax machine, but an answering machine? that's really funny, lol |
| 12:55.41 | ikarus | Zeeek: I am not going to risk it |
| 12:55.46 | Zeeek | haha |
| 12:56.08 | Zeeek | I even heard you can fast rewing fast forward with the newest version |
| 12:56.09 | ikarus | my boss specificly said to keep it as it would keep the changes to a minimum even with my annoyance |
| 12:56.26 | Zeeek | well maybe you can wean them over after a while |
| 12:56.52 | ikarus | Zeeek: well, for any other extension vmail will be used |
| 12:57.04 | Zeeek | oh it's only for the boss? |
| 12:57.39 | many | for the 60y.o. secretary who complained that the telephone talks to her |
| 12:57.54 | ikarus | Zeeek: well, only for the secretary who normally answers the phone |
| 12:58.38 | many | never change their workflow or they will kill you |
| 12:58.59 | ikarus | many: exactly |
| 12:59.10 | ikarus | many: although this one is only 40 |
| 12:59.15 | Zeeek | yeah I went through that when CAD was introduced in the shop |
| 12:59.16 | ikarus | (and was just annoying me) |
| 12:59.22 | fugitivo | ok |
| 12:59.25 | fugitivo | this is really funny |
| 12:59.42 | Zeeek | ok, here's the solution: |
| 12:59.46 | fugitivo | ikarus: the linksys pap2-na is a cheap ata |
| 12:59.57 | fugitivo | ikarus: and it works well |
| 13:00.03 | ikarus | fugitivo: price on that one ? |
| 13:00.10 | Zeeek | Have asterisk call the answering machine and leave the message :) |
| 13:00.21 | fugitivo | ikarus: aprox $60 |
| 13:00.26 | *** join/#asterisk Ansonmus (n=ahaeser@a213-84-26-148.adsl.xs4all.nl) |
| 13:00.32 | fugitivo | ikarus: and it's 2 fxs |
| 13:00.53 | many | "hi, this is your asterisk. some one called, but i just forgot their number. they even left a message, but alas, i also forgot that one. Call ya later!" |
| 13:00.53 | Zeeek | so you can add the fax |
| 13:01.05 | ikarus | fugitivo: the less features the better |
| 13:01.27 | Zeeek | "you have a messages fom one,zero,zero,three,four,one,five at one oh three P M. |
| 13:01.32 | fugitivo | ikarus: it's the cheaper you can get |
| 13:02.19 | ikarus | Zeeek: heh, I actually wish I could hack the old answering machine into being just a front end for asterisk |
| 13:02.53 | ikarus | fugitivo: erm, is that thing even availible in Europe |
| 13:03.05 | fugitivo | ikarus: it should |
| 13:03.39 | fugitivo | ikarus: why don't you use a telephone and program it to look like the answering machine? so you can use the voicemail |
| 13:04.11 | ikarus | fugitivo: well, I won't force the change now, btw, Linksys The Netherlands claims they don't have it on their product site |
| 13:04.21 | Zeeek | where would you out the cassette? |
| 13:04.27 | fugitivo | lol |
| 13:04.30 | fugitivo | cassette |
| 13:04.34 | LostFrog | Zeek: I have a few suggestions. |
| 13:04.35 | LostFrog | :) |
| 13:04.38 | fugitivo | it isn't digital? |
| 13:04.41 | Zeeek | oh, it's a tape? |
| 13:04.52 | ikarus | Zeeek: it is a digital answering machine (which adds to stupidity) |
| 13:05.06 | Zeeek | so you can't fast forward? |
| 13:05.08 | ikarus | But this secretary is a bit erm, slow and technofoob |
| 13:05.11 | ikarus | Zeeek: you can |
| 13:05.22 | Zeeek | ooohhhh that's nice |
| 13:05.23 | fugitivo | why don't they fire the secretary? |
| 13:06.11 | ikarus | fugitivo: she does the job she is hired for |
| 13:06.20 | Zeeek | all joking aside, I do understand the issue. It is a little disconcerting though for something so trivial as an answering machine |
| 13:06.28 | fugitivo | but she can't adapt to new things |
| 13:06.29 | LostFrog | So.. for all practial purposes, she is useing voicemail. |
| 13:06.43 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost, -- if anyone is interested send me a private message (i might be afk) i have a reference in the channel if you need. -- |
| 13:06.46 | LostFrog | practical.. |
| 13:06.50 | fugitivo | bbz: me! lol |
| 13:06.58 | LostFrog | bbz: the number was 30 last week. :) |
| 13:07.26 | ikarus | LostFrog: except for the UI |
| 13:07.43 | ikarus | the answering machine doesn't require a voice to tell you to press what button |
| 13:07.53 | fugitivo | so the voice is the problem? |
| 13:07.59 | Zeeek | after the first couple times, you din't listen to the voice |
| 13:08.03 | fugitivo | she's scared of the voice? |
| 13:08.35 | ikarus | fugitivo: nah, it just isn't that it takes more time to listen to voice prompts then it is to look at 5 buttons |
| 13:08.58 | LostFrog | It does? |
| 13:09.11 | LostFrog | I hear.. 'Pr.." and hit 7. |
| 13:09.21 | ikarus | LostFrog: there are some people.... |
| 13:10.29 | ikarus | Anyway, no reason to break someones routine |
| 13:10.44 | fugitivo | if it' |
| 13:10.58 | fugitivo | if it's for better, yes there's a reason |
| 13:11.16 | *** join/#asterisk KampfCaspar (n=KampfCas@194.230.72.250) |
| 13:11.22 | KampfCaspar | Hi everyone! |
| 13:11.23 | ikarus | fugitivo: vmail doesn't give an actual improvement, except maybe that message can be stored indefinatly (which is not needed |
| 13:11.41 | astcryz | Nov 22 15:10:10 NOTICE[16107]: chan_sip.c:10815 handle_request_register: Registration from '21990401<sip:21990401@asterisk>' failed for '80.213.133.68' - Username/auth name mismatch |
| 13:11.41 | Zeeek | and forwarded |
| 13:11.45 | astcryz | but i cannot see any mismatch in asterisk |
| 13:11.46 | astcryz | :( |
| 13:11.51 | mut | speaking of voicemail |
| 13:11.54 | astcryz | nor the config files/sql files |
| 13:12.01 | mut | i need to create a script to remove old users voicemailboxes |
| 13:12.03 | fugitivo | ikarus: you have folders, you can redirect the vm to other user, you can make an interface to see the vm from the web |
| 13:12.15 | many | you save wires. |
| 13:12.15 | mut | so new users don't get the old voicemails |
| 13:12.17 | many | you save a blackbox |
| 13:12.21 | fugitivo | ikarus: http://www.voip-info.org/wiki-Linksys |
| 13:13.00 | fugitivo | ikarus: be sure you get the -NA, it's unlocked |
| 13:13.01 | LostFrog | mut: rm -rf ? |
| 13:13.09 | mut | yea |
| 13:13.17 | mut | but my user and voicemail configurations are updated dynamically |
| 13:13.21 | ikarus | fugitivo: erm, those things are not needed |
| 13:13.32 | mut | so i need to script it in there to remove old mailboxes somehow |
| 13:13.41 | fugitivo | ikarus: maybe a web interface for your vm will be easy to use for the secretary |
| 13:13.56 | LostFrog | lol.. then her computer will talk to her. |
| 13:14.00 | fugitivo | lol |
| 13:14.12 | ikarus | fugitivo: nah, computer doesn't even have speakers |
| 13:14.32 | mut | <3 headphones |
| 13:14.41 | mut | zone out and code at work |
| 13:14.46 | ikarus | fugitivo: ah, Sipura 2000 is essentially identical (and still availible here |
| 13:14.51 | fugitivo | ikarus: thanks god, or the boss with ask you to replace the computer with an old writing machine |
| 13:15.19 | ikarus | fugitivo: *shudder* |
| 13:15.26 | fenlander | fugitivo,ikarus: iirc the pap2-na is the north american version, we have the pap2-uk in the uk - maybe something similar in the netherlands? |
| 13:15.30 | LostFrog | Maam, ok.. here is your ream of parchment.. here is your quill.. and your bottle of ink. |
| 13:15.45 | fugitivo | fenlander: really? -uk version? didn't know that |
| 13:15.55 | fugitivo | LostFrog: lol |
| 13:16.19 | LostFrog | Install a telegraph wire to her office and have asterisk signal in morse code. |
| 13:16.21 | robin_sz | I am still having CAPI problems ... > CAPI INFO 0x3302: Protocol error layer 2. im in Switzerland .. anyhting "special" I should know about capi.conf?? |
| 13:16.28 | ikarus | fenlander: let's see |
| 13:16.42 | ikarus | fugitivo: the worst bit, I would have to support it aswell |
| 13:16.46 | astcryz | using realtime asterisk here? |
| 13:16.54 | fugitivo | is she using the old telephone that doesnt have buttons? the ones with a disc to dial |
| 13:17.04 | fugitivo | astcryz: not me |
| 13:17.08 | *** join/#asterisk seong (i=seong@60.50.118.174) |
| 13:17.09 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 13:17.38 | ikarus | fugitivo: nah, a very old ISDN phone (I am really afraid how much training it will take to get her to use the new phone |
| 13:17.38 | LostFrog | fugitivo: I still have a rotary testset. |
| 13:17.49 | robin_sz | sigh .. or .. does anyone else have Eicon Diva BRI working? |
| 13:17.53 | *** join/#asterisk jcath (n=skycat@221.218.12.129) |
| 13:18.23 | fugitivo | LostFrog: l337 |
| 13:18.35 | ikarus | robin_sz: which, that name is used on I think a dozen models |
| 13:18.51 | robin_sz | wait ... I find out :) |
| 13:18.55 | LostFrog | Hey.. it helps in premise wiring. I'd be dead without it. |
| 13:19.18 | robin_sz | ikarus, Eicon Networks Corporation Diva Server 4BRI-8M Rev 2 (rev 01) |
| 13:20.43 | ikarus | robin_sz: I heard people got it working |
| 13:21.19 | ikarus | fenlander: hmmmm, they just have one called PAP2 here, no idea what kind of sillyness there is |
| 13:22.02 | LostFrog | PAP just sounds too feminine and painful. |
| 13:22.19 | LostFrog | What are they going to name their next one? HPV? |
| 13:23.51 | ikarus | Wish they had the good old Sipura 2000 (or 1000), I have used those before |
| 13:24.11 | fugitivo | ikarus: sipura and linksys is the same, both cisco :) |
| 13:24.20 | ikarus | fugitivo: I know |
| 13:24.30 | *** part/#asterisk KampfCaspar (n=KampfCas@194.230.72.250) |
| 13:24.34 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 13:24.40 | ikarus | fugitivo: but Sipura is being intergrated with Linksys |
| 13:24.41 | many | robin: its not contrl, but contr1, which should be the name of your controller in chan_capi.conf |
| 13:25.24 | ikarus | ah, see here Linksys |
| 13:25.25 | ikarus | PAP2-EU |
| 13:25.52 | fugitivo | pap2 is nice, i have 2 installations with a lot of pap2 |
| 13:26.05 | Zeeek | try the new Linksys Pr0-state router |
| 13:26.35 | fugitivo | all this work for an answering machin |
| 13:26.37 | fugitivo | lol |
| 13:26.49 | robin_sz | ikarus, I dont suppose you heard HOW they got it working?? |
| 13:27.02 | ikarus | robin_sz: nope |
| 13:27.21 | many | robin: do you see incoming calls hitting asterisk in the log? |
| 13:27.35 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 13:27.41 | ikarus | fugitivo: I sure hope they are greatful |
| 13:27.42 | robin_sz | nope .. but there again .. I m,ight not have any .. Im tryingot get outgoing working first .,... |
| 13:27.49 | many | no |
| 13:27.59 | ikarus | I'll probably be here around the christmas holiday installing all this mess in the final setup |
| 13:28.02 | ikarus | gah |
| 13:28.09 | many | call yourself (you have a mobile, dont you?) and check wether the call hits the console log |
| 13:28.23 | many | you may need to raise the debuglevel first. |
| 13:28.50 | Zeeek | fortunately the lusers email clients don't have the "check every" parameter in ms |
| 13:28.56 | Zeeek | or they'd have it set at 1ms |
| 13:29.14 | robin_sz | many, easier said than done ... we have 3 BRI and Im not sure how each one gets rung ... I may need to diss off the old exchange first ... |
| 13:29.21 | many | argh. |
| 13:29.30 | robin_sz | footgun? |
| 13:29.37 | nfi|ermes | help please |
| 13:29.39 | nfi|ermes | http://pastebin.ca/30290 |
| 13:29.40 | many | usually they are rung round robin |
| 13:29.46 | robin_sz | hmmm |
| 13:29.52 | ikarus | Waaaaaaay too much to do |
| 13:29.54 | many | btw, did you configure pmp/ptp correctly? |
| 13:30.03 | nfi|ermes | i ve some problems with dialparties.agi |
| 13:30.04 | robin_sz | dunno? did I? |
| 13:30.06 | clive- | robin where you from? |
| 13:30.15 | robin_sz | Im in Geneva |
| 13:30.48 | clive- | ok:), what interace card are you using |
| 13:30.57 | many | robin: Check PMP/PTP before testing on, usually 3 BRIs on one MSN means they are PTP, but cards usually are PMP |
| 13:31.02 | many | (or vice versa?) |
| 13:31.03 | robin_sz | Eicon Networks Corporation Diva Server 4BRI-8M Rev 2 (rev 01) |
| 13:31.31 | clive- | and what trouble do you have? |
| 13:31.38 | robin_sz | unable to dial out ... |
| 13:31.53 | clive- | which version of chan_capi do you sue |
| 13:31.55 | clive- | use |
| 13:32.03 | robin_sz | how do I tell? |
| 13:32.24 | clive- | look at the filename |
| 13:32.33 | robin_sz | of? |
| 13:32.39 | robin_sz | capi.conf |
| 13:33.27 | puzzled | robin_sz: for starters use chan_capi-cm-0.6.1 from sourceforge.net |
| 13:33.31 | clive- | it looks like can_capi-rm.....something, . If I rememebr correctly |
| 13:33.39 | robin_sz | 'k |
| 13:33.47 | clive- | follow puzzled instructions |
| 13:33.50 | *** join/#asterisk Koenvi (n=kova@labs.ascom.be) |
| 13:33.58 | Koenvi | hello all! |
| 13:34.09 | *** join/#asterisk wunderkin (i=kev@12-201-105-27.client.mchsi.com) |
| 13:34.10 | puzzled | robin_sz: I installed it yesterday on asterisk-1.2.0 with a single BRI Eicon Diva Server and it works like a charm. |
| 13:34.19 | puzzled | Koenvi: morning |
| 13:34.20 | nfi|ermes | anyone can help me ? |
| 13:34.22 | robin_sz | puzzled, right ... |
| 13:34.31 | puzzled | nfi|ermes: just ask |
| 13:34.35 | Koenvi | anybody can help me out with installing a TE110P on astrisk1.2? |
| 13:34.45 | robin_sz | clive-, chan_capi maybe? |
| 13:34.58 | nfi|ermes | [14:29] <nfi|ermes> http://pastebin.ca/30290 |
| 13:35.09 | nfi|ermes | [14:29] <nfi|ermes> i ve some problems with dialparties.agi |
| 13:35.13 | robin_sz | chan_capi-cm-0.6 |
| 13:35.17 | nfi|ermes | in the cvs version |
| 13:35.25 | Koenvi | I'm actually looking for a configured example of zaptel.conf ... preferabbly for Belgium |
| 13:35.29 | robin_sz | puzzled, I have chan_capi-cm-0.6 |
| 13:35.59 | puzzled | robin_sz: get the new one |
| 13:36.02 | puzzled | 0.6.1 |
| 13:36.05 | robin_sz | right |
| 13:36.06 | fugitivo | nfi|ermes: well, there you have the problem |
| 13:36.31 | fugitivo | nfi|ermes: i don't know how that script works, but it's easy to debug |
| 13:36.36 | robin_sz | do I need to set isdnmode=ptp or something else? |
| 13:37.15 | puzzled | robin_sz: which country are you in? |
| 13:37.35 | Ansonmus | Is there anyone which have experience with Pika Technologies (www.pikatech.com) ? |
| 13:37.58 | fugitivo | nfi|ermes: if you are using nonnumeric values for comparison, you should use "eq", not "==" |
| 13:38.06 | robin_sz | puzzled, .ch |
| 13:38.14 | robin_sz | Swisscom |
| 13:38.24 | puzzled | robin_sz: isdnmode=ptp does not exist in 0.6.1. It's either msn or did. I have msn. try that |
| 13:38.26 | robin_sz | or swisspost.. thats it |
| 13:38.34 | robin_sz | right |
| 13:39.13 | robin_sz | oh, foo. the new SF.net is odd |
| 13:40.10 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 13:40.59 | nfi|ermes | thx fugitivo |
| 13:41.09 | nfi|ermes | but the problem is the second one |
| 13:41.39 | nfi|ermes | Unable to get channels at /var/lib/asterisk/agi-bin/dialparties.agi line 236, <STDIN> line 47. |
| 13:41.48 | brettnem | hey all |
| 13:42.22 | brettnem | does anyone know the correct syntax for the dial application where you specify username and password inline? |
| 13:42.23 | fugitivo | nfi|ermes: i don't have the script, but maybe if you fix the first problem, it'll work |
| 13:42.32 | nfi|ermes | i fixed |
| 13:42.41 | brettnem | the one specified in extensions.conf doesn't seem to work.. |
| 13:43.01 | nfi|ermes | row nr 236 of the script is: ($schannels)=$tn->waitfor('/.*active SIP channel/') or die "Unable to get channels", $tn->lastline; # wait for asterisk to process |
| 13:43.24 | astcryz | What can be the problem when i am registering to an asterisk, and trying to call, the called person can hear me but i cannot hear him/her |
| 13:43.27 | astcryz | what can be the problem? |
| 13:43.50 | brettnem | astcryz: That is typically a nat problem. |
| 13:44.08 | astcryz | is there a way to debug or find out what the problem can be? |
| 13:44.17 | astcryz | i am using asterisk realtime |
| 13:44.21 | brettnem | is the sip phone behind NAT? |
| 13:44.24 | astcryz | yes |
| 13:44.26 | brettnem | realtime has nothing ot do with it.. |
| 13:44.28 | astcryz | ok |
| 13:44.29 | brettnem | it's a nat problem then.. |
| 13:44.35 | brettnem | do you have nat=yes set in the sip user? |
| 13:44.45 | astcryz | yes, else the sip user would not be able to register |
| 13:44.47 | astcryz | :-) |
| 13:44.57 | brettnem | what about qualify? |
| 13:45.02 | astcryz | not set |
| 13:45.05 | brettnem | set it.. |
| 13:45.15 | brettnem | what is the nat device? and what is the sip endpoint? |
| 13:45.16 | astcryz | to? 1000? |
| 13:45.24 | astcryz | xten eyebeam |
| 13:45.25 | brettnem | anyhing really.. it doesn't matter.. |
| 13:45.45 | fugitivo | nfi|ermes: sorry, can't help you only looking at one line |
| 13:45.51 | nfi|ermes | ok |
| 13:45.56 | nfi|ermes | thx anyway |
| 13:46.30 | brettnem | the qualify value is a value the qualify must be within.. doesn't change the frequency (directly).. setting qualify to yes or a positive integer will cause asterisk to OPTIONS the device every 60 seconds for working devices or 10 seconds for non-responsive devices.. this OPTIONS will keep NAT translations from going stale in your router. |
| 13:46.55 | wunderkin | brettnem: something like this Dial(IAX2/uid:pw@iaxtel.com/${EXTEN}@iaxtel) |
| 13:47.28 | astcryz | now after i set qualify, it says username/auth name mismatch |
| 13:47.33 | brettnem | wunderkin: I tried that.. but it keeps giving me errors like "no such host: 7135551212@iaxtel.com |
| 13:48.12 | DYOGI_B | anyone know a good live stun cd |
| 13:48.21 | astcryz | ah my fault |
| 13:48.23 | brettnem | wunderkin: btw, I'm using SIP |
| 13:48.27 | astcryz | brettnem: qualify did not work |
| 13:48.28 | fugitivo | brettnem: for iax or sip? |
| 13:48.53 | brettnem | astcryz: Do a SIP debug and see if the c= line shows public or private addresses |
| 13:48.58 | brettnem | fugitivo: SIP |
| 13:49.15 | *** join/#asterisk niZon (n=ilt@S0106deadbeefbeef.wp.shawcable.net) |
| 13:49.20 | brettnem | extensions.conf clearly shows: ;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT) |
| 13:49.24 | brettnem | but that does NOT work |
| 13:49.26 | *** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg) |
| 13:49.27 | fugitivo | brettnem: add the user/pass in your sip.conf and use SIP/${EXTEN}@ip to dial |
| 13:49.37 | wunderkin | brettnem: sip for iaxtel? |
| 13:50.04 | brettnem | fugitivo: The problem I am having is that I send calls to one server.. So I have one terminating IP, but I need to use seperate credentials for certain calls.. get it? |
| 13:50.28 | fugitivo | brettnem: ok |
| 13:50.29 | wunderkin | oh |
| 13:50.35 | *** join/#asterisk jimmy_deanPB (n=jhodapp@72.244.232.226) |
| 13:50.38 | brettnem | which means.. I have to create seperate PEER entries for each dialout.. which I don't know if that works to begin with.. but where the real problem is in RECIEVING calls frmo that IP.. |
| 13:50.47 | yxa | anyone has polycom sip image and bootrom? |
| 13:50.51 | brettnem | so any ideas |
| 13:51.01 | brettnem | yxa: check the wiki.. there are plenty of links for that. |
| 13:51.03 | brettnem | ~wiki |
| 13:51.15 | brettnem | hm |
| 13:51.36 | fugitivo | brettnem: didn't try sip dial with user and pass, from the dial line, sorry |
| 13:51.38 | wunderkin | yeah i always specify the uid/pw in the conf too |
| 13:52.02 | brettnem | yeah.. just need to specifiy a bunch of usernames for the same terminating IP |
| 13:52.03 | yxa | brettnem thought they are only for certified resellers or something? |
| 13:52.04 | *** join/#asterisk RoyK (n=roy@213.160.242.69) |
| 13:52.31 | brettnem | yxa: go to wiki.. search for polycom.. find link for firmware.. be happy... |
| 13:52.52 | brettnem | yxa: there is a good one at freedomphones.net or something.. has the manuals and all |
| 13:53.21 | brettnem | oh.. |
| 13:53.24 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:53.31 | brettnem | yxa: actually I made a shortcut to it some time ago.. http://doiop.com/polycomfirmware |
| 13:53.40 | wunderkin | im not sure if you can have multiple peers for the same ip or not |
| 13:53.52 | brettnem | wunderkin: right.. desperatly need that |
| 13:54.02 | brettnem | seems like you should.. |
| 13:54.14 | brettnem | the whole peer/user thing seems totally f-d.. really.. |
| 13:54.21 | yxa | brettnem dude you da man |
| 13:54.58 | brettnem | when you do a dial(sip/peername)... it looks for peername as a peer.. when a call comes in from a certain IP, it looks for that IP in entries marked as PEERS.. WTF is a user for? it's totally messed up |
| 13:55.04 | brettnem | yxa: np |
| 13:55.22 | *** join/#asterisk ChrisDE_ (n=ChrisDE+@tmo-069-249.customers.d1-online.com) |
| 13:55.26 | wunderkin | i dont understand the workings of that all personally yet |
| 13:55.32 | ChrisDE_ | hi |
| 13:55.42 | brettnem | yxa: dont' upgrade to BR 3 unless you absolutely know what you are doing. you have been warned |
| 13:55.54 | ChrisDE_ | is there a chance of getting the ip address of the caller? |
| 13:56.02 | brettnem | wunderkin: it's one of "those" "features" of asterisk |
| 13:56.11 | wunderkin | brett can you sell me a toll free did and get it listed it toll free directory assistance? |
| 13:56.28 | fugitivo | brettnem: user is for register information, if you're not registered, you can't receive calls |
| 13:56.29 | brettnem | ChrisDE_: Check "show functions" and see if you see anything promising in there.. |
| 13:56.45 | brettnem | wunderkin: I might be able to.. don't do a whole lot of TF |
| 13:56.51 | *** join/#asterisk tracinet (n=tracinet@24-50-29-205.atlsfl.adelphia.net) |
| 13:56.56 | wunderkin | ah |
| 13:57.06 | brettnem | fugitivo: thanks for that.. still seems messed up |
| 13:57.23 | brettnem | wunderkin: I can give you a texas number and get it listed. :) |
| 13:57.25 | [TK]D-Fender | I run BR 2.6.1 and SIP 1.5 on my Polycom IP 600's and see no reason to upgrade. Works well. |
| 13:57.48 | wunderkin | brettnem: heh ya.. well that wouldnt work, it would have to be local or tf |
| 13:57.54 | ChrisDE_ | I mean something like a channel variable showing the ip address? |
| 13:57.59 | brettnem | I run BR 2.6.1 and SIP 1.6.. wanted the lastest to support subscriptions.. |
| 13:58.22 | brettnem | ChrisDE_: Do a show channel during an active call to list all registered channel variables. |
| 13:58.26 | brettnem | you might see itin there. |
| 13:58.26 | tracinet | i see asterisk and zaptel 1.0.10 is out on asterisk.org - anyone know if libpri will stay at 1.0.9 or be upgraded to 1.0.10 as well ? |
| 13:58.46 | brettnem | [TK]D-Fender: do SIP SUBSCRIPTIONS work with SIP 1.5? |
| 13:58.55 | fugitivo | tracinet: 1.2 is the last version |
| 13:58.56 | yxa | brettnem what happens if i use BR3? |
| 13:59.17 | wunderkin | brettnem: the listing part of it is pretty important right now, i need it done asap :/ i read the at&t manual and they say normal submissions are done within 5 biz days and thats about all i have so i need to make sure its all done right ;/ |
| 13:59.20 | brettnem | yxa: you won't ever be able to downgrade and it only offers some limited extra functionality.. like SSL for the configuration website.. |
| 13:59.25 | tracinet | 1.2 is latest version in the 1.2 tree - but 1.0.10 is newest version in the 1.0.x tree |
| 13:59.45 | brettnem | wunderkin: I can understand that.. I'm not a resporg, so I'd have to check with my resporg. |
| 13:59.47 | tracinet | it contains all fixes since 1.0.9 for those usng the 1.0.x tree |
| 13:59.57 | wunderkin | brettnem: ah yeah that was the other question.. alright |
| 14:00.00 | brettnem | brb kids |
| 14:00.11 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 14:00.36 | ChrisDE_ | there is no ip-address displayed when running show channel xxxxx |
| 14:01.03 | yxa | brettnem so what should i use? 1.6.2 with 2.6.1 fine? |
| 14:01.38 | fugitivo | ChrisDE_: show channels |
| 14:01.41 | [TK]D-Fender | brettnem : not sure... I'm not that well versed in SIP implementation. |
| 14:02.23 | ChrisDE_ | show channels also doesn't show any ip adress, but sip show peer xxxx does. But I need the ip-address as a variable |
| 14:02.33 | ChrisDE_ | or it should be logged in the cdr |
| 14:03.04 | brettnem | yxa: 1.6.2 is probably good. |
| 14:03.31 | brettnem | I got Buddy watching and peerstate working real nice in asterisk 1.2 with polycom sip 1.6 |
| 14:04.10 | fugitivo | what is buddy watching? |
| 14:04.23 | yxa | brettnem which phone do you have? i'm trying out the ip301 |
| 14:04.24 | brettnem | still trying to figure out how to indicate a ringing state on those polycom phones.. I can't figure out what status to send to the phone... requires a patch to chan_sip just need the damn text for the status name |
| 14:04.29 | brettnem | IP500 |
| 14:04.45 | wunderkin | brettnem: do you think you can find something out today? ive been battling with a local voip provider over the past week and havent accomplished anything so im trying to find someone asap since im running out of time |
| 14:04.51 | brettnem | fugitivo: It allows a softkey to display the status of another extensions (idle, inuse, ringing, busy, etc) |
| 14:04.56 | ChrisDE_ | so is the best way of getting an ip-address to parse the sip username out of the channel (e.g. SIP/test-jdfjf) and then do a sip show peer test and then parse the ip-address out from the output? |
| 14:05.10 | *** join/#asterisk Lurr (n=pr0ph3t@63.69.20.3) |
| 14:05.21 | brettnem | wunderkin: email to our sales guy bdehaven at worldcall dot net |
| 14:05.28 | wunderkin | ok |
| 14:05.29 | *** part/#asterisk Lurr (n=pr0ph3t@63.69.20.3) |
| 14:05.41 | fugitivo | brettnem: that works using hint ? |
| 14:05.44 | brettnem | ChrisDE_: That sounds super kludgy |
| 14:06.03 | brettnem | fugitivo: yes, hints.. I'd like to get app_devstate to work, but I can't get it to complile against asterisk 1.2 |
| 14:06.18 | ChrisDE_ | right... sounds bad... so what would be better? |
| 14:06.26 | [TK]D-Fender | Hint works find with my setup right now... |
| 14:06.41 | ChrisDE_ | thats the only possibility I can think of |
| 14:07.08 | fugitivo | [TK]D-Fender: what's your setup? |
| 14:07.53 | wunderkin | brettnem: is there a phone number so can speak to someone? |
| 14:09.12 | [TK]D-Fender | fugitivo : 26 Polycom IP 600's, BR 2.6.1 SIP 1.5, D-Link 1526 PoE Switches, some recent version of CVS HEAD.... |
| 14:09.34 | [TK]D-Fender | PIR & Rhino CB on Sangoma A104d |
| 14:10.10 | fugitivo | [TK]D-Fender: is the dlink nice? |
| 14:10.13 | *** part/#asterisk ChrisDE_ (n=ChrisDE+@tmo-069-249.customers.d1-online.com) |
| 14:10.35 | [TK]D-Fender | fugitivo : definately no complaints. Its the best bang / buck I've ever seen for its kind... |
| 14:10.56 | fugitivo | i'll keep it in mind, how much did you pay for it? |
| 14:11.08 | [TK]D-Fender | 1050$CDN |
| 14:11.18 | fugitivo | CDN? |
| 14:11.25 | [TK]D-Fender | for 24 port w/ 2GB uplink ports |
| 14:11.27 | [TK]D-Fender | yup |
| 14:11.48 | *** join/#asterisk edmondD (n=Jade@dsl-202-173-190-245.qld.westnet.com.au) |
| 14:11.56 | yxa | brettnem cant i set my sip info from the phone directly? i'm reading the docs as we speak too... |
| 14:11.58 | edmondD | any asterisk experts on the Gold Coast, Austrlaia |
| 14:12.04 | edmondD | will pay |
| 14:12.16 | [TK]D-Fender | Dang.. they're $890CAD now! |
| 14:12.18 | brettnem | yxa: in the new firmware you can.. not sure abotu the older firmware.. |
| 14:12.28 | brettnem | it's not as bad as it looks to use the polycom XML files |
| 14:12.31 | *** join/#asterisk __a (n=a@85.100.123.10) |
| 14:13.20 | brettnem | so any ideas on how to dial to the same sip peers with different credentials? |
| 14:13.22 | brettnem | anyone? |
| 14:13.31 | brettnem | [TK]D-Fender: I'd love to check that out.. I bet it's cool |
| 14:13.44 | [TK]D-Fender | I made some pages that use the AMI to do company-wide SIP HINT polling, etc... |
| 14:14.02 | brettnem | nice |
| 14:14.10 | [TK]D-Fender | And my CSR's have live queue stats on their idle screen along withour logo :) |
| 14:14.21 | brettnem | nice |
| 14:14.28 | brettnem | expensive phone for call center. |
| 14:14.42 | __a | any idea how to enable/disable zap channel programmaticaly? |
| 14:14.47 | MacRohard | brettnem: register them with different names? |
| 14:14.55 | [TK]D-Fender | What do you mean call center? I got them for EVERYONE ;) We ar a 35 person office +/- and 4 CSR's :) |
| 14:15.13 | brettnem | MacRohard: right, but what about incoming calls for that peer? |
| 14:15.18 | *** part/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 14:15.30 | brettnem | MacRohard: it's a sip proxy.. so I recieve calls w/o authentication.. |
| 14:15.39 | yxa | brettnem i'm completely lost here. after i unzip i get several files phone1.cfg sip.cfg 0000000000 etc |
| 14:15.45 | brettnem | MacRohard: but I send calls out with custom auth.. which specifies calling scope. |
| 14:15.52 | [TK]D-Fender | Basically when I looked at what it would ahve cost to get PoE on an IP 500 and factored in the microbrowser and more lines (buddy-watch etc usage) I said "why the hell not?" |
| 14:15.58 | brettnem | yxa: those are the default configuration files |
| 14:16.03 | fugitivo | how to check hint from the manager? |
| 14:16.11 | MacRohard | brettnem, don't you do dial number@peer? |
| 14:16.33 | brettnem | MacRohard: right, but I need to set authentication based on the tenant that is making the call |
| 14:16.33 | *** join/#asterisk Katty (n=angela@68-112-15-110.dhcp.cpgr.mo.charter.com) |
| 14:16.35 | [TK]D-Fender | fugitivo : I "cheated" and did a CLI command "SIP SHOW HINTS" and parsed it the hard way :) |
| 14:16.54 | brettnem | heh |
| 14:16.58 | brettnem | cheater... I'm tellin |
| 14:17.00 | MacRohard | brettnem: oh. well i gues you'd need to register all the different tenants and then do dial number@tenant |
| 14:17.04 | [TK]D-Fender | I am the King of "brute force" |
| 14:17.07 | fugitivo | [TK]D-Fender: i was looking at that command :) |
| 14:17.25 | brettnem | MacRohard: No, this is for my tenants, who are registered to asterisk to make calls out to a sip proxy |
| 14:17.42 | MacRohard | yea.. so each tenant would need a different login on the proxy |
| 14:17.44 | brettnem | each tenant (in asterisk) sends calls to a sip proxy (the same proxy) with their own authentication |
| 14:17.45 | __a | guys, is it possible to enable/disable zap channel via a script or something? I know zap destroy channel would disable it, but there's no corresponding zap create channel :( |
| 14:18.05 | MacRohard | and asterisk would register multiple times, once for each tenant |
| 14:18.12 | brettnem | MacRohard: Right, and that works.. but what abotu calls coming from the SIP proxy? |
| 14:18.34 | brettnem | MacRohard: oh.. asterisk register to the proxy.. interesting.. let me contemplate that... |
| 14:18.35 | MacRohard | brettnem, well i guess that depends where it sends them |
| 14:18.52 | fugitivo | i don't get something about hints |
| 14:19.07 | robin_sz | puzzled, OK, I have 0.6.1 installed now .. same problem ... |
| 14:19.23 | fugitivo | i can use for example hint,SIP/xxx&SIP/xxx&SIP/xxxx |
| 14:19.34 | yxa | brettnem phone1.cfg and sip.conf completely lost me. can you tell me what they do? |
| 14:19.35 | brettnem | MacRohard: hmm.. not sure how a registration would jive with my routing module in SER. |
| 14:19.41 | [TK]D-Fender | fugitivo : you do those 1-line at a time |
| 14:19.42 | fugitivo | but show hints, will tell only the state of the first dial command, right? |
| 14:19.57 | brettnem | yxa: phone1.cfg is custom config for THAT PHONE.. sip.cfg is the sitewide configuration |
| 14:19.59 | MacRohard | brettnem, well it's no different than having multiple people registering with the sip proxy from different hosts |
| 14:20.33 | fugitivo | [TK]D-Fender: if i setup one extension to call 3 SIP devices |
| 14:20.42 | fugitivo | [TK]D-Fender: i should use hint for the 3 sip devices |
| 14:20.43 | [TK]D-Fender | fugitivo : You put the HINT in the same context as your phone uses and point them to other phones. Then tell the phone to watch a specific #. |
| 14:20.56 | [TK]D-Fender | fugitivo : not sure how that'll work.... |
| 14:21.07 | [TK]D-Fender | dnd't know you could do multiple. |
| 14:21.11 | [TK]D-Fender | will look into. |
| 14:21.13 | brettnem | MacRohard: Right.. well the sip proxy right now is really designed to route calls from server to server.. not really user to user.. so i use LCR to autofailover to backup servers.. not sure how that will work with registrations if I can't set the q-factor |
| 14:21.30 | brettnem | [TK]D-Fender: or you set subscribecontext in sip.conf |
| 14:21.48 | MacRohard | brettnem, presumably when it fails asterisk will attempt to re-login |
| 14:22.15 | brettnem | [TK]D-Fender: btw, a HINT extension MUST have some action in it.. you can't set an arbitratry extension with a HINT without any priority 1.. I actually set my hint extension to a PICKUP application with works super cool |
| 14:22.21 | fugitivo | [TK]D-Fender: i'm doing right now, a call from extension 1 that has 3 sip devices to another extension |
| 14:22.33 | brettnem | MacRohard: No, I mean my proxy will attempt to send it to a backup asterisk |
| 14:22.41 | fugitivo | [TK]D-Fender: and show hints, shows extension 1 is in use, but not all devices are in use |
| 14:23.05 | brettnem | MacRohard: with the registrations, both asterisk servers would be registered and the SIP proxy will ring both lines at the same time. |
| 14:23.16 | fugitivo | [TK]D-Fender: do the test |
| 14:23.30 | [TK]D-Fender | I would but my GUI makes that painful..... |
| 14:23.33 | MacRohard | brettnem, well then i guess you need to figure out a way of making your tenants also fail over to registering with your backup asterisk server |
| 14:23.38 | fugitivo | [TK]D-Fender: lol |
| 14:23.44 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 14:23.49 | [TK]D-Fender | and no personal need, though I might test it at home... |
| 14:23.55 | robin_sz | OK, so what is the difference between protocol layer 1 errors and layer 2 errors??? |
| 14:24.07 | brettnem | MacRohard: right, I already have that going with CARP.. hmm.. I suppose that would work for SER as well.. interesting.. |
| 14:24.07 | MacRohard | brettnem, possibly they could also register with the sip proxy i guess |
| 14:24.08 | fourcheeze | if I use asterisk realtime for sip users am I stuck with a particular schema? |
| 14:24.19 | fugitivo | fourcheeze: yes, using a database |
| 14:24.40 | fourcheeze | what I want to do is hold several servers full of information in one database |
| 14:24.43 | brettnem | MacRohard: well now that I have my proxy coming together, I might just have my tenants register to SER and totally bypass asterisk unless there is a good reason to use it. |
| 14:24.45 | __a | guys, is it possible to enable/disable zap channel via a script or something? I know zap destroy channel would disable it, but there's no corresponding zap create channel :( |
| 14:24.55 | fugitivo | fourcheeze: that'd be a mess if the database is down |
| 14:25.07 | brettnem | __a: why would you do that? |
| 14:25.16 | MacRohard | brettnem, yea.. i guess hte only reason would be if you want to have some sort of an internal dialplan |
| 14:25.21 | fourcheeze | fugitivo:how would it be better if I had each one on its own database? |
| 14:25.30 | brettnem | MacRohard: My routing module supports that |
| 14:25.43 | fugitivo | fourcheeze: i don't like databases for that :/ |
| 14:26.08 | fourcheeze | fugitivo: ok, there's obviously some good way to scale up asterisk |
| 14:26.15 | fourcheeze | so what is it :-) |
| 14:26.19 | brettnem | fourcheeze: HAHA yeah right! |
| 14:26.23 | fugitivo | text files |
| 14:26.33 | brettnem | "dont' use asterisk" |
| 14:26.35 | brettnem | ;) |
| 14:26.38 | fugitivo | lol |
| 14:26.46 | fourcheeze | brettnem: and use what instead? |
| 14:26.49 | brettnem | asterisk != scalable |
| 14:26.53 | fugitivo | fourcheeze: cisco call manager! |
| 14:27.02 | brettnem | fourcheese: openpbx someday.. |
| 14:27.12 | fugitivo | yatE? |
| 14:27.18 | fourcheeze | ok, let's rephrase the question |
| 14:27.19 | brettnem | <shrug> |
| 14:27.29 | brettnem | I think yate would probably scale better.. but harder to manage.. |
| 14:27.35 | brettnem | SER+asterisk is a good mix... |
| 14:27.37 | fourcheeze | in the next year I'm hoping to go from 0 to several thousand sip users |
| 14:27.40 | fugitivo | i have to try it |
| 14:27.41 | brettnem | but it's quite a learning curve. |
| 14:27.48 | fourcheeze | what things should I do now that will help me in 1 year |
| 14:27.57 | brettnem | fourcheeze: learn SER |
| 14:28.02 | fourcheeze | I played with SER |
| 14:28.02 | robin_sz | OK, so ... it looks like it is connecting to the line ... when I unplug the line I get a layer 1 error, when I plug the line in I get a layer 2 eror .. so its seeign "something" of the world at least |
| 14:28.02 | brettnem | really |
| 14:28.07 | fourcheeze | don't like SER |
| 14:28.12 | brettnem | fourcheeze: tough |
| 14:28.15 | fourcheeze | it doesn't have enough features |
| 14:28.21 | __a | brettnem: i need to. can zap be used with realtime? |
| 14:28.35 | brettnem | __a: I don't "do" realtime.. sorry |
| 14:28.44 | brettnem | fourcheeze: right, which is why it scales |
| 14:28.52 | fugitivo | i hate realtime, why did they put that feature in asterisk? |
| 14:28.56 | file[laptop] | why, why oh why, would you do zap realtime? |
| 14:29.08 | fourcheeze | brettnem: ok, well it doesn't do what I need it to do, so I have to rule it out |
| 14:29.08 | brettnem | fourcheeze: so combine asteisk and ser so when people actually need features, it gets shuttled over to asterisk |
| 14:29.18 | brettnem | fourcheeze: it probably does, actually |
| 14:29.20 | __a | file[laptop]: for enabling/disabling zap channel by changing a context e.g. |
| 14:29.33 | __a | file[laptop]: you got any better idea? |
| 14:29.46 | brettnem | __a: perhaps if you explain wha tyou are actually trying to do, we can make some suggestions |
| 14:29.49 | fourcheeze | brettnem: do you know of a tutorial for SER asterisk integration? |
| 14:30.05 | fourcheeze | brettnem: where do you keep your users in such a configuration - flat files? |
| 14:30.10 | brettnem | fourcheeze: no, the ser community is incredibly tight-lipped. don't really know why |
| 14:30.20 | brettnem | fourcheeze: yes, I use flat files.. lots of includes |
| 14:30.30 | fugitivo | it's time to write some documentation |
| 14:30.31 | __a | brettnem: i'm building a callshop app and the customer would like to enable/disable a booth via the interface. |
| 14:30.35 | brettnem | fourcheeze: BTW, I'd recommend openser instead of SER |
| 14:30.36 | fourcheeze | brettnem: so you have the same users in SER and asterisk? |
| 14:30.57 | brettnem | fourcheeze: sorta.. |
| 14:31.09 | brettnem | __a: why not just make them interfaces in a call queue? just login and logout.. |
| 14:31.10 | robin_sz | bah ... CAPI INFO 0x3302: Protocol error layer 2 |
| 14:31.32 | fourcheeze | brettnem: a lot of the stuff I'm doing involves complex group rings etc etc and it needs to be configurable via a "normal person" with a web browser |
| 14:31.46 | __a | brettnem: could you elaborate please? |
| 14:31.52 | robin_sz | so ... having failed miserably to get this eicon diva to work .. where should I turn next?? or just dig a hole and bury it?? |
| 14:32.01 | brettnem | fourcheeze: well either develop in ser or asterisk.. it's just a database.. and sip signalling. |
| 14:32.04 | fourcheeze | hence why I immediately thought of using mysql or something |
| 14:32.04 | Hmmhesays | eicon diva? |
| 14:32.09 | Hmmhesays | send it to me |
| 14:32.18 | robin_sz | 4 x BRI |
| 14:32.19 | brettnem | __a: do you know how call queuing in asterisk works? |
| 14:32.30 | __a | yes |
| 14:32.40 | robin_sz | Hmmhesays, you understand these cards? |
| 14:32.50 | Hmmhesays | robin_sz, no never touched one |
| 14:33.00 | clive- | robin, sell it to me:) |
| 14:33.15 | brettnem | __a: ok, so make your zapata interfaces (they are subscriber lines, right?) log in and log out dynamically... see "show application addqueuemember" I think that's the app |
| 14:33.23 | robin_sz | clive-, I was mor thinkiong of getting it working before the boss kills me :) |
| 14:33.39 | robin_sz | 25 SNOM 360s, a nice new server and a poxy card. |
| 14:33.51 | clive- | robin, you need to have the drivers from melaware, not the stock kernel 2.6 drivers |
| 14:33.59 | robin_sz | I do ... I do |
| 14:34.07 | brettnem | those ISDN problems sound like they are below asterisk and have nothing to do with it... |
| 14:34.17 | clive- | hmmm, dunno..... |
| 14:34.22 | clive- | time to go home |
| 14:34.26 | clive- | good luck |
| 14:34.31 | robin_sz | thnaks :) |
| 14:35.17 | brettnem | bbiab.. |
| 14:35.58 | fourcheeze | yate seems to support having external python modules |
| 14:36.20 | fourcheeze | sounds promising already |
| 14:37.04 | fourcheeze | ok, plan C |
| 14:37.19 | fourcheeze | I could have each asterisk server have its own mysql with the realtime schema |
| 14:37.36 | fourcheeze | and each one can suck over its own config from the more complicated central schema |
| 14:37.54 | fourcheeze | then if the central DB goes down each one can keep going in a read-only mode |
| 14:37.55 | brettnem | look up #exec |
| 14:38.08 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:39.10 | fourcheeze | brettnem: hmm? |
| 14:39.16 | *** join/#asterisk jmacz (n=jmacz@208.195.215.48) |
| 14:40.14 | fugitivo | brettnem: why you want to use a database? |
| 14:40.34 | fourcheeze | fugitivo: do you mean me? |
| 14:40.46 | fugitivo | err, yes |
| 14:40.49 | fourcheeze | ok |
| 14:40.58 | fourcheeze | fugitivo: for ease of setting options from web pages |
| 14:41.22 | fourcheeze | I don't want editing a sip user to be really editing sip.conf |
| 14:42.10 | fourcheeze | also I want to keep data about all our servers in the same place so I can easily route xalls beween them |
| 14:42.18 | fourcheeze | *calls |
| 14:42.34 | yxa | brettnem i got a Registration failed username/auth name mismatch |
| 14:42.48 | fourcheeze | fugitivo: is that bad? |
| 14:43.20 | fugitivo | personally, i don't like databases for a critical systems like communications, but it's my preference |
| 14:43.46 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
| 14:44.03 | fourcheeze | well, I'm not ruling out writing out sip.conf, extensions.conf from DB |
| 14:44.24 | fourcheeze | that's really the kind of opinions I want |
| 14:44.33 | fourcheeze | how are people doing this kind of centralised admin |
| 14:47.12 | *** part/#asterisk __a (n=a@85.100.123.10) |
| 14:47.18 | *** part/#asterisk Defraz (n=t0tal@24-119-12-238.cpe.cableone.net) |
| 14:48.48 | fugitivo | i don't have a lot of asterisk servers, but some time ago i worked with a lot of apache servers (a LOT), and we used to copy the config files with scp |
| 14:49.05 | fugitivo | using a script |
| 14:49.21 | fourcheeze | yeah but in this case each server wants to be different# |
| 14:49.36 | fugitivo | then you don't have to copy anything |
| 14:49.37 | fourcheeze | it's not so much a load balancing situation |
| 14:49.50 | fourcheeze | I want a central information store |
| 14:50.21 | fourcheeze | do you think it's feasible to have N asterisk servers all interacting in some way and have all the configuration done by ssh/emacs ? |
| 14:50.22 | fugitivo | you can use a database, and then generate the text files |
| 14:50.32 | fourcheeze | fugitivo: yes, that's an option |
| 14:50.42 | fugitivo | i know people that is doing that |
| 14:51.22 | *** join/#asterisk __a (n=a@85.100.123.10) |
| 14:51.29 | fourcheeze | I don't really know why, since we can now have replication in mysql, people aren't using realtime more, or why realtime doesn't support my choice of schema |
| 14:52.22 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-19-69.44-151.net24.it) |
| 14:52.28 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 14:53.04 | fugitivo | maybe people doesn't want to have another point of failure on their servers |
| 14:53.19 | fugitivo | that's my reason of not using realtime |
| 14:53.30 | fourcheeze | is it possible / feasible / stupid / desirable to hand over routing to a single AGI script which then does everything that needs to be done, thereby replcing extensions.conf ? |
| 14:53.44 | fourcheeze | fugitivo: even if you can have 2 database servers replicating? |
| 14:53.51 | fourcheeze | that's not a single point |
| 14:53.59 | fugitivo | another point of failure :) |
| 14:55.14 | fugitivo | kiss is better |
| 14:55.24 | *** join/#asterisk burtonez (i=mimx@w201.ljudmila.org) |
| 14:55.40 | Hmmhesays | fourcheese why would you do that? |
| 14:55.59 | Hmmhesays | you want to fire up a new interpreter for every single call? |
| 14:56.04 | fourcheeze | because it seems to present a scalable solution |
| 14:56.37 | fourcheeze | call can come in, find out which server it's on, find out where it needs to go, do whatever needs doing, finish |
| 14:57.08 | fourcheeze | it seems to make more sense to do it once |
| 14:57.14 | fugitivo | fourcheeze: i think the best solution in that case, would be using a unique database, and then generating the config files for each server |
| 14:57.15 | fourcheeze | than to be running several AGI scripts |
| 14:57.27 | Hmmhesays | register your endpoints to SER |
| 14:57.39 | __a | guys, is there any way of disabling/enabling dialtone on zap channels dynamically? |
| 14:57.40 | fourcheeze | Hmmhesays: do you have a tutorial? |
| 14:57.47 | yxa | someone help in polycom config? i'm getting Registration from '<sip:polycom@192.168.2.10>' failed for '192.168.2.15' - Username/auth name mismatch. anyone knows where it gets the "sip" part from? |
| 14:58.10 | Hmmhesays | use asterisk for the media, voicemail, routing, whatever |
| 14:58.19 | __a | yxa: polycom is sending it |
| 14:58.23 | Hmmhesays | fourcheeze: no but the SER manual is very in depth |
| 14:58.41 | yxa | __a that's not what i want. where do i change that? |
| 14:58.43 | fourcheeze | Hmmhesays: so my users exist in SER? |
| 14:58.47 | fourcheeze | not asterisk |
| 14:58.50 | __a | in polycom obviously |
| 14:58.57 | yxa | __a i'm not playing with config files yet |
| 14:59.05 | *** join/#asterisk VoIPA (n=VoIPA@mail.kancharla.com) |
| 14:59.07 | Hmmhesays | yes, because SER can handle a ridiculously large amount of registrations |
| 14:59.29 | chapeaurouge | if i have 2 ISDN lines, that means i can only have 4 simultaneous calls going at once, correct? |
| 14:59.30 | fourcheeze | Hmmhesays: but then we lose things like contexts |
| 14:59.35 | Hmmhesays | why? |
| 14:59.45 | fourcheeze | ok, maybe not |
| 14:59.53 | fourcheeze | possibly my knowledge of SER is not great :-) |
| 15:00.01 | Hmmhesays | theres ways around it |
| 15:00.02 | fourcheeze | but when I last looked I didn't see contexts |
| 15:00.08 | Hmmhesays | you lose your sip registration contexts |
| 15:00.36 | Hmmhesays | but there are ways around it |
| 15:01.00 | fourcheeze | like :-) |
| 15:01.02 | Hmmhesays | imagine having an ata with 400 phones attached to it |
| 15:01.05 | *** part/#asterisk VoIPA (n=VoIPA@mail.kancharla.com) |
| 15:01.05 | [TK]D-Fender | I just got my shipment in from VoipSupply :D |
| 15:01.14 | fourcheeze | Hmmhesays: ok |
| 15:01.15 | *** join/#asterisk VoIPA (n=VoIPA@mail.kancharla.com) |
| 15:01.42 | Hmmhesays | that is the most simply way you can probably think of using SER with asterisk, slightly innaccurate but close enough |
| 15:01.59 | *** join/#asterisk izo (n=izo@193.202.114.43) |
| 15:02.26 | robin_sz | chapeaurouge, correct 2 ISDN lines is 4 D channes and 2 B channels .. I 4 voice, 2 controls |
| 15:02.34 | Hmmhesays | how many users are you looking to accomodate? |
| 15:02.36 | izo | anybody knows how to get rid of unnecesary channel briding on outgoing channels ? |
| 15:02.58 | VoIPA | D channels are signalling |
| 15:03.03 | robin_sz | or is it 4B+2D |
| 15:03.05 | VoIPA | 4 B + 2 D |
| 15:03.12 | *** join/#asterisk ramtha (n=ramtha@195.14.234.162) |
| 15:03.13 | robin_sz | yeah, anywya, 4 voice 2 data |
| 15:03.14 | yxa | __a its giving me errors but it works |
| 15:03.14 | ramtha | hi |
| 15:03.17 | chapeaurouge | ok |
| 15:03.25 | ramtha | trying to set up meetme makes me crazy |
| 15:03.32 | robin_sz | Oh, unless you are my neighbour ... |
| 15:03.34 | ramtha | i compiled ztdummy and load the modul |
| 15:03.40 | fourcheeze | Hmmhesays: as many as I need to :-) |
| 15:03.41 | ramtha | added 1234 to meetme |
| 15:03.50 | VoIPA | anyone know how to turn off hold on asterisk? |
| 15:03.54 | fourcheeze | Hmmhesays: for the sake of argument let's say 50,000 |
| 15:04.02 | robin_sz | he gets two phone calls and a analogue modem down his single channel ISDN ssytem at the same time ... |
| 15:04.03 | ramtha | but if i dial into the conferenze, asterisk sais |
| 15:04.08 | ramtha | conf-invalid |
| 15:04.08 | robin_sz | or so he tells me ... |
| 15:04.11 | Hmmhesays | providing what kind of services? |
| 15:04.11 | ramtha | why is that? |
| 15:04.36 | fourcheeze | Hmmhesays: virtual pbx stuff, call groups, voicemail, parking, |
| 15:04.38 | robin_sz | but hes a highly paid UNISYS consultant ... so, it must be true |
| 15:04.53 | fugitivo | ramtha: setup correctly your meetme.conf |
| 15:05.15 | fourcheeze | Hmmhesays: one important thing is that I have to be able to set callerid for each outgoing call |
| 15:05.22 | fourcheeze | but I guess that would go throuhg asterisk anyway |
| 15:05.23 | ramtha | fugitivo: conf => 9876 |
| 15:05.32 | ramtha | i thought that could be enough? |
| 15:05.44 | Hmmhesays | I would start reading about SER |
| 15:06.03 | fourcheeze | Hmmhesays: ok |
| 15:06.04 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 15:06.04 | fugitivo | ramtha: then you use xxx,x,Meetme(9876) in your extensions.conf ? |
| 15:06.10 | fourcheeze | Hmmhesays: do you keep your data in mysql with ser? |
| 15:06.16 | ramtha | extensions.conf links 9876 to Meetme,9876 |
| 15:06.19 | Hmmhesays | you can |
| 15:06.26 | ramtha | fugitivo: yes |
| 15:06.32 | robin_sz | hmmm how can I set an extension so when the user dials it .. it goes to their mailbox and asks for a password .. |
| 15:06.35 | fourcheeze | Hmmhesays: how do you do it? |
| 15:06.40 | Hmmhesays | read the manual |
| 15:06.41 | robin_sz | so they dont need to enter the mailbox number?? |
| 15:06.41 | fourcheeze | for your users |
| 15:07.05 | Hmmhesays | exten => _X.,1,VoicemailMain(${EXTEN}) |
| 15:07.11 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:07.12 | robin_sz | ooh, ta. |
| 15:07.17 | fourcheeze | Hmmhesays: I mean do you use mysql or flat files in your installation |
| 15:07.18 | Hmmhesays | if your voicmail boxes are the same as your extensions |
| 15:07.20 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 15:07.24 | Hmmhesays | me personally? |
| 15:07.27 | robin_sz | mmm ... mostly |
| 15:07.31 | robin_sz | but not always |
| 15:07.48 | Hmmhesays | robin_sz: you probably realize the syntax anyway if you wanted to change that |
| 15:08.06 | fourcheeze | Hmmhesays: yes persoonally :-) |
| 15:08.27 | *** join/#asterisk RussCC (n=face@216.157.205.211) |
| 15:08.40 | Hmmhesays | I don't use SER much, but if I do... whatever database they have running is fine |
| 15:10.59 | Hmmhesays | postgresql, mysql |
| 15:14.00 | burtonez | hi, anyone ever try to flash gs100 (grandstream) phone with jtag adapter? becouse my phone is dead :( |
| 15:16.25 | *** join/#asterisk grimse (n=grimse@p5481FA5E.dip.t-dialin.net) |
| 15:17.12 | *** join/#asterisk aio (n=aio@adsl-223-72-14.aep.bellsouth.net) |
| 15:17.49 | VoIPA | Everytime my softswitch sends a sip re-invite to asterisk, asterisk v1.2.0 puts the call on hold. |
| 15:17.56 | VoIPA | anyone know how to turn this off? |
| 15:18.26 | file[laptop] | look at the reinvite. |
| 15:18.33 | Hmmhesays | on hold as in you can take the call off hold? |
| 15:18.40 | VoIPA | I have, it looks fine |
| 15:18.52 | VoIPA | yes, by issuing a new re-invite to the original IP |
| 15:18.56 | file[laptop] | pastebin 'da contents |
| 15:19.15 | Hmmhesays | giddy'up |
| 15:19.19 | file[laptop] | :P |
| 15:19.28 | VoIPA | It will take a minute |
| 15:19.34 | Katty | i'll giddy your up in a minute |
| 15:19.45 | Katty | just as soon as i'm done with dcpromo |
| 15:19.52 | Hmmhesays | in mother russia, the giddy's up you |
| 15:20.04 | Katty | teehee |
| 15:21.23 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 15:21.28 | aio | is there a way to install * from source without destroying an already-installed version? |
| 15:21.38 | aio | i don't see a configure script. |
| 15:23.00 | [TK]D-Fender | file[laptop] : I got my orders in... all of them :) Got my SPA-941 set up at my desk right now. |
| 15:23.00 | chapeaurouge | would the TE411P work with ISDN BRI ? |
| 15:23.10 | VoIPA | Here is the re-invite for my hold problem: |
| 15:23.12 | VoIPA | <-- SIP read from 172.16.100.1:5060: |
| 15:23.12 | VoIPA | INVITE sip:12567055012@172.16.100.40 SIP/2.0 |
| 15:23.12 | VoIPA | Via: SIP/2.0/UDP 172.16.100.1:5060 |
| 15:23.12 | VoIPA | From: "Randa Green" <sip:3129482994@172.16.100.1:5060>;tag=3757.8162 |
| 15:23.12 | VoIPA | To: <sip:12567055012@172.16.100.40:5060>;tag=as636ff348 |
| 15:23.13 | VoIPA | Call-ID: 0800.20C2.37D6.4383.3757.8162@172.16.100.1 |
| 15:23.15 | VoIPA | CSeq: 6075 INVITE |
| 15:23.17 | VoIPA | Content-Length: 151 |
| 15:23.19 | VoIPA | Max-Forwards: 70 |
| 15:23.21 | Hmmhesays | nice |
| 15:23.21 | VoIPA | Contact: sip:3129482994@172.16.100.1:5060 |
| 15:23.23 | VoIPA | Content-Type: application/sdp |
| 15:23.25 | Hmmhesays | ~pastebine |
| 15:23.26 | VoIPA | Session-Expires: 1800;refresher=uac |
| 15:23.26 | hypa7ia | dude pastebin |
| 15:23.27 | lme | argh |
| 15:23.27 | VoIPA | Supported: 100rel |
| 15:23.27 | Hmmhesays | ~pastebin |
| 15:23.28 | jbot | pastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 15:23.29 | VoIPA | Supported: timer |
| 15:23.31 | VoIPA | <PROTECTED> |
| 15:23.33 | VoIPA | v=0 |
| 15:23.34 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 15:23.35 | VoIPA | o=VocalData 55264 27632 IN IP4 172.16.100.1 |
| 15:23.37 | VoIPA | s=VocalData SIP Call |
| 15:23.39 | VoIPA | c=IN IP4 172.16.100.1 |
| 15:23.41 | VoIPA | t=0 0 |
| 15:23.43 | VoIPA | m=audio 65000 RTP/AVP 0 |
| 15:23.45 | VoIPA | a=ptime:20 |
| 15:23.47 | VoIPA | a=sendonly |
| 15:23.51 | VoIPA | sorry |
| 15:23.54 | Hmmhesays | ~pastebin |
| 15:23.56 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 15:24.07 | Hmmhesays | paste it there |
| 15:24.08 | VoIPA | http://pastebin.ca/30308 |
| 15:24.17 | chapeaurouge | would the TE411P work with ISDN BRI ? |
| 15:24.18 | VoIPA | new to this... my apologies |
| 15:24.54 | file[laptop] | it might be the sendonly |
| 15:25.21 | lme | chapeaurouge: the 101P does |
| 15:25.49 | lme | chapeaurouge: te110P sorry |
| 15:26.24 | VoIPA | BTW, this worked fine under cvs-head from a couple of months ago |
| 15:26.27 | lme | chapeaurouge: mmhmm... I'm reconsidering that... you want to use te411P on a bri line ? |
| 15:26.50 | aio | has anyone tried to do a make install after running make and then chrootting to some directory so that everything will be self-contained? would that work or would it break something? it should work if everything is statically linked, right? |
| 15:26.52 | chapeaurouge | lme, yes. or a card that works. I am looking around |
| 15:26.52 | lme | chapeaurouge: or at the same time than another device supporting bri |
| 15:27.02 | Hmmhesays | paste the original invite, and the re invite |
| 15:27.19 | BladeRunner05 | I have to config asterisk to use a sip_proxy to allow incoming call and allow certain outgoing call, any help is apreciated |
| 15:27.23 | Hmmhesays | actually a sip debug from start to end of the call would be cool |
| 15:27.29 | lme | chapeaurouge: tdm411P does not bri... only pri.. You need a bri card from ie junghanns.net |
| 15:27.41 | VoIPA | OK, just a min |
| 15:27.42 | chapeaurouge | lme, i currently have 2 ISDN BRI lines, which i may want to double, to connect to the PSTN |
| 15:27.51 | chapeaurouge | lme, ok will look there. thx |
| 15:28.37 | lme | chapeaurouge: try the quadbri |
| 15:28.48 | chapeaurouge | lme, yep, was looking at that :) |
| 15:29.11 | lme | chapeaurouge: having quadbri + TE110P for a t2 interconnect with a pabx working great here... |
| 15:29.48 | lme | chapeaurouge: do NOT try the bristuff 0.3.0-PRE1.... Heavy disconnect problems for me... |
| 15:30.08 | chapeaurouge | ok |
| 15:30.14 | VoIPA | Here is the first invite: |
| 15:30.16 | VoIPA | http://pastebin.ca/30310 |
| 15:31.32 | VoIPA | and here is the re-invite: |
| 15:31.34 | VoIPA | http://pastebin.ca/30311 |
| 15:32.51 | chapeaurouge | lme, so no pb with * with these card? |
| 15:33.41 | Hmmhesays | why do those look near identical |
| 15:34.28 | lme | chapeaurouge: you'll have to use the bristuff package... So, * will not be always up to date, depending on reactivity of junghanns |
| 15:34.55 | VoIPA | All the signalling is from a softswitch |
| 15:35.07 | VoIPA | Asterisk is being used as a sip to PSTN gateway only |
| 15:35.18 | lme | chapeaurouge: that's the only point i found... |
| 15:35.26 | VoIPA | The phones are hidden behind the softswitch |
| 15:36.28 | chapeaurouge | lme, ok. I guess there is not much other choice anyway right? |
| 15:37.24 | *** join/#asterisk DrDeke (i=dekemar@deculator.engin.umich.edu) |
| 15:37.27 | *** part/#asterisk __a (n=a@85.100.123.10) |
| 15:38.28 | Hmmhesays | so your softswitch is issuing the reinvite? |
| 15:38.36 | VoIPA | yes. |
| 15:40.16 | lme | chapeaurouge: there is another manufacturer but i can't remember his name... But problem will be the same... you'll have to patch libpri & asterisk to support the card... |
| 15:40.30 | chapeaurouge | lme, ok. thanks a lot for your tips :) |
| 15:40.56 | lme | u'r welcome ! |
| 15:41.04 | hypa7ia | sangoma? |
| 15:41.25 | hypa7ia | lme: was that who you were thinkign of? |
| 15:41.39 | lme | correct ! |
| 15:41.44 | chapeaurouge | kool |
| 15:45.45 | *** join/#asterisk testmachine (n=assink@ip237-239-58-62.adsl.versatel.nl) |
| 15:45.56 | testmachine | hm, anyone using asterisk on ubuntu here? |
| 15:46.16 | testmachine | there is an gtk-console in the ubuntu repositories |
| 15:46.46 | testmachine | but i cant start it or something |
| 15:47.12 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 15:48.06 | VoIPA | Hmmhesays: any ideas? |
| 15:48.53 | jontow | my opinion of the graphical consoles.. |
| 15:48.59 | jontow | useless, absolutely useless, and wasteful |
| 15:49.11 | chapeaurouge | lme, can i PM you? |
| 15:49.14 | jontow | run asterisk on a headless, X-less machine.. then ssh to it from your workstation, running, X, in an xterm |
| 15:49.18 | jontow | "screen -x" |
| 15:49.24 | testmachine | i configured everything by commandline now |
| 15:49.26 | jontow | that'll get you a 'graphical console' |
| 15:49.41 | jontow | you have to anyway.. the gtk console is buggy and quite slow |
| 15:49.50 | jontow | and it doesn't allow you to do anything more than the CLI does |
| 15:50.06 | jontow | except that the scrollbar sucks, and it doesn't track well enough :) |
| 15:50.20 | jontow | the standard CLI in screen(1) will give you all of that, plus some |
| 15:50.39 | burtonez | hi, anyone ever try to flash gs100 (grandstream) phone with jtag adapter? becouse my phone is dead :( |
| 15:50.41 | testmachine | but i would like too see the graphical way too |
| 15:50.44 | jontow | (a configurable scrollback buffer that doesn't need recompiling... pasting between session windows, freeze-frame, etc) |
| 15:50.44 | testmachine | just for testing |
| 15:50.57 | jontow | so do like i said, and get a machine running X, ssh to your * box and run asterisk in screen |
| 15:51.00 | jontow | :P |
| 15:51.03 | chapeaurouge | lme, you're french! heh :P |
| 15:51.10 | testmachine | hehe |
| 15:51.13 | testmachine | asterisk -r :P |
| 15:51.15 | *** join/#asterisk simong (n=simong@h100n1fls34o884.telia.com) |
| 15:51.16 | testmachine | will do the job |
| 15:51.25 | jontow | yeah, but you don't save progress that way.. |
| 15:51.34 | jontow | anything that has happened in the past you won't see ;) |
| 15:51.39 | testmachine | hmmk |
| 15:52.20 | testmachine | thats true |
| 15:52.20 | testmachine | nah, everything inside our network here runs |
| 15:52.20 | testmachine | i only have to connect remotely |
| 15:52.20 | jontow | ok. |
| 15:52.20 | testmachine | with an isdn card |
| 15:52.25 | testmachine | even the dutch voicemail ;-) |
| 15:52.35 | jontow | so you dial into servers? |
| 15:52.40 | Hmmhesays | the from and to fields should contain the ip of the server issiuing the reinvite |
| 15:52.58 | testmachine | we want to connect it on our phone central thing |
| 15:53.14 | jontow | hehe, i think you've got some reading to do :) |
| 15:53.33 | testmachine | hehe |
| 15:53.38 | testmachine | prolly |
| 15:53.52 | testmachine | or just test 99999999999 ways ;P |
| 15:53.54 | VoIPA | The from field does, but the to field should be the endpoint, should it not? |
| 15:53.58 | testmachine | what i am used to do |
| 15:54.01 | testmachine | *grin* |
| 15:54.07 | VoIPA | endpoint meaning asterisk, in this case |
| 15:54.17 | jontow | no, you really need to do some reading |
| 15:54.25 | jontow | trial and error doesn't get you very far with asterisk |
| 15:54.31 | testmachine | yep i know |
| 15:54.31 | jontow | (... until after you've read) |
| 15:54.47 | testmachine | but extern its pretty hard? |
| 15:54.53 | jontow | what? |
| 15:54.59 | Hmmhesays | VoIPA: if you put put a packet sniffer on the network and you should see 4 invites |
| 15:55.03 | testmachine | go extern with asterisk |
| 15:55.08 | testmachine | isdn |
| 15:55.12 | VoIPA | 4? |
| 15:55.13 | testmachine | internal its easy to configure |
| 15:55.27 | jontow | ok.. isdn works from what i know, but we don't even have it in this area so i can't offer advice |
| 15:55.39 | testmachine | ah ok |
| 15:55.40 | testmachine | :) |
| 15:55.47 | Hmmhesays | asterisk --> server server --> phone server-->phone server --> asterisk |
| 15:56.03 | VoIPA | Oh, phone is skinny, softswitch is converting to sip |
| 15:56.04 | jontow | however.. they make this thing now, ethernet... ;) you can ssh into the machine to do whatever you want, testing over IP only, until you've done your research on the ISDN cards |
| 15:56.26 | VoIPA | only two invites between softswitch and asterisk |
| 15:56.32 | testmachine | :) |
| 15:56.53 | jontow | chan_capi is what i hear a lot of people are using for ISDN -- i honestly know nothing about it. |
| 15:56.54 | testmachine | i just want to test everything |
| 15:57.01 | testmachine | what i can test with asterisk |
| 15:57.10 | jontow | your creativity is your only limit |
| 15:57.11 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 15:57.17 | testmachine | jup |
| 15:57.40 | VoIPA | Hmmhesays: Asterisk never sees the phone, it is completely hidden by the softswitch |
| 15:57.59 | Hmmhesays | it has to if a reinvite is going to work |
| 15:58.12 | testmachine | nah, somethimes i use vnc here in the network |
| 15:58.23 | testmachine | becouse i can get more terminals |
| 15:58.28 | testmachine | without 9999999999 connections |
| 15:58.38 | testmachine | so not only ssh |
| 15:59.41 | VoIPA | The reinvite takes the stream off a session border controller (172.16.100.100) and redirects it to the softswitch itself (172.16.100.1) |
| 16:00.00 | VoIPA | You can think of the session border controller as the phone in this scenario |
| 16:00.23 | VoIPA | The SDP shows these IPs in the invites |
| 16:01.00 | Hmmhesays | are you originating the call from asterisk? |
| 16:01.03 | puzzled | anyone know what replaced the StripLSD app in 1.2? |
| 16:01.11 | testmachine | but its 5 o clock |
| 16:01.12 | testmachine | byebye |
| 16:01.13 | testmachine | :) |
| 16:01.30 | [TK]D-Fender | mmmm LSD :D |
| 16:01.38 | VoIPA | The original invite comes from the softswitch and the switch tells asterisk to stream to and listen to the SBC |
| 16:01.59 | Hmmhesays | ahh so I have this backwards |
| 16:02.21 | mocker | I'm having a problem w/ MusicOnHold. I've setup mpg123, the musiconhold.conf, and an extension to call in. |
| 16:02.28 | VoIPA | Asterisk then places the call on the PSTN using a TE410P |
| 16:02.29 | mocker | When I call the extension I get no music though. |
| 16:02.31 | mocker | ;) |
| 16:02.39 | VoIPA | :-) |
| 16:03.23 | *** join/#asterisk Dead-Bum (n=Satan@tor/session/x-066730614c0d8031) |
| 16:03.55 | Hmmhesays | ok then yes your invites look ok, I was confused for a minute |
| 16:03.55 | VoIPA | Hmmhesays: Asterisk is 172.16.100.40 |
| 16:03.59 | Hmmhesays | yeah |
| 16:04.26 | VoIPA | Does asterisk assume hold is wanted when it receives an invite to listen only? |
| 16:04.30 | Dead-Bum | So is asterisk 1.2 usable enough to replace the old version in a production environment? |
| 16:04.32 | *** join/#asterisk ramtha (n=ramtha@gw.01063telecom.de) |
| 16:04.56 | ramtha | is there anything diffrent between meetme with zaptel hardware or without? |
| 16:04.59 | Hmmhesays | what console output do you get? |
| 16:05.27 | Hmmhesays | are you still having trouble with meetme ramtha? |
| 16:05.58 | ramtha | Hmmhesays: yep |
| 16:06.01 | VoIPA | It says something about starting music on hold on Zap/1-1 or whatever PSTN channel is in use, and then it immediatly says stopping MOH because I haven't configured the moh.conf |
| 16:06.06 | ramtha | can not figure out what the prob is |
| 16:06.15 | VoIPA | If I do a sip show channels, it shows that channel as being on hold |
| 16:06.17 | ramtha | on three asterisk mashines invalid-conference |
| 16:06.20 | Hmmhesays | VoIPA you got me there, I have no idea |
| 16:06.33 | Hmmhesays | whats it worth for me to just fix it for you? |
| 16:06.39 | Hmmhesays | ;) |
| 16:06.59 | `Sauron | Hmm he said. |
| 16:07.00 | VoIPA | :-) I have a ticket open with digium on this that I'm waiting for |
| 16:07.10 | Hmmhesays | that should be interesting VoIPA |
| 16:07.11 | ramtha | meetme.conf is correct. extensions.conf too |
| 16:07.20 | ramtha | but still getting invalid-conf |
| 16:07.21 | VoIPA | Plus, mark sometimes does free programming for me in exchange for homebrew |
| 16:07.21 | Hmmhesays | did you actually load ztdummy? |
| 16:07.26 | ramtha | yes |
| 16:07.30 | Hmmhesays | mmm homebrew |
| 16:07.33 | Zeeek | ramtha what version * ? |
| 16:07.41 | Hmmhesays | I was down in msp last week and had some of my uncles |
| 16:07.42 | VoIPA | :-) |
| 16:07.56 | VoIPA | I have an ESB waiting to be kegged |
| 16:07.58 | Hmmhesays | I believe he has perfected the art |
| 16:08.08 | VoIPA | may have to bribe markster with some |
| 16:08.12 | zoa | what the hell is homebrew ? |
| 16:08.16 | Hmmhesays | beer |
| 16:08.18 | VoIPA | beer |
| 16:08.21 | zoa | that tobacco tasting thing that you people call beer ? |
| 16:08.22 | ramtha | zeeek: 1.0.7 |
| 16:08.25 | syzygyBSD | is there a way to perform an action after a user has been in a queue for so long, besides play an annoncement |
| 16:08.26 | ramtha | and cvs head |
| 16:08.31 | zoa | thats not beer!!!! |
| 16:08.35 | fugitivo | tobacoo tasting?? |
| 16:08.41 | fugitivo | did you ever drink beer? |
| 16:08.42 | VoIPA | WTF??? |
| 16:08.43 | zoa | i was at astricon :) |
| 16:08.53 | Hmmhesays | you're brewing it wrong if your homebrew comes out tasting like tabacco |
| 16:08.56 | zoa | i had the worst beer of my life there |
| 16:09.03 | Hmmhesays | fosters? |
| 16:09.05 | VoIPA | Ask markster about my beer |
| 16:09.21 | fugitivo | you have to take your cigarette out of your mouth before drinking beer :) |
| 16:09.26 | VoIPA | name is Brian in Huntsville |
| 16:09.35 | Hmmhesays | good call fugitivo |
| 16:09.38 | syzygyBSD | or is timeouts the best way to handle it |
| 16:09.42 | VoIPA | lol |
| 16:10.01 | VoIPA | Thanks for your help, Hmmhesays. |
| 16:10.08 | [TK]D-Fender | syzygyBSD : You can have it dump them from the queue to leave VM etc as well as the fact you can offer 1-digit exit extensions so the caller has the OPTION of what they would like to do instead of waiting. |
| 16:10.16 | Hmmhesays | ramtha if your ztdummy module is loaded, you must have a syntax error in your extensions.conf or meetme.conf |
| 16:10.20 | Hmmhesays | VoIPA, np |
| 16:10.33 | VoIPA | (y) 'm out |
| 16:10.33 | syzygyBSD | Thanks [TK]D-Fender |
| 16:10.35 | *** part/#asterisk VoIPA (n=VoIPA@mail.kancharla.com) |
| 16:11.15 | syzygyBSD | is there a ringing mp3? |
| 16:11.55 | [TK]D-Fender | syzygyBSD : I think somewhere but why on earth would you want to subject them to that? |
| 16:12.08 | [TK]D-Fender | ICK |
| 16:12.19 | *** join/#asterisk jmacz (n=jmacz@208.195.215.48) |
| 16:12.36 | syzygyBSD | client wants it to ring for 20 seconds before they realize they are in a queue |
| 16:13.51 | trixter | it would be better if the customer never realizes they are in a queue :P |
| 16:14.04 | Beirdo | make a MOH with a ringing sound |
| 16:14.05 | syzygyBSD | ya, but ringing for 5 minutes... |
| 16:14.21 | Beirdo | it can't be hard to make |
| 16:14.24 | syzygyBSD | Ya, that is what I was going to do, just wondering if the ringing mp3 was around |
| 16:14.42 | trixter | use bananaphone.mp3 it has a ringing sound in it |
| 16:14.44 | syzygyBSD | ya, monitor a line that is ringing |
| 16:15.31 | trixter | sunos4 and solaris 2.3 shipped with ringing.au I am sure htat you can find those somewhere |
| 16:15.37 | trixter | sunsite used to have em, transcoding isnt that hard |
| 16:16.03 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
| 16:16.35 | paryl | i don't get it... ever since 1.2 came out i can't compile it... i always get "pbx_dundi.c:1409: error: âZ_OKâ undeclared (first use in this function)" does anyone know what the problem might be? |
| 16:17.50 | Zeeek | anyone here use asterlink? |
| 16:17.59 | jlewis | paryl: did that paste badly, or are there really 8-bit chars in the error |
| 16:18.08 | fugitivo | paryl: are you going to use dundi? |
| 16:18.12 | Hmmhesays | imagine 5 minutes of ringing |
| 16:18.15 | *** join/#asterisk mrtwister (n=mrtwiste@cable-9-42.cgates.lt) |
| 16:18.25 | paryl | jlewis: yeah, i always get them throughout the compile |
| 16:18.26 | [TK]D-Fender | syzygyBSD : I think you can jsut do that in the dial-plan : "exten => 1,1,Ringing", "exten => 1,2,Wait(20)", "exten => 1,2,Queue(wastingmytime)" |
| 16:18.30 | *** join/#asterisk cnet2 (n=jjohn@adslnat-sanjose-4.ice.co.cr) |
| 16:18.42 | paryl | fugitivo: not really... at least not for now |
| 16:18.46 | mrtwister | hi.. for load-balancing on T1/E1 line (Zap/g0) should i use superdial macro or specify settings in zapata.conf? |
| 16:18.56 | file[laptop] | Zeeek: I do, but then again - I'm biased |
| 16:19.14 | mrtwister | hi.. for load-balancing on T1/E1 line (Zap/g0) should i use superdial macro or specify settings in zapata.conf? now it fill calls from port 1 to higher.. |
| 16:19.43 | fugitivo | paryl: go to dir pbx, edit makefile, remove pbx_dundi.so from PBX_LIBX=, i think * will still work without that |
| 16:20.01 | syzygyBSD | [TK]D-Fender: I want the option for an agent to answer it before the ringing stops |
| 16:20.03 | [TK]D-Fender | mrtwister : as in to find the first available free channel to dial out from? |
| 16:20.18 | syzygyBSD | I could probably do it with a local channel then |
| 16:20.30 | tzafrir_laptop | mrtwister, the super-dial macro is intended for use in extensions.conf. You still need to configure the group settings in zapata.conf |
| 16:21.10 | [TK]D-Fender | syzygyBSD : Well the only way to do that while in queue is to leave the ringing MP3 as the PERMANENT MOH... which would be *&^#$ annoying. You could simply have your queue use ringing instead of MOH when it is actually DIALING a rep though. |
| 16:21.30 | tzafrir_laptop | A nice local channel to point to a Playbck(something) extensions, MusicOnHold, or whatever |
| 16:22.04 | mrtwister | thanks for help! |
| 16:22.39 | syzygyBSD | [TK]D-Fender: I was going to have one queue setup with ringing as the permanent MOH with a timeout of 20, then another with normal music annoncements, etc. and put the agents in both |
| 16:23.02 | *** join/#asterisk slak- (i=slak@make-world.info) |
| 16:23.10 | slak- | hey how do i set calling id for SIP? |
| 16:23.30 | slak- | exten => _91NXXNXXXXXX,1,SetCallerID(##########) doesnt work anymore as it did for iax2 |
| 16:24.14 | viperdude | slak-: i think it depends if your SIP provider allows you to set CallerID |
| 16:24.35 | slak- | they did for iax2 via that way |
| 16:24.49 | paryl | fugitivo: that worked... what will i not be able to do without dundi compiled? (I don't really know what it does) |
| 16:24.51 | viperdude | ok just a thought I know some in the UK don't |
| 16:25.33 | *** join/#asterisk lehel (n=lehel@82.79.20.17) |
| 16:25.38 | lehel | hello |
| 16:27.43 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 16:28.09 | Nugget | http://flightaware.com/live/special/N225GV/N225GV_movie.rvt <-- yesterday's nike jet flight track (animated) |
| 16:30.41 | DrDeke | nugget! what are you doing in here? |
| 16:30.45 | DrDeke | :) |
| 16:31.10 | DrDeke | i thought i was in #distributed for a minute there |
| 16:31.42 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 16:33.06 | Nivex | Nugget: I don't have QT installed but I see the static images. what am I looking at? |
| 16:33.39 | Zeeek | . |
| 16:34.17 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-17.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 16:34.29 | Nugget | it's the flight track of that nike jet that had landing gear problems yesterday |
| 16:34.53 | Nivex | hadn't heard about that. |
| 16:34.56 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:35.19 | Nugget | http://news.google.com/news?hl=en&ned=&q=nike+jet |
| 16:35.24 | twisted[asteria] | this is a no mooing channel. |
| 16:35.26 | jlewis | interesting...same system builds zaptel from cvs-stable without complaint, but zaptel 1.2.0 bombs out with lots of kernel inlcude issues |
| 16:35.48 | *** join/#asterisk trixter (i=trixter@65.172.209.246) |
| 16:35.54 | twisted[asteria] | :P |
| 16:35.55 | DrDeke | That's OK I'm hiding in the corner anyway on account of running f@h instead of d.net... So I won't moo over here ;) |
| 16:36.04 | twisted[asteria] | eeek |
| 16:36.25 | twisted[asteria] | i can't believe you admitted that in public |
| 16:36.26 | twisted[asteria] | :P |
| 16:36.36 | DrDeke | haha |
| 16:36.57 | DrDeke | me either |
| 16:37.07 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 16:37.07 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:37.15 | DrDeke | well hey it's cold in this office... i have to keep warm some how and this 3.4ghz cpu running at full steam helps ;) |
| 16:37.21 | twisted[asteria] | hahaha |
| 16:37.55 | Hmmhesays | thank god for redbull |
| 16:38.00 | Hmmhesays | and beer too |
| 16:38.02 | twisted[asteria] | yeah |
| 16:38.06 | twisted[asteria] | oh, hey Hmmhesays |
| 16:38.10 | twisted[asteria] | i work 2 jobs now :P |
| 16:38.18 | [TK]D-Fender | Tastes nifty, but does nothing for me... especially at that cost |
| 16:38.21 | twisted[asteria] | telecom by day, barback/bouncer by night |
| 16:38.22 | trixter | Hmmhesays: do you do bullbombs in your beer? |
| 16:38.25 | Hmmhesays | hey twisted[asteria] been awhile |
| 16:38.32 | iDunno | hmmmm - beer |
| 16:38.42 | Hmmhesays | no, I don't mix them, beer to forget the day, redbull to survive it |
| 16:38.44 | fugitivo | don't forget whisky |
| 16:38.53 | iDunno | redbull isn't for drinking. |
| 16:39.00 | iDunno | it's alloy wheel cleaning fluid. |
| 16:39.08 | [TK]D-Fender | file : The SPA-941 is just a mini-Cisco, but at a much better price-point :) |
| 16:39.10 | fugitivo | that's coke |
| 16:39.17 | twisted[asteria] | iDunno, no, that's cocacola |
| 16:39.33 | iDunno | heh - fair play. |
| 16:39.39 | twisted[asteria] | you can shine a penny with coke |
| 16:39.42 | iDunno | redbull is just to give you ulcers, then? |
| 16:39.42 | DrDeke | I dunno; I don't trust Linksys. |
| 16:39.43 | file | [TK]D-Fender: neat |
| 16:39.51 | fugitivo | iDunno: YES |
| 16:39.52 | *** join/#asterisk ihf (n=ihf@jstorpex8.Princeton.EDU) |
| 16:39.58 | [TK]D-Fender | twisted[asteria] : and EAT a penny too.. try leaving a fork in a can over night.... |
| 16:40.17 | twisted[asteria] | [TK]D-Fender, yeah, i left a spoon in one once |
| 16:40.25 | [TK]D-Fender | DrDeke : Well I've just unpacked and set it up. Looks and feels great |
| 16:40.36 | fugitivo | [TK]D-Fender: how much did you pay for it? |
| 16:40.45 | [TK]D-Fender | 150$USD (normal retail) |
| 16:40.45 | ihf | can someone please tell me if there exists a busy-redial function that will work with a POTS line and the Zaptel TDM400 card? I realize that busy detection is not necessarily reliable. |
| 16:40.48 | Hmmhesays | i haven't touched an sipura unit in awhile... or any voip for that matter |
| 16:40.49 | twisted[asteria] | BUT, sour milk can eat through plastic in no time flat |
| 16:41.09 | mut | no |
| 16:41.11 | synthetiq | how can a user sent multiple voicemails to people with out making a change int eh dialplan? |
| 16:41.17 | mut | i had mik sitting in a jug outside for a month |
| 16:41.18 | mut | milk |
| 16:41.22 | synthetiq | suppsoedly other analog pbx's have that feature |
| 16:41.28 | mut | it didn't eat any plastic |
| 16:41.41 | twisted[asteria] | mut, i've seen this happen.. i've done impromptu experiments in my kitchen |
| 16:41.47 | [TK]D-Fender | synthetiq : You could always send YOURSELF a message and then forward it to each box 1 by 1..... |
| 16:41.50 | file | twisted[asteria]: O.o |
| 16:41.55 | *** join/#asterisk Assid (n=assid@203.115.64.62) |
| 16:41.56 | mut | all that happened in mine was a big hunk formed at hte top |
| 16:41.58 | twisted[asteria] | file, long time ago. |
| 16:41.58 | fugitivo | synthetiq: copy the files to the mailboxes you want |
| 16:42.01 | mut | and it eventually turned yellow |
| 16:42.03 | Hmmhesays | why does all the sipura stuff on voip-supply say linksys in front of it? |
| 16:42.16 | twisted[asteria] | mut, must have been skim milk you used |
| 16:42.16 | Qwell | because Linksys owns Sipura |
| 16:42.20 | mut | 2% |
| 16:42.24 | twisted[asteria] | close enough |
| 16:42.24 | [TK]D-Fender | Hmmhesays : because Linksys bought them and they are rebranding everything |
| 16:42.25 | synthetiq | the user will not have access to the console to copy the files |
| 16:42.29 | twisted[asteria] | try with whole or D |
| 16:42.29 | Assid | hrmm.. anyone know the default user/pass for poly501? |
| 16:42.31 | mrtwister | hi. situation. T1 in channel bank, gsm lines... i calling (iax -> asteirsk -> zap/g0), i hear remote person, he dont hear me. i turning off echo cancel, calling - i hear myself, remote person, he hear his echo, but not me. what i have to look now, for this one-way audio issue? |
| 16:42.32 | fugitivo | Hmmhesays: cisco owns linksys, and linksys owns sipura |
| 16:42.38 | Hmmhesays | ahh |
| 16:42.50 | [TK]D-Fender | Assid : Polycom/456 |
| 16:42.51 | fugitivo | synthetiq: make a script |
| 16:42.51 | puzzled | anyone know if the call-limit variable set in sip.conf for a phone has a variable name that can be read/modified in a dialplan? |
| 16:42.58 | mut | i'll do that |
| 16:43.09 | twisted[asteria] | puzzled, not that I'm aware of |
| 16:43.15 | Zeeek | The proof of the pudding is in the eating.... The XXX SPA941 works great |
| 16:43.21 | twisted[asteria] | puzzled, but you can use setgroup/checkgroup to perform the same tasks |
| 16:43.23 | twisted[asteria] | -s |
| 16:43.29 | Qwell | They have a XXX version of the 941? |
| 16:43.34 | [TK]D-Fender | synthetiq : How many destination boxes do you have in mind |
| 16:43.34 | Qwell | that's hot |
| 16:43.35 | puzzled | twisted[asteria]: cool thanks for the tip |
| 16:43.39 | Zeeek | XXX is "fillin the brand here" |
| 16:43.40 | twisted[asteria] | puzzled, np |
| 16:43.49 | synthetiq | 10 or so |
| 16:44.05 | synthetiq | imean ic an do it for this one numebr but i dont know who else wants the option |
| 16:44.07 | twisted[asteria] | mmm...cookie. |
| 16:44.09 | synthetiq | number |
| 16:44.14 | [TK]D-Fender | Just leave yourself the message and forward it from within your mailbox. |
| 16:44.20 | fourcheeze | If i'm generating my extensions.conf statically from mysql is it feasible to have one context per user? |
| 16:44.27 | fourcheeze | does that cause any problems for * ? |
| 16:44.32 | puzzled | twisted[asteria]: dou you know if call-limit is not set in sip.conf is it then set by default to 1? |
| 16:44.38 | synthetiq | well in an awaya pbx u can creeat a list |
| 16:44.43 | synthetiq | create |
| 16:44.46 | synthetiq | and nortel |
| 16:44.47 | twisted[asteria] | puzzled, no, it is controlled by your UA |
| 16:45.01 | [TK]D-Fender | synthetiq : you can do FAR more with * but you'd have to create some dial-plan logic for it. |
| 16:45.15 | Assid | thanks [TK]D-Fender |
| 16:45.25 | [TK]D-Fender | synthetiq : I can picture what it would take right now, and its not that hard... |
| 16:45.32 | [TK]D-Fender | Assid : no, any time |
| 16:45.34 | [TK]D-Fender | np* |
| 16:45.40 | twisted[asteria] | haha |
| 16:45.41 | synthetiq | what would you do |
| 16:45.44 | twisted[asteria] | JUST SAY NO |
| 16:45.57 | puzzled | twisted[asteria]: weird cause if I don't set it it defaults to 1 and I get a busy when calling myself. if I set it to 2 I can call myself |
| 16:46.09 | synthetiq | suppsoedly peopel can call up and generate the list |
| 16:46.18 | twisted[asteria] | puzzled, make sure you dont' have it set in the [general] section |
| 16:46.29 | fugitivo | synthetiq: you can do all that with agi scripting |
| 16:46.31 | fugitivo | and more |
| 16:46.37 | [TK]D-Fender | synthetiq : I'd poll for box#'s one at a time with READ's, increment a counter, assemble the VMB string from there and then dump you into "voicemail({$mypileofboxes}) |
| 16:46.38 | puzzled | twisted[asteria]: had missed that one. that's another cookie :) |
| 16:46.45 | twisted[asteria] | :) |
| 16:46.45 | file | do you really want to hurt me? |
| 16:46.54 | mut | YES! |
| 16:47.01 | [TK]D-Fender | :O |
| 16:47.15 | Assid | [TK]D-Fender: i couldnt get into the ftp tho |
| 16:47.16 | Inkubot | a pickgroup can go from 0 to 63 ? |
| 16:47.23 | [TK]D-Fender | ... and then hands him back his soose :D |
| 16:47.28 | synthetiq | ahh crud |
| 16:47.35 | [TK]D-Fender | Assid : into FTP? |
| 16:47.46 | Assid | doesnt the phone support ftp or something? |
| 16:48.04 | *** part/#asterisk ramtha (n=ramtha@gw.01063telecom.de) |
| 16:48.05 | [TK]D-Fender | Assid : You don't FTP INTO the phone.... the phone can use FTP to pick up its config from a server |
| 16:48.11 | Assid | OH |
| 16:48.13 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 16:48.22 | *** part/#asterisk ihf (n=ihf@jstorpex8.Princeton.EDU) |
| 16:48.25 | [TK]D-Fender | Assid : and that part works VERY well. |
| 16:49.02 | Assid | man.. this waiting period is very bad |
| 16:49.09 | Assid | spa-841 was soo much easier to configure |
| 16:49.14 | doug | win 21 |
| 16:49.15 | Assid | i dont know what im doing here |
| 16:50.56 | docelmo | YIPPIE! |
| 16:52.39 | yxa | are there search functions for the user directory for higher-end polycom phones? i'm using the 301 |
| 16:52.46 | Assid | Nov 22 11:52:17 NOTICE[1387]: chan_sip.c:10685 handle_request_register: Registration from '<sip:204@192.168.1.3>' failed for '192.168.1.37' - Not a local SIP domain |
| 16:52.50 | Assid | hrmm |
| 16:53.18 | *** join/#asterisk Rav1974 (n=r@static-70-19-119-112.ny325.east.verizon.net) |
| 16:53.51 | Rav1974 | hi guys, I'm compiling asterisk & I wanted to know if I can use the march=pentium4 in the make file? |
| 16:54.45 | [TK]D-Fender | Assid : The SPA-941 I'm sure is nearly identical, but its the quality of the phone that really changed. |
| 16:54.58 | Assid | spa better? |
| 16:55.07 | [TK]D-Fender | 941 >>>> 841 |
| 16:55.16 | Hmmhesays | anyone got a new iaxy they're using? |
| 16:55.26 | [TK]D-Fender | Its a Cisco phone with Linksys written on it :) |
| 16:55.38 | *** join/#asterisk jtoy (n=toy@gramthree.hsd1.ma.comcast.net) |
| 16:55.44 | jtoy | anyone running 1.2 on mac? |
| 16:55.54 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
| 16:56.01 | twisted[asteria] | jtoy, ;) |
| 16:56.03 | jtoy | I stat up with -vvvc , there are no error messages, but it doesnt start |
| 16:56.24 | twisted[asteria] | it does on my box |
| 16:57.11 | oej | It starts on my box as well |
| 16:57.24 | *** join/#asterisk madeddie (i=madeddie@raina.madtech.cx) |
| 16:57.38 | X-Files | twisted[asteria]: one question, why i can't use *8 in ## (transfer) ? |
| 16:58.27 | twisted[asteria] | because *8 is PICKUP for a ringing phone in your callgroup/pickupgroup... not transfer. |
| 16:58.39 | X-Files | :( |
| 16:58.47 | jtoy | can I see on of your modules.conf ? I think the problem is there |
| 16:58.56 | X-Files | twisted[asteria]: but , i wanna in talking pickup other call ringing ... |
| 16:59.07 | twisted[asteria] | no.... |
| 16:59.26 | twisted[asteria] | you'd need another line for that |
| 16:59.37 | }cytrak{ | I can make my asterisk be a like a PSTN for a PBX right ? |
| 16:59.45 | X-Files | twisted[asteria]: :( |
| 16:59.53 | [TK]D-Fender | }cytrak{ : ?? |
| 16:59.58 | chapeaurouge | wow |
| 17:00.01 | X-Files | twisted[asteria]: i can insert beep sound , abount new call ? |
| 17:00.03 | Assid | [TK]D-Fender: i dont know what i gotta change.. but that thing doesnt wanna register |
| 17:00.27 | X-Files | abount = about .. |
| 17:00.33 | [TK]D-Fender | Assid : are you using the web interface or are you provisioning them yet? |
| 17:00.40 | Assid | web interface |
| 17:01.15 | jtoy | twisted[asteria] or oej can i see your modules.conf? |
| 17:01.23 | *** join/#asterisk limo (n=limo@elvira-gw.opf.slu.cz) |
| 17:01.26 | Assid | just submitted... there goes 2 mins of my life i never get back |
| 17:01.41 | oej | jtoy: Mine is the same as the sample config |
| 17:02.02 | oej | Very simple installation |
| 17:02.03 | }cytrak{ | well I want to connect the asterisk to a PBX through a T1 , remove the PSTN T1 line from the PBX |
| 17:02.05 | limo | hello all ! Please can anybody help me with message "Didn't get a frame on channel" on SIP ? Thanx ! |
| 17:02.07 | Assid | i would like to use ftp.. once i learn how to |
| 17:02.13 | [TK]D-Fender | Assid : I'll take a look at how mine looks from the web interface.... |
| 17:02.28 | }cytrak{ | something like an intranet for the phone |
| 17:02.50 | [TK]D-Fender | }cytrak{ : yeah you can do that. so you want * to act as the IVR and then dial DID into your old PBX? |
| 17:02.55 | }cytrak{ | I would be able to use a sip phone behind asterisk to call a phone extension that is in the PBX system |
| 17:03.06 | [TK]D-Fender | }cytrak{ : sure |
| 17:03.11 | limo | This mesage appears only sometimes. Not on all calls.. Please.. |
| 17:03.12 | Assid | [TK]D-Fender: phone is remote.. office is remote.. im not even in the office.. i dont have the manual.. am guessing whatever it is.. so far.. its so so.. |
| 17:03.14 | jtoy | can I see it anyway? |
| 17:03.21 | }cytrak{ | cool gotta go lunch |
| 17:03.23 | limo | Has somebody same problem? |
| 17:04.49 | limo | I cannot find something about this.. If you have some url, please let me know. I think that problem can be in local network which sometimes drops packets maybe ? |
| 17:04.52 | jtoy | haha oej: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+modules.conf |
| 17:05.28 | jtoy | I was going to use that, but the last time you changed it was in 2003 |
| 17:05.40 | oej | Use the one in the distribution instead |
| 17:05.46 | oej | But I don't think much has changed |
| 17:05.54 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 17:06.16 | jtoy | changefd enough that it doesnt work :( |
| 17:06.33 | chapeaurouge | Asterisk::AGI... yummy |
| 17:06.37 | chapeaurouge | anyone played with it? |
| 17:07.15 | *** join/#asterisk pussfeller (n=todd@12.150.129.170) |
| 17:08.23 | jtoy | see, it worked now! |
| 17:08.43 | limo | please.. can somebody help me ? |
| 17:09.36 | Assid | hrmm.. i cant seem to have it register |
| 17:09.47 | [TK]D-Fender | Assid : PM time... |
| 17:09.48 | *** part/#asterisk ful|work (n=fulgas@209.8.233.201) |
| 17:09.48 | *** join/#asterisk ful|work (n=fulgas@209.8.233.201) |
| 17:12.05 | burtonez | <PROTECTED> |
| 17:12.49 | *** part/#asterisk jtoy (n=toy@gramthree.hsd1.ma.comcast.net) |
| 17:14.18 | *** join/#asterisk cnet2 (n=jjohn@adslnat-sanjose-4.ice.co.cr) |
| 17:14.21 | cnet2 | what softphone do you guys suggest? (iax + sip).. Most of the ones I've downloaded, don't even show if you are registered or not!!! (that's basic!) |
| 17:16.18 | Hmmhesays | i use firefly, some people hate it, some like it |
| 17:16.33 | file | picky people |
| 17:16.40 | file | next you'll be asking for one that *works |
| 17:16.43 | file | and that's just crazy |
| 17:16.55 | cnet2 | jqjq |
| 17:16.57 | cnet2 | jaja |
| 17:17.05 | Hmmhesays | heh, for a free softphone it suits my needs |
| 17:17.45 | zoa | Hmmhesays: did you every try idefisk ? |
| 17:17.53 | Hmmhesays | i was just going to mention that too |
| 17:18.03 | Hmmhesays | the last version I had liked to crash |
| 17:18.09 | robin_sz | Hmmhesays, was it exten => 8501,1,Voicemailmain(${EXTEN}) ?? |
| 17:18.13 | zoa | what version is that ? |
| 17:18.20 | Hmmhesays | but I imagine you've updated it in the last 4 months since i've been out of the game |
| 17:18.22 | zoa | if you get it to crash report it ot me |
| 17:18.24 | zoa | aha |
| 17:18.24 | zoa | yes |
| 17:18.28 | Hmmhesays | :) |
| 17:18.34 | zoa | 4 months ago thats like the first month or so |
| 17:18.34 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 17:18.34 | zoa | :) |
| 17:18.40 | file | and... |
| 17:18.52 | jojo | , |
| 17:18.58 | Hmmhesays | robin_sz that would get you to mailbox 8501 if it existed |
| 17:19.06 | robin_sz | grr. |
| 17:19.45 | robin_sz | so what was that _X. stuff about then? |
| 17:19.55 | Hmmhesays | _X. would catch anything longer than 1 digit |
| 17:20.07 | GiabboO | burtonez, i use isdn4linux |
| 17:20.25 | robin_sz | well, surely that would stop me dialing anything except the voicemail>> |
| 17:20.35 | Hmmhesays | not if you set it up right |
| 17:20.44 | Hmmhesays | theres eleventy billion ways you can do it robin_sz |
| 17:20.58 | Hmmhesays | I just gave you a catch all example |
| 17:21.00 | robin_sz | thats the trouble .. but I only want one |
| 17:22.03 | Hmmhesays | exten => <extension number>,<priority>,Voicemailmain(<voice mailbox number>) |
| 17:22.10 | Hmmhesays | man people are freaking picky ;) |
| 17:22.36 | marcus2 | zoa! |
| 17:22.37 | robin_sz | yes ,,, |
| 17:23.31 | Hmmhesays | that'll be 37 cents, for the food i'll have to eat to replace the engergy it took to type that |
| 17:23.40 | robin_sz | so .... exten => 8501,1,Voicemailmain(${EXTEN}) will pick up the users mailbox *assuming it is the same as their extension number)( when they dial 8501 . right. |
| 17:24.09 | Hmmhesays | ${EXTEN} stores the extension dialed |
| 17:24.14 | robin_sz | oh. |
| 17:24.29 | Zeeek | why not use the callerid? |
| 17:24.31 | Hmmhesays | you could also use exten => 8501,1,VoiceMailmain(8501) |
| 17:24.33 | file | it's not a regular variable either so you can't do set or anything on it... ${EXTEN} that is |
| 17:24.49 | robin_sz | what stores their extension number they dialled from then? |
| 17:25.27 | Hmmhesays | you can't dial from an extension |
| 17:25.32 | Assid | Nov 22 12:25:06 NOTICE[1387]: chan_sip.c:10685 handle_request_register: Registration from '<sip:204@192.168.1.3>' failed for '192.168.1.37' - Not a local SIP domain |
| 17:25.40 | robin_sz | hmmm |
| 17:25.47 | Zeeek | Time for http://www.asteriskdocs.org |
| 17:25.48 | Assid | not a local sip domain? |
| 17:25.50 | file | you dial from a phone, crazy concept - I know ;) |
| 17:26.00 | robin_sz | sigh ... but it has a number |
| 17:26.04 | Hmmhesays | http://www.voip-info.org/wiki-Asterisk+variables |
| 17:26.06 | Zeeek | Time for http://www.asteriskdocs.org |
| 17:26.06 | Assid | why is it saying that ? |
| 17:26.08 | GiabboO | ^^ |
| 17:26.08 | Hmmhesays | just for you robin_sz |
| 17:26.12 | robin_sz | I dial 790, it rings ... |
| 17:26.18 | robin_sz | well, it beeps anyway |
| 17:26.26 | robin_sz | usually. |
| 17:26.34 | robin_sz | unless the power is off. |
| 17:26.54 | mrtwister | hi, where i can find info about echo cancelation on sangoma. how to adjuct.... now it is terrible :) |
| 17:26.57 | *** join/#asterisk Gerriall (n=NonYa@209.42.198.18) |
| 17:27.55 | file | robin_sz: see you gotta get over the concept "this extension is this phone" because not really, it's "this extension runs this sets of applications and instructions" |
| 17:28.07 | file | but you could use callerid... |
| 17:28.27 | robin_sz | ${CALLERIDNUM} |
| 17:29.02 | *** join/#asterisk toddf (n=toddf@adsl-65-66-19-158.dsl.okcyok.swbell.net) |
| 17:29.06 | file | if you don't explicitly put it in sip.conf, it'll probably come out to be their username |
| 17:29.14 | Hmmhesays | it would be cool if the sip error code was stored in ${DIALSTATUS} |
| 17:29.27 | file | Hmmhesays: or... another variable! |
| 17:29.42 | Hmmhesays | file, yes you are right |
| 17:29.47 | Hmmhesays | i started to play with that many moons ago |
| 17:29.56 | file | it's a crazy idea XD |
| 17:30.06 | mocker | I'm trying to make an IVR menu option to go to someone's extension. |
| 17:30.17 | mocker | Can someone tell me what the exten line should look like? |
| 17:30.29 | file | exten => 5,1,Goto(users,145,1) |
| 17:30.29 | Hmmhesays | bwhahaha |
| 17:30.41 | file | dial 5 and it'll go to extension 145 in the context users, priority 1 :P |
| 17:30.53 | mocker | file: Thanks. |
| 17:30.56 | mocker | I'll play with that. |
| 17:31.07 | mocker | I was trying Transfer(extension) but it just hung up on me. ;) |
| 17:31.12 | Hmmhesays | now is it worth it for me to go back an take a look at it |
| 17:31.20 | file | mocker: yeah use goto... |
| 17:31.32 | LostFrog | Transfer transfers the other party. |
| 17:31.41 | LostFrog | I believe. |
| 17:32.05 | LostFrog | I don't.. I just pulled that out of my ass. |
| 17:39.33 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:42.14 | LostFrog | If anyone ever needs audiocodes firmware, alliancesystems has them on their website under 'drivers' |
| 17:42.36 | Hmmhesays | i got a 4 port audiocodes box sitting here |
| 17:42.49 | *** join/#asterisk Nukemizer (n=Nuke@67.137.28.163) |
| 17:43.09 | LostFrog | I have four of them, one 108 and 3 104s |
| 17:43.29 | *** join/#asterisk phifli (n=mike@127.sub-70-197-106.myvzw.com) |
| 17:43.53 | phifli | Im having a few issues... i just learned of jitterbuffer although its not stopping it from dropping my NuFone DID call when a packet becomes out of order |
| 17:44.01 | phifli | first time ive ever encountered this situation.. anyone familiar? |
| 17:44.22 | phifli | jitter buffer debugging spits out stuff to stdout although I haven't researched what each char symbolizes (guessing v is voice? ;) |
| 17:47.06 | phifli | ah no one awake to help? :-\ |
| 17:47.15 | robin_sz | ok, so thats users, mailboxes, dial-out over IAX sorted |
| 17:47.27 | robin_sz | all that remains is .,... this poxy Eicon Diva |
| 17:47.59 | phifli | hehe |
| 17:48.17 | robin_sz | oh, and users |
| 17:48.21 | phifli | robin_sz: happen to be familiar with jitterbuffer and out of order packets? :) |
| 17:48.31 | phifli | i wanna get into extensions.ael later today :) seems very nice |
| 17:48.50 | robin_sz | phifli, nope ... never heard of her. |
| 17:48.50 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 17:50.00 | phifli | oh k |
| 17:50.09 | phifli | just something to i guess help you out with lag and dropped packets |
| 17:50.13 | phifli | since it is a UDP protocol heh |
| 17:50.34 | robin_sz | shrug ... |
| 17:50.39 | robin_sz | not that you can tell ... |
| 17:50.56 | robin_sz | thats the trouble with UDP .. you never really know if you got it all |
| 17:51.44 | phifli | lol yeah but its also an awesome protocol for other aspects |
| 17:52.01 | phifli | ie: you dont have to get a reponse for every packet therefore your call isnt lagged as hell as if your home needs inet2 speeds |
| 17:52.10 | phifli | i personally like it, a good base around it and its wonderful |
| 17:52.28 | robin_sz | sigh ... |
| 17:52.31 | robin_sz | <PROTECTED> |
| 17:52.32 | robin_sz | <PROTECTED> |
| 17:53.10 | phifli | hmm lemme check source, ive never saw that before |
| 17:53.18 | phifli | whats CAPI? |
| 17:53.21 | phifli | ive only dealt with SIP/IAX2 |
| 17:53.40 | robin_sz | ISDN |
| 17:53.42 | }cytrak{ | usually when you connect * to a T1 line from the phone company (bellsouth) all you need is to make sure the signalling is the pri_cpe as shown in the digium examples right |
| 17:53.53 | phifli | no capi info oni here... when i do grep -R "CAPI INFO" $ASTHOME |
| 17:54.16 | phifli | so sorry I'm not in the position to help you :( |
| 17:54.20 | phifli | I wanna throw a machine at a NOC and get some lines |
| 17:54.28 | phifli | maybe when new orleans gets back to normal ;) |
| 17:54.35 | robin_sz | 2027? |
| 17:54.40 | phifli | lol |
| 17:54.43 | phifli | definitely not |
| 17:54.46 | phifli | bellsouth has a lot of plans |
| 17:54.50 | phifli | they were in the process of fiber to homes |
| 17:54.53 | phifli | of all places... |
| 17:54.59 | robin_sz | fiber? |
| 17:55.05 | phifli | but ive lived there almost my whole life ... (subtract 1 year for california) |
| 17:55.05 | phifli | yes |
| 17:55.05 | robin_sz | is that like fibre? |
| 17:55.10 | phifli | you wouldn't get full speed of course |
| 17:55.20 | phifli | lol |
| 17:55.24 | phifli | nah gigabit :) |
| 17:55.28 | phifli | to neighborhoods |
| 17:55.33 | phifli | bellsouth is doing major upgrades |
| 17:55.34 | robin_sz | I guess its measured in meters? |
| 17:55.42 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 17:55.43 | phifli | i just hope i can have access to a NOC with a backbone to them |
| 17:55.46 | robin_sz | instead of metres too. |
| 17:55.51 | phifli | lol |
| 17:56.07 | robin_sz | ooh, then avoif worldcom. |
| 17:56.36 | robin_sz | they am spineless |
| 17:56.58 | ctooley | When I start a call from a call file and then bridge it to another extension via Dial, I lose the numbers in the CDR. I was wondering if I could start the call via the Manager and not have this side-effect? |
| 17:57.20 | phifli | lose the numbers in CDR? |
| 17:57.26 | ctooley | src and dst are empty |
| 17:57.30 | phifli | what type of call file? |
| 17:57.40 | phifli | IAX or SIP? |
| 17:58.06 | ctooley | IAX, with a standard call file dropped into the outgoing spool directory |
| 17:58.43 | phifli | hmm |
| 17:58.47 | phifli | i noticed that in 1.2.0... |
| 17:58.51 | ctooley | it's a problem I've known and accepted for a long time. the Dst is set to (s) which is the extension in the context that the call is routed to after answering. |
| 17:58.52 | phifli | they do a lot more with the ANI |
| 17:58.59 | phifli | therefore caller id might be based more upon that |
| 17:59.13 | phifli | whereas in 1.0.9 .... it didn't seem to play with the ANI too much |
| 17:59.22 | phifli | considered (hidden caller id) in asterisk source |
| 17:59.28 | phifli | so.. that might be your issue? |
| 17:59.38 | ctooley | don't think so |
| 17:59.38 | phifli | or is it the CDR itself? |
| 17:59.45 | phifli | or was I on the money with caller id? |
| 17:59.52 | phifli | hmm |
| 17:59.54 | ctooley | it's not really a caller id issue. it's a CDR issue |
| 17:59.57 | phifli | your using SetCDRUserField? |
| 18:00.00 | phifli | ForkCDR? |
| 18:00.09 | phifli | ie: how are you implementing it? |
| 18:00.13 | ctooley | I'm using SetCDRUserField |
| 18:00.25 | ctooley | but that just sets the "userfield" value |
| 18:00.45 | ctooley | I'm missing the src and dst field values, unlike an inbound call that gets routed to an extension. |
| 18:01.15 | phifli | Yeah |
| 18:01.16 | phifli | I understand |
| 18:01.19 | phifli | do you set them? |
| 18:01.24 | ctooley | no |
| 18:01.25 | phifli | and then it doesnt copy over when you call? |
| 18:01.40 | phifli | dst field im not sure |
| 18:01.45 | phifli | src... i know that in 1.2.0... |
| 18:01.50 | phifli | to make it follow through to the next channel |
| 18:01.55 | phifli | you need _ infront of variables |
| 18:01.57 | phifli | __ is global |
| 18:02.06 | phifli | and Set whereas before we used SetVar |
| 18:02.30 | phifli | check out www.voip-info.org/wiki-Asterisk+Set |
| 18:02.35 | ctooley | Yeah, but these aren't "variables" they're magic values that Asterisk determines from the call |
| 18:03.18 | phifli | i see |
| 18:03.25 | phifli | honestly outgoing spool directory.. |
| 18:03.26 | phifli | hows that work? |
| 18:03.31 | ctooley | The are the callerid of the call and the destination number that was dialed for the most part. |
| 18:03.36 | phifli | im not familiar but ill go check out and test real quick possibly |
| 18:03.53 | ctooley | you write a file in the proper format and copy the file into the directory |
| 18:04.05 | phifli | I see |
| 18:04.09 | phifli | and it adds to your queue? |
| 18:04.31 | phifli | if you could msg me a sample simple easy as can be config ill debug it for you |
| 18:04.36 | ctooley | Asterisk is constantly polling the spool, finds the file, dials the number, and when someone answers directs the call to the context/extension/priority that you define in the call file |
| 18:04.36 | phifli | best i can do |
| 18:04.57 | phifli | i have a few extensions and chans i can test on real quick |
| 18:05.06 | ctooley | nah, it's not that big of a deal, I've lived with writing my own CDRS for a long time. |
| 18:05.15 | phifli | hehe |
| 18:05.18 | Delvar | does asterisk support t38 passthough? |
| 18:05.19 | ctooley | thanks though |
| 18:05.20 | phifli | yeah im about to have to do my own but |
| 18:05.35 | phifli | wouldnt you like to let ast devels know of the bug? |
| 18:05.44 | phifli | i'd like to just see what you rtalking about for that purp[ose |
| 18:05.55 | ctooley | :) They know of it, it's not a "bug" it's a difficult problem to solve |
| 18:06.28 | ctooley | check out the wiki on how to create a call file and copy one into /var/spool/asterisk/outgoing, then watch the console, you'll see. |
| 18:06.46 | *** join/#asterisk Berkey (n=berkey@nv-208-13-143-215.sta.sprint-hsd.net) |
| 18:06.58 | phifli | ok |
| 18:07.05 | phifli | back to my prob :( |
| 18:07.13 | phifli | frame out of order, so its dropping my call |
| 18:07.20 | phifli | any ideas anyone? i have jitterbufer=yes |
| 18:09.12 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost, -- if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
| 18:09.55 | Berkey | If you are using POTS don't forget to have your telco set cutoff on disconect otherwise called diconnect supervision on their switch. Very important to keep * from tying up all your lines with dead calls! |
| 18:09.58 | phifli | bbz: do you have a link? |
| 18:10.13 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 18:10.21 | Berkey | Learned that one the hard way : ) |
| 18:10.30 | phifli | berkey: definitely will keep it in mind, caused you some grief? |
| 18:10.38 | phifli | figured =/ |
| 18:10.42 | phifli | glad you got it sorted out |
| 18:12.47 | Berkey | Yep when I set up some transfer to external numbers IE. call forwarding for the weekend all my lines got tied up. |
| 18:15.39 | *** join/#asterisk YoMama (n=r00t@pcp02689850pcs.roylok01.mi.comcast.net) |
| 18:16.15 | phifli | ah |
| 18:16.26 | phifli | that sucks to be losing calls |
| 18:16.39 | YoMama | son of a beyotch...ok..i need a dialplan expert |
| 18:17.08 | file | experts are expensive. |
| 18:17.25 | YoMama | file: ok..not an expert..i just n eed some freakin' help |
| 18:17.31 | file | then ask a question |
| 18:17.52 | phifli | oh wow it seems every bunch of packets im getting one out of order and 135ms lag |
| 18:17.59 | phifli | but it sounds fine on phone |
| 18:18.01 | phifli | any ideas anyone |
| 18:19.07 | YoMama | I can make outbound calls on my SIP proxy fine..but when i try to bridge a call from my Zap through to the SIP proxy...it gives me 400 Bad Request |
| 18:19.44 | *** join/#asterisk draven (n=draven@user-0cdvmgb.cable.mindspring.com) |
| 18:20.33 | draven | is there a comprehensive list of modems/isdn cards that support asterisk, i'm curious for a small home network basically as a test system |
| 18:22.17 | Nivex | draven: might start here: http://www.asterisk.org/hardware |
| 18:22.29 | Nivex | draven: failing that I think there might be a list on the wiki (voip-info.org) |
| 18:22.34 | YoMama | no one any clues eh? |
| 18:22.45 | draven | thanks Nivex |
| 18:23.47 | marcus2 | should i have any issues using a carrier access ABII channel bank with *? |
| 18:24.15 | blop | hi, i'm looking on information on how to use QSIG protocol with asterisk, i'd like to link * to a panasonic pabx. does it need only a BRI link between them or more ? |
| 18:25.11 | YoMama | bizarre |
| 18:26.21 | YoMama | wait a sec.. |
| 18:26.29 | draven | would a v.92 hardware modem be supported? or am i just an idiot? |
| 18:26.49 | Juggie | it could be |
| 18:26.51 | Juggie | but its not |
| 18:27.22 | [TK]D-Fender | draven : onlt a very specific chipset of modem is compatible. Other than that be prepare to make a purchase... |
| 18:27.25 | *** join/#asterisk loick (n=loic@APuteaux-151-1-22-105.w82-124.abo.wanadoo.fr) |
| 18:27.51 | draven | so thats why the intel software one is supported but not a true hardware v.92 modem |
| 18:27.56 | Juggie | ahh i kinda musunderstood his question, i read it as would * be able to become a modem |
| 18:27.58 | pifiu | ever since I upgraded to asterisk 1.2 my call quality went down the drain |
| 18:28.00 | pifiu | argh wtf |
| 18:28.27 | draven | thans for the info |
| 18:29.03 | Nivex | pifiu: Interesting. I've noticed no change in my call quality since I upgraded. |
| 18:29.25 | YoMama | is there any possibility that when I do Dial() from an incoming Zap channel to a SIP proxy, that it's sending the SIP proxy a different message than if i was just dialing outbound? |
| 18:30.30 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 18:30.55 | [TK]D-Fender | he should have to scroll down in the "phone" menu... |
| 18:30.57 | phifli | hmm |
| 18:31.08 | [TK]D-Fender | maybe he's not bright enough to have tried on his own :) |
| 18:31.15 | [TK]D-Fender | oops, wrong chan :) |
| 18:31.25 | file | someone took over my testing number |
| 18:31.27 | phifli | file: any idea with out of order packets? ;) |
| 18:31.49 | synthetiq | how can i get a voicemail() to write to two vmail boxes? |
| 18:32.06 | phifli | i think whats going on is... |
| 18:32.07 | file | phifli: if I knew I would have answered the other 5 times you said it |
| 18:32.13 | marcus2 | does anyone happen to know if CAC ABII channel banks have an integrated csu? |
| 18:32.19 | phifli | its fixing the Sequence or window and |
| 18:32.29 | phifli | then its ACKS cuz theres nothing on the line for a moment |
| 18:32.30 | IronHelix | syn- voicemail(box1@context&box2@context) |
| 18:32.36 | phifli | therefore when it sees the ACK the same as the last one |
| 18:32.41 | phifli | it wants to kill it |
| 18:32.42 | synthetiq | i did that but it didnt work |
| 18:32.43 | phifli | i just read source |
| 18:32.43 | IronHelix | it will play the greeting for box1 (or whatever the first box is), then copy the message once its left |
| 18:32.48 | phifli | file: sorry ;) |
| 18:32.55 | pifiu | yeah it seems the codecs changed to gsm instead of g711 |
| 18:32.55 | phifli | so i might just try +1 this thing in source |
| 18:32.59 | phifli | and see how it goes |
| 18:33.41 | synthetiq | all it does is write the voicemail to the first box |
| 18:34.23 | YoMama | IronHelix: wanna help me out with my problem? :) |
| 18:35.03 | IronHelix | i only have like 2 misn but sure |
| 18:35.28 | YoMama | Iron: thx man..i got a sip proxy set up...i can dial from a sip phone thru the sip proxy no problem |
| 18:35.47 | synthetiq | ahh my fault i put the u in front of each box? |
| 18:35.51 | YoMama | however..when i call in on my Zap..and try to bridge a call using the sip proxy..i get a 400 Bad Request |
| 18:35.52 | synthetiq | now it works |
| 18:36.23 | YoMama | Iron: I'm thinking that the sip proxy is getting a different command when it's trying to bridge a Zap than when I'm just doing it straight sip |
| 18:36.38 | YoMama | maybe it's trying a transfer command and my sip proxy provider's not allowing it? |
| 18:36.39 | IronHelix | using * as sip proxy? |
| 18:36.42 | YoMama | yes |
| 18:37.04 | IronHelix | ok then its just all going into *, where does the 400 bad request come from? |
| 18:37.19 | YoMama | from my sip provider |
| 18:37.37 | YoMama | SIP Phone -> * -> SIP Proxy = works |
| 18:37.38 | IronHelix | oh ok, so SIP-> itsp works, but zap-> itsp doesnt? |
| 18:37.49 | YoMama | Incoming Zap -> * -> SIP Proxy = no workie |
| 18:38.13 | IronHelix | that is VERY strange, asterisk should dial() the channel the same way no matter what you do |
| 18:38.35 | YoMama | you'd think so... |
| 18:38.39 | IronHelix | IMHO, assuming asterisk isnt broken, the problem lies in extensions.conf. something has to be different in the two entries that dial the sip channel |
| 18:38.42 | IronHelix | pastebin them |
| 18:38.54 | YoMama | k |
| 18:39.01 | IronHelix | along with the sip.conf entry for your provider |
| 18:41.05 | *** join/#asterisk saftsack (n=oliver@p54A7F1D0.dip.t-dialin.net) |
| 18:41.09 | saftsack | hi |
| 18:41.24 | robin_sz | nah just paste them in the channel. its more fun that way :) |
| 18:41.58 | IronHelix | lol |
| 18:43.08 | *** join/#asterisk DrPickles (n=strangeb@sitcomfurniture.net) |
| 18:43.14 | YoMama | Iron: http://pastebin.com/436416 |
| 18:43.16 | saftsack | exten => _[1-9], |
| 18:43.22 | saftsack | i forgot what it does :( |
| 18:43.31 | IronHelix | that will match extensions 1 thru 9 |
| 18:43.34 | IronHelix | but nothing else |
| 18:43.42 | saftsack | exten => _[1-9],1,Dial(CAPI/@20:${EXTEN}) |
| 18:43.45 | saftsack | thats the full line |
| 18:44.10 | IronHelix | if you dial any one digit, it will create a capi channel to that digit |
| 18:44.11 | IronHelix | but not 0 |
| 18:44.21 | IronHelix | yomama ok what about extnesions.conf |
| 18:44.48 | DrPickles | i am running asterisk@home and i purchased a 400b card with 2 FXO is there anyone that can help me out? |
| 18:44.49 | YoMama | looks below |
| 18:44.55 | YoMama | it's in there |
| 18:45.06 | saftsack | IronHelix, ok but i think i need 0 because i have to call 0 + telephone number because i am behind a tk |
| 18:45.29 | IronHelix | ah |
| 18:46.01 | IronHelix | yomama- try dialing it like this- Dial(SIP/vbuzzer/17343954542,,r) |
| 18:46.08 | IronHelix | eh |
| 18:46.15 | *** join/#asterisk _T3_ (n=rposada@200.63.231.210) |
| 18:46.15 | IronHelix | make that vbuzzer.com |
| 18:46.31 | YoMama | hmm..but it works just like that further below |
| 18:46.34 | IronHelix | if you dial like chantype/channel/exten it will match the 'channel' to the [channel entry] in the .conf file |
| 18:46.55 | _T3_ | anybody know how to setup codecs order |
| 18:46.57 | IronHelix | also- if you do it using the chan type, it will use the fromuser= correctly |
| 18:47.03 | IronHelix | that could be why you get 400 bad request |
| 18:47.09 | *** join/#asterisk Nukemizer (n=Nuke@67.137.28.165) |
| 18:47.16 | IronHelix | if you dial it like you are now, it wont use fromuser so your callerid will be whatever there is on the channel |
| 18:47.22 | saftsack | IronHelix, howto configure extensions.conf for my needs? i wanna dial 0 on my telephone and then the number |
| 18:47.25 | YoMama | Iron: hmm...lemme try that.. |
| 18:47.50 | IronHelix | for a sip call taht might be an extension number, which the switch will write off as useless, whereas if you come in on pstn it will be a real caller id whihc the switch might take the wrong way |
| 18:48.11 | saftsack | IronHelix, is this for me? |
| 18:48.17 | _T3_ | i make international calls using g729 passtrough in asterisk, my sipphone use g729, but when i want call to zap channels the call hangs up |
| 18:48.26 | IronHelix | softsack- exten => _0.,1,Dial(CAPI/@20:${EXTEN}) will take any extension that starts with 0 and send the whole thing including the 0 to your capi channel |
| 18:48.30 | *** part/#asterisk loick (n=loic@APuteaux-151-1-22-105.w82-124.abo.wanadoo.fr) |
| 18:48.42 | IronHelix | T3 thats because a zap channel by nature cannot support 729 passthru |
| 18:48.53 | IronHelix | you have to decode the codec and put it out as analog audio or tdm audio |
| 18:48.56 | *** join/#asterisk loick (n=loic@APuteaux-151-1-22-105.w82-124.abo.wanadoo.fr) |
| 18:49.13 | DrPickles | is there a que? |
| 18:49.15 | IronHelix | try ilbc, its uses a bit more CPU than 729 and a bit more bandwidth but its almost as good |
| 18:49.30 | _T3_ | i know IronHelix, but how i setup to use one codec for international call and ulaw for zap channels from the same sipphone |
| 18:49.35 | IronHelix | no theres just me and a bunch of ppl piling on asking questions |
| 18:49.51 | DrPickles | ah well let me know how i can ask without being a butthead |
| 18:49.57 | IronHelix | for your sip phone, do allow=g729, then allow=ulaw? |
| 18:50.04 | IronHelix | it should fall thru if it cant use 729 |
| 18:50.09 | YoMama | Iron: dammit...same thing |
| 18:50.24 | IronHelix | otherwise you can buy a 729 license from digium, its only like $10 (assuming you only have one channel) |
| 18:50.32 | _T3_ | hmmm |
| 18:50.39 | _T3_ | i have 6 |
| 18:50.41 | fourcheeze | is there any maximum theoretical/practical limit to the size of extensions.conf or the number of contexts? |
| 18:50.51 | fourcheeze | does it get slower with more contexts? |
| 18:50.52 | IronHelix | yeah how much memory you have... |
| 18:51.00 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 18:51.15 | fourcheeze | I think I'm unlikely to exceed my memory |
| 18:51.23 | IronHelix | yeah dont worry too much about it |
| 18:51.29 | IronHelix | if you are worried you can always use realtime |
| 18:51.36 | fourcheeze | IronHelix: the irony |
| 18:51.38 | _T3_ | i have in may sip.conf allow=g729 allow=ulaw but the sipphone doesnt fall thru |
| 18:51.38 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 18:51.45 | saftsack | <PROTECTED> |
| 18:51.53 | saftsack | what does it do? |
| 18:51.55 | fourcheeze | IronHelix: people have been spending all day warning me off realtime |
| 18:52.04 | YoMama | bizarre |
| 18:52.10 | fourcheeze | IronHelix: do you use realtime? |
| 18:52.17 | YoMama | Iron: nope :( |
| 18:52.24 | _T3_ | i do |
| 18:52.45 | IronHelix | fourcheese- nah, i dont have enough stuff to need realtime |
| 18:53.00 | _T3_ | work fine, but my computer resources are always low |
| 18:53.01 | fourcheeze | IronHelix: but you would consider it? |
| 18:53.16 | fourcheeze | IronHelix: I may potentially need to scale things up |
| 18:53.17 | IronHelix | well how many contexts do you have? |
| 18:53.26 | draven | whats the recommended card for a hobby user of * basically 1/2 phones on a linux os |
| 18:53.44 | IronHelix | T3- if the phone doesnt fall thru, is it configured to fall thru to ulaw or to require 729? |
| 18:53.46 | YoMama | Iron: i even changed my LD dialplan to match it the way you had it and it worked just fine |
| 18:53.54 | IronHelix | draven- digium tdm400 series |
| 18:53.56 | fourcheeze | IronHelix: I would need more contexts if I didn't use realtime |
| 18:53.58 | IronHelix | hmmm that is strange |
| 18:54.01 | fourcheeze | essentially one per user |
| 18:54.13 | YoMama | Iron: so...i'm thinking there's some keyword i'm missing where it's trying to send a transfer request instead of a regular dial request to my sip provider |
| 18:54.26 | draven | IronHelix, thanks for the info |
| 18:54.31 | IronHelix | that is less likely, it does the same thing |
| 18:54.36 | IronHelix | draven try voipsupply.com |
| 18:54.41 | IronHelix | a bunch of ppl here use them |
| 18:55.14 | fourcheeze | what I really want is to be able to use my own schema for realtime |
| 18:55.16 | draven | IronHelix, thats the first site google sent me to :) |
| 18:55.19 | fourcheeze | but I don't seem to be able to |
| 18:55.38 | fourcheeze | I mean, what's the point of having DB storage if you can't use your own schema and make it interoperate with other software |
| 18:55.40 | fourcheeze | ? |
| 18:55.44 | YoMama | Iron: could it be a context issue? |
| 18:55.54 | IronHelix | yomama- that is very strange. perhaps if you are calling out to the pstn using vbuzzer, it is denying your request because you already have a channel open? |
| 18:56.11 | _T3_ | may send the codec setup on the dial command???? |
| 18:56.15 | YoMama | no...i'm dialing incoming over the Zap channel |
| 18:56.22 | YoMama | and trying to bridge a call using the SIP proxy |
| 18:56.26 | IronHelix | is the zap chan a fxs port? |
| 18:56.27 | _T3_ | my sipphone is an pa1688 based |
| 18:56.29 | fourcheeze | unless I missed something |
| 18:56.35 | _T3_ | nop fxo |
| 18:56.40 | YoMama | Iron: no..FXO |
| 18:56.45 | IronHelix | so its connected to a line |
| 18:56.46 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 18:56.49 | YoMama | Iron: yes |
| 18:56.52 | IronHelix | gah i am talking to like 3 people at once |
| 18:56.53 | IronHelix | lol |
| 18:56.58 | IronHelix | so how are you calling intot he line? |
| 18:57.03 | YoMama | Iron: over a cell phone |
| 18:57.04 | IronHelix | are you using a cell phone? |
| 18:57.06 | IronHelix | ah |
| 18:57.07 | IronHelix | hmmm |
| 18:57.46 | YoMama | Iron: yup..calling using one cell..then dialing 501..then it tries to use the SIP proxy peer and it comes back with that bad request |
| 18:57.53 | YoMama | i have two cell phones |
| 18:58.22 | YoMama | i'm calling in on one cell into my landline and simply trying to get asterisk to use my SIP proxy to call me out on my other cell |
| 18:58.28 | fourcheeze | does anyone know of a cell phone that's also wifi/sip ? |
| 18:58.41 | tzanger | Treo650? |
| 18:58.51 | YoMama | tzanger: no..the 650 doesn't have wifi |
| 18:58.58 | IronHelix | fourcheeze- you will be using a pda for that |
| 18:59.05 | YoMama | some of the WinCE phones do..and then u could buy xlite for wince and boom..there ya go |
| 18:59.06 | tzanger | YoMama: it doesn't have SDIO? |
| 18:59.07 | IronHelix | probably something windows mobile based |
| 18:59.13 | YoMama | tzanger: no..it does |
| 18:59.13 | fourcheeze | ok, is there any non-windows pda I can look at? |
| 18:59.20 | YoMama | even my 600 has SDIO |
| 18:59.23 | IronHelix | nokia is releasing something that might work |
| 18:59.28 | DrDeke | fourcheeze: The only phones like that that I know of are pda-type phones. |
| 18:59.31 | tzanger | YoMama: well it then "can" have wifi :-) |
| 18:59.36 | IronHelix | yomama- pastebin the debug log |
| 18:59.37 | YoMama | ha |
| 18:59.39 | tzanger | I don't know of any Palm SIP phones though |
| 18:59.48 | YoMama | is there a sip client for Palm? |
| 18:59.59 | YoMama | that'd be the only trick |
| 19:00.08 | fourcheeze | because if I can find something that does wifi + sip and isn't windows and can also be a cell phone, I think I might be able to persuade my boss to get one |
| 19:00.33 | fourcheeze | I know I can do the wifi/sip thing with my Zaurus |
| 19:00.38 | fourcheeze | but that doesn't have a phone |
| 19:01.05 | fourcheeze | gsm/g3/gprs ir whatever |
| 19:01.12 | fourcheeze | s/g3/3g |
| 19:02.40 | YoMama | so far...all of the SDIO cards i've seen only have dirvers for WinCE |
| 19:02.50 | mocker | What is the default key for configuring a voicemail box? |
| 19:03.24 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 19:03.37 | mocker | Or how can users record new voicemail greetings? |
| 19:04.11 | YoMama | mocker: have them call into it and follow the voice prompts? |
| 19:05.15 | fourcheeze | apparently there's a sip client for http://europe.nokia.com/nokia/0,,75023,00.html |
| 19:06.39 | trixter | VoIPGrows in Asia $1B Yearly http://www.redherring.com/Article.aspx?a=14585&hed=VoIP+Grows+in+Asia+%241B+Yearly§or=Industries&subsector=Communications |
| 19:06.44 | fourcheeze | YoMama: http://linuxdevices.com/news/NS3384158651.html |
| 19:09.49 | YoMama | what's SER stand for? |
| 19:10.04 | fourcheeze | ~ser |
| 19:10.05 | jbot | extra, extra, read all about it, ser is Sip Express Router - see http://www.iptel.org/ser/ |
| 19:10.21 | saftsack | exten => 20,1,DISA,no-password|sipgateohneneun |
| 19:10.24 | saftsack | what does this line do? |
| 19:10.30 | YoMama | got it |
| 19:10.41 | *** join/#asterisk loick (n=loic@APuteaux-151-1-22-105.w82-124.abo.wanadoo.fr) |
| 19:10.47 | YoMama | saftsack: oh man..it lets outside people connect to your * server and dial outbound |
| 19:11.13 | saftsack | yes theoretical |
| 19:11.16 | saftsack | but it doesnt work |
| 19:11.25 | saftsack | i thought the same as you think |
| 19:11.35 | saftsack | if youre german can we query? |
| 19:11.46 | trixter | disa gives them a 'dialtone' as if they are local to the server, they can call anything in the context they are in. |
| 19:12.05 | saftsack | ok .. |
| 19:12.22 | saftsack | how can i see, if this link works? |
| 19:12.26 | saftsack | <PROTECTED> |
| 19:12.26 | saftsack | <PROTECTED> |
| 19:12.49 | saftsack | that are the last lines if i call the isdn here |
| 19:13.07 | jimmy_deanPB | is there something that I need to do before utilizing the special 's' extension in extensions.conf? I'm trying to use it...when I specify a real extension number like 200 and dial 200 from another phone, it works, but I can't seem to make it generic with 's' in place |
| 19:13.28 | YoMama | jimmy: no |
| 19:14.00 | jimmy_deanPB | YoMama, any idea why the s extension wouldn't work for me? |
| 19:14.14 | IronHelix | are you in the right context? |
| 19:14.33 | twisted[asteria] | s is the dialplan equivalent of 'no extension' |
| 19:14.35 | jimmy_deanPB | IronHelix, yes, like I said...if I replace it with '200' instead of 's', it works |
| 19:14.55 | YoMama | Iron: AAAUGH |
| 19:15.33 | twisted[asteria] | the s extension is NOT a matchall extension |
| 19:15.48 | Katty | is it a hug extension? |
| 19:15.50 | jimmy_deanPB | twisted[asteria], so what is it then and what's the proper use for it? |
| 19:15.56 | twisted[asteria] | ~docs |
| 19:15.57 | jbot | from memory, 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 |
| 19:16.08 | YoMama | Iron: i just don't get it..when i use my sip phone or x-lite...it uses the sip proxy with zero problems |
| 19:16.24 | jimmy_deanPB | twisted[asteria], I've read over tons of docs...I still don't get it |
| 19:16.35 | Katty | jimmy_deanPB: i still don't get it. |
| 19:17.12 | Katty | twisted[asteria]: i got a shiny new shirt today! |
| 19:17.17 | Katty | twisted[asteria]: it has nextel and nascar on it |
| 19:17.19 | fourcheeze | has anyone managed to run * on a linksys router yet? |
| 19:17.37 | yxa | jimmy_deanPB what are you trying to achieve? |
| 19:17.38 | IronHelix | yomama- pastebin the logs |
| 19:17.59 | IronHelix | fourcheeze- yeah its been done on a WRT with openwrt firmware. no transcoding but it switches fine |
| 19:18.13 | *** join/#asterisk Berkey (n=berkey@nv-208-13-143-215.sta.sprint-hsd.net) |
| 19:18.23 | fourcheeze | wow |
| 19:18.28 | IronHelix | jimmydean- try this, exten => 200,1,Goto(s,1)? |
| 19:18.31 | fourcheeze | might have to look at that |
| 19:18.36 | *** join/#asterisk jcazor (n=jcazor@62-14-210-92.inversas.jazztel.es) |
| 19:18.41 | jimmy_deanPB | yxa, just a simple context to accept all incoming calls from an external SIP connection, present a voice prompt, and allow users to dial an internal extension |
| 19:19.01 | IronHelix | then use exten => _X. |
| 19:19.08 | IronHelix | it will match any extension of one or more digits |
| 19:19.30 | jimmy_deanPB | IronHelix, ooh, hmm...that's simple yet it really wasn't presented clearly in any of the docs I read |
| 19:19.33 | jimmy_deanPB | that makes sense |
| 19:19.52 | Katty | twisted[asteria]: mew? |
| 19:19.58 | Berkey | On Polycom 501s why wouldn't the other two lines ring if one line is in use but the other lines are set as members of the same q? |
| 19:20.20 | Katty | twisted[asteria]: you didn't answer beepbeep :< |
| 19:20.20 | jimmy_deanPB | IronHelix, explain to me one other thing...the context in [general] in sip.conf...is that the context that all calls outside of the internal extensions come into? |
| 19:20.35 | Berkey | You know so you could place one call on hold then answer another caller? |
| 19:20.47 | jimmy_deanPB | IronHelix, and then the context that is defined, per each internal extension definition, is the extension for that extension |
| 19:20.59 | IronHelix | [general] isnt a context and you shouldnt be using it as one |
| 19:21.13 | jimmy_deanPB | IronHelix, no sorry, I didn't mean to imply is was a context |
| 19:21.20 | jimmy_deanPB | IronHelix, config section |
| 19:21.24 | YoMama | Iron: ok...here's a twist for ya..i put in a DISA line..and it worked |
| 19:21.44 | IronHelix | yomama- pastebin logs of it not working |
| 19:21.47 | IronHelix | that is odd tho |
| 19:22.02 | IronHelix | jimmy- a call will come into whatever context it is set to |
| 19:22.13 | IronHelix | you define that in the channels config section (sip.conf iax.conf zapata.conf) |
| 19:22.24 | jimmy_deanPB | IronHelix, right, ok...yes |
| 19:22.56 | jimmy_deanPB | IronHelix, how do I then setup any extension to be able to call out via SIP via my VOIP provider? |
| 19:23.04 | jimmy_deanPB | IronHelix, that's the only other thing unclear to me at this time |
| 19:23.12 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 19:23.14 | IronHelix | so if you have 5 sip phones, you can in sip.conf put context=myphones and they can then dial any extensions in the [myphones] context |
| 19:23.20 | YoMama | bah...is pastebin down? |
| 19:23.29 | IronHelix | you then make a context that can dial your voip provider |
| 19:23.40 | IronHelix | and use include => that in myphones |
| 19:23.46 | jimmy_deanPB | IronHelix, I put that in extensions.conf? |
| 19:24.33 | jimmy_deanPB | IronHelix, is this new outbound context the place to put the 'register' line and will I need a "Dial" command for that? |
| 19:24.39 | YoMama | Iron: i think pastebin just went down |
| 19:24.47 | YoMama | or i'm having routing problems to it |
| 19:24.53 | IronHelix | jimmy- yeah this is all extensions.conf |
| 19:24.56 | IronHelix | no |
| 19:25.03 | IronHelix | register => is a channel thing |
| 19:25.06 | IronHelix | that goes in sip.conf or iax.conf |
| 19:25.15 | jimmy_deanPB | ahh ok, in the general section? |
| 19:25.26 | IronHelix | no pastebin works |
| 19:25.35 | IronHelix | see you have to separate the concept of extensions and channels |
| 19:25.51 | jimmy_deanPB | that is a confusing concept for a new person to * |
| 19:25.52 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 19:25.57 | IronHelix | in sip.conf, you put a register=> line in the general section, and then below that you define [channels] |
| 19:26.09 | YoMama | hmm |
| 19:26.15 | IronHelix | each one of those [configs] is a channel which you can dial |
| 19:26.51 | IronHelix | in each [config] there should be context= or it will use the context= thats in [general] |
| 19:27.21 | jimmy_deanPB | IronHelix, ahh, hmm...so there's literally a [channel] config section that I should have in sip.conf? |
| 19:27.22 | IronHelix | when a call comes in on that channel, it checks in extensions.conf to see if theres an extension matching the number dialed in the context that that channel is in and any included contexts |
| 19:27.33 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 19:27.35 | IronHelix | no by [channel] i mean that it defines a channel |
| 19:27.41 | jimmy_deanPB | ahh, ok |
| 19:27.50 | IronHelix | like [voxee] [asterlink] [myprovider] [mysipphone] etc |
| 19:27.57 | jimmy_deanPB | right, ok |
| 19:27.58 | IronHelix | a channel can be to a sip phone or a provider |
| 19:28.21 | IronHelix | an EXTENsion is nothing more than a target that can be dialed and what will happen when you dial it |
| 19:28.24 | jimmy_deanPB | so I would have [my-voip-provider] register => ... |
| 19:28.39 | IronHelix | so you'd do something like this |
| 19:28.45 | YoMama | the register goes in the [general] context |
| 19:28.59 | IronHelix | in sip.conf, put [my-voip-provider] then below that put all its info (host username secret type context etc) |
| 19:29.14 | *** join/#asterisk DrPickles (n=strangeb@sitcomfurniture.net) |
| 19:29.32 | YoMama | Iron: man..my world is coming to an end..u say pastebin works, but i'm getting connection refused |
| 19:29.43 | YoMama | WAAAH...why doesn't anything freaki'n work? |
| 19:29.46 | YoMama | i'm having one of those days |
| 19:29.51 | IronHelix | then in [general] put register => username:secret@my-voip-provider/optionalextension |
| 19:30.27 | IronHelix | THEN when a call comes in, the call goes to optionalexten in the context provided in the [my-voip-provider] block |
| 19:30.43 | IronHelix | or if you leave optionalextension off, it will go to whatever the dude dialed |
| 19:30.53 | jimmy_deanPB | interesting...there doesn't seem to be any good docs out there outlining this in a straightforward manner |
| 19:30.54 | IronHelix | so the call comes in and assume everything works ok and it matches an extension |
| 19:30.58 | *** join/#asterisk litb (n=litb@p54B19018.dip0.t-ipconnect.de) |
| 19:31.04 | marcus2 | where are updated voicemail passwords stored? |
| 19:31.26 | IronHelix | it will then execute that extension by priority |
| 19:31.41 | IronHelix | one of those priorities can be Dial() which will dial something else |
| 19:32.08 | IronHelix | so say you have two sip channels defined, [my-voip-provider] and [my-sip-phone] |
| 19:32.15 | *** join/#asterisk lehel (n=lehel@82.79.20.17) |
| 19:32.15 | litb | hi! i have a VOIP number and only want to telphone with my pc rather than with the telphone. what does i need from them: Asterisk Version 1.2.0 |
| 19:32.15 | litb | <PROTECTED> |
| 19:32.15 | litb | <PROTECTED> |
| 19:32.15 | litb | <PROTECTED> |
| 19:32.15 | litb | <PROTECTED> |
| 19:32.26 | IronHelix | you then have two contexts, [dial-out] and [internal] |
| 19:32.35 | IronHelix | litb you just need zaptel and asterisk |
| 19:32.44 | IronHelix | sounds is good too but you dont need addons or libpri |
| 19:32.48 | litb | i use 1und1 |
| 19:33.02 | litb | and have a fritzbox phone wlan 7500 |
| 19:33.16 | phifli | any reason why i keep getting out of packets and the call is being dropped? |
| 19:33.21 | litb | but thats irrelevant, i think |
| 19:33.21 | phifli | IAX2 |
| 19:33.44 | jimmy_deanPB | IronHelix, ok, I think I got it...I'll give it a try some more...thanks |
| 19:34.00 | IronHelix | so you can put an exten somewhere, that will Dial(SIP/my-voip-provider/${EXTEN}) |
| 19:34.13 | IronHelix | ${EXTEN} will be replaced with whatever number you dialled |
| 19:34.26 | IronHelix | if you want to use something like 9 to dial out |
| 19:34.38 | IronHelix | you can do like ${EXTEN:1} which will strip off the first digit |
| 19:34.50 | litb | IronHelix, but i have no zaptel card. just a dsl modem |
| 19:35.00 | fourcheeze | litb: if you just want to make a call you don't need asterisk |
| 19:35.31 | IronHelix | so if you make an extension like exten => _91XXXXXXXXXX,1,Dial(SIP/my-voip-provider/${EXTEN:1}) then you can dial 9 followed by any 11 digit number and just the number will be passed to the provider |
| 19:35.52 | IronHelix | litb- you want zaptel for the timing. edit the makefile and uncomment ztdummy |
| 19:35.57 | IronHelix | more info on that is available at voip-info |
| 19:35.59 | IronHelix | ~wiki |
| 19:36.07 | IronHelix | ~voip-info |
| 19:36.08 | jbot | well, voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 19:37.23 | litb | fourcheeze, i just want to place a coll. what do i need then? |
| 19:37.34 | litb | s/coll/call |
| 19:37.42 | fourcheeze | you want to call from your pc? |
| 19:37.48 | fourcheeze | you just need a softphone |
| 19:38.18 | IronHelix | bbl |
| 19:38.22 | IronHelix | sorry i cant help you all now :( |
| 19:38.31 | fourcheeze | ~softphone |
| 19:38.33 | jbot | something that should be drug out into the street and shot |
| 19:38.43 | IronHelix | hahahahahhaha |
| 19:38.46 | chapeaurouge | lol |
| 19:38.58 | fourcheeze | well that's one way of looking at it |
| 19:39.10 | *** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m) |
| 19:39.13 | *** join/#asterisk otwin (n=Use@217.31.78.25) |
| 19:39.32 | crash3m | can anyone tell me what the password length limit is for a polycom ip300/500? |
| 19:40.08 | fourcheeze | litb: http://www.voip-info.org/wiki-VOIP+Phones#SoftPhonesvoiceonly |
| 19:40.15 | tzanger | crash3m: shorter than what you're providing if you're having trouble. :-) |
| 19:40.21 | crash3m | heh, gee thanks tzafrir_laptop |
| 19:40.25 | *** join/#asterisk loick (n=loic@APuteaux-151-1-22-105.w82-124.abo.wanadoo.fr) |
| 19:40.25 | crash3m | err tzanger |
| 19:43.25 | litb | thanks, fourcheeze |
| 19:44.34 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 19:50.13 | *** join/#asterisk convey (n=test@66.55.43.2) |
| 19:52.09 | convey | Why woudl I get a failure to authenticate invide to "extension" error for Outbound calls? |
| 19:52.20 | convey | wow my spelling sucks |
| 19:52.44 | convey | let me rephrase |
| 19:53.37 | convey | Why would I get "Failure to authenticate on INVITE to "extension#" error for outbound calls? |
| 19:57.32 | *** join/#asterisk gLAsgowMonkey (n=gmclachl@i-83-67-88-174.freedom2surf.net) |
| 19:57.48 | *** join/#asterisk [chico] (n=user@p5491132A.dip0.t-ipconnect.de) |
| 19:58.32 | *** join/#asterisk pointer (i=pointer@aj.catt.com) |
| 19:58.46 | pointer | anyone have any PoE switch recommendations from personal exp? |
| 19:59.03 | YoMama | not directly..but i hear the SMCs are pretty good |
| 19:59.39 | ender | we use other SMC switches and I'm quite happy with them. No PoE on the ones we use though. |
| 19:59.50 | pointer | YoMama: are you being facetious? |
| 19:59.58 | pointer | wow, ok |
| 20:00.05 | pointer | I haven't used SMC gear in a long time |
| 20:00.10 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
| 20:00.29 | pointer | we're talking about managed switches here, right? |
| 20:01.01 | *** join/#asterisk dos000 (n=dos000@CPE00119572fd49-CM00137186e53a.cpe.net.cable.rogers.com) |
| 20:01.08 | ender | pointer: yes |
| 20:01.08 | pointer | we're mostly a cisco/extreme/force10 network here |
| 20:01.18 | ender | pointer: SMC managed switches are pretty darn nice. |
| 20:01.18 | [TK]D-Fender | pointer : I use D-Link DES-1526's here and they are quite nice and inexpensive. |
| 20:01.19 | paryl | i've set up a queue, and i have agents logging in, but when a call rings through it goes to the voicemail of the agent instead of ringing the next available agent. is there a setting that i can change to fix that? |
| 20:01.28 | ender | pointer: web/ssh interface, highly configurable, lots of features. |
| 20:02.05 | dos000 | when one does "sip show peers" are you supposed to see all IPs of registered sip endpoints? The problem is that i am only able to see 4 of them |
| 20:02.48 | hardwire | blah and 1/2 dookie |
| 20:04.37 | docelmo | does anyone know if the Grandstream BT102 will do auto answer for paging feature? |
| 20:05.41 | convey | Anyone know why I would get a "Failure to authenticate on INVITE to "extension#" error for outbound calls? |
| 20:05.42 | docelmo | like the GXP2000 does? |
| 20:06.01 | YoMama | docelmo: i just bought a GXP2000 |
| 20:06.25 | *** join/#asterisk [chico] (n=user@p5491132A.dip0.t-ipconnect.de) |
| 20:09.07 | docelmo | I own 10 of them and buying 40 more.. But I need something slightly more inexpensive with almost the same featues.. Namely the paging or autoanswer on one of the lines |
| 20:09.46 | trixter | what price are you getting for the gxp2000? |
| 20:09.54 | docelmo | 90 |
| 20:10.05 | *** part/#asterisk pointer (i=pointer@aj.catt.com) |
| 20:10.12 | *** part/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
| 20:10.25 | trixter | I may be able to beat that, email trixter@0xdecafbad.com with details on what you want where it will be shipped etc.. that way I have a record of it and can get back to you more easily :) |
| 20:10.45 | trixter | and does the 90 incl shipping? |
| 20:10.49 | docelmo | yes |
| 20:10.51 | gLAsgowMonkey | folks i need some advice i am a software engineering student on a placement, i have been given a project to set up a 30 user network |
| 20:11.01 | trixter | ok, well I will see I still may be bale to beat that |
| 20:11.05 | gLAsgowMonkey | so i need some hardware advice |
| 20:11.09 | docelmo | But I have another project in PA I need the same type of phone but cheaper over all |
| 20:11.25 | trixter | the problem is the quality starts to degrade :( |
| 20:11.28 | docelmo | I was looking at the BT102 but I need it to have Auto Answer |
| 20:11.30 | gLAsgowMonkey | and some software, i looked at aterisk@home but i don't think it's going to scale |
| 20:11.42 | docelmo | Asterisk @ home blows |
| 20:11.44 | *** join/#asterisk backblue (n=moo@87-196-3-8.net.novis.pt) |
| 20:11.58 | trixter | although the voip IC industry is supposed to grow 5 times over the next few years so maybe stuff will start to be cheaper, but as for right now its not going to be that much cheaper for anything reasonable |
| 20:12.30 | docelmo | What do you think about the BT102 Does it have the Auto Answer feature like the GXP2000? |
| 20:12.39 | trixter | one thing I dont like about the BT102 is the fact that its 10Mb ethernet.. which means that your switch has to compensate -- normally this isnt a problem unless someone is doing a cheap network |
| 20:12.56 | trixter | I dont like the BT100 line, too many people have complained about audio quality |
| 20:13.02 | trixter | I havent used em personally though |
| 20:13.37 | gLAsgowMonkey | docelmo, is there any other options that offers the same as aterisk@home |
| 20:13.56 | docelmo | yes the white book from oriley on asterisk and CVS |
| 20:14.13 | gLAsgowMonkey | docelmo, just bought it today |
| 20:14.25 | *** join/#asterisk saftsack (n=saftsack@p54A7CFBA.dip.t-dialin.net) |
| 20:14.25 | saftsack | hi |
| 20:14.31 | gLAsgowMonkey | although it hasn't arrived yet |
| 20:14.37 | gLAsgowMonkey | how about hardware specs |
| 20:14.43 | saftsack | are there any fxo cards with support for more than one telephone? |
| 20:14.48 | gLAsgowMonkey | we have a 30 channel isdn line |
| 20:15.21 | docelmo | FXO isnt for Telephone.. FXS is |
| 20:15.27 | *** join/#asterisk cnet2 (n=jjohn@adslnat-sanjose-4.ice.co.cr) |
| 20:15.28 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 20:15.30 | docelmo | 30 channel ISDN you mean E1? |
| 20:16.21 | gLAsgowMonkey | docelmo, um don't know it's all new to me, i spoke to the phone engineer who isntalled the system and he was tight lipped because he knows it will take business from him |
| 20:16.30 | gLAsgowMonkey | all he said was a 30 channel ISDN |
| 20:16.48 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:16.53 | juanjoc | Is anybody using chan_oh323 with Asterisk 1.0.x and getting this error: WARNING[2218]: file.c:980 ast_waitstream: Unexpected control subclass '8' |
| 20:20.42 | *** part/#asterisk loick (n=loic@APuteaux-151-1-22-105.w82-124.abo.wanadoo.fr) |
| 20:21.38 | DrDeke | Hey, can anyone here verify that they are using IAX2 trunking with Voicepulse Connect on outboudn calls? |
| 20:21.47 | [chico] | spricht irgendjemand hier deutsch und hätte kurz zeit mir bei einem Problem zu helfen?? |
| 20:21.55 | Katty | DrDeke: could you monitor the port number maybe? |
| 20:22.27 | docelmo | hehe.. I know.. Tis life.. Better quality and better hardware |
| 20:23.09 | DrDeke | Katty: What I mean is that I get no outgoing audio on calls to VoicePulse when I have trunk=yes set, and I am wondering if this is the case for everyone else, because I have seen people mention using trunking with them without having thsi problem. |
| 20:23.11 | mocker | Does anyone know how to make MeetMe prompt for what conference room to join? |
| 20:23.38 | mocker | All I can get is an extension that prompts for a pass. |
| 20:24.01 | mocker | I'd like for it to prompt for both a conference and pass. |
| 20:25.37 | DrDeke | Hey, this 'chico' guy is trying to install Asterisk from a package in Ubuntu and is getting the message from dpkg: "Unterprozess post-installation script gab den Fehlerwert 255 zurĂĽck" |
| 20:25.38 | DrDeke | doh |
| 20:25.52 | DrDeke | "Child process post-installation script returned error code 255." |
| 20:26.01 | DrDeke | Any ideas? I have only ever installed asterisk from source... |
| 20:26.02 | *** join/#asterisk backblue (n=moo@87-196-44-186.net.novis.pt) |
| 20:26.23 | *** join/#asterisk spiekey (n=kvirc@p549D139D.dip0.t-ipconnect.de) |
| 20:26.25 | spiekey | hi |
| 20:26.36 | spiekey | has anyone a link fo rme how i can set up a meeting room with asterisk? |
| 20:26.42 | Corydon-w | DrDeke: ask the guy who compiled the package |
| 20:26.46 | spiekey | i already have a normal asterisk setup running, but now i would like to talk with 3 ppl at once |
| 20:27.04 | DrDeke | yeah |
| 20:27.28 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
| 20:27.56 | tzafrir_laptop | DrDeke, I figure you can get an English error message using: LC_ALL=C LANGUAGES=C whatever command |
| 20:28.42 | tzafrir_laptop | DrDeke, but anyway, that message tells you that there was an error in some script. Could you find the actual error message itself? |
| 20:28.51 | tzafrir_laptop | (Maybe re-run the apt-get command) |
| 20:28.52 | gLAsgowMonkey | DrDeke, i installed it in Ubuntu with no problems via apt-get |
| 20:29.18 | Katty | DrDeke: i see. |
| 20:29.35 | tzafrir_laptop | Also: it would help if you write what specific Ubuntu version you refer to |
| 20:29.51 | DrDeke | Yeah, actually it looks like the problem was simple; it took me a few minutes to decipher the German :) |
| 20:30.08 | DrDeke | tzafrir_laptop: hm, good idea, thanks! |
| 20:33.43 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 20:36.27 | trixter | woo fetching far more data for my countrycode listing.. http://www.0xdecafbad.com/Global-Numbering-Plan.html -- trying to expand that list a bit |
| 20:38.27 | _T3_ | IronHelix are you there? |
| 20:41.33 | saftsack | Wildcard TDM400P Quad-Port analog or ADSI telephones and POTS |
| 20:41.39 | saftsack | this card is good for entry? |
| 20:44.37 | _T3_ | help, anubody knows how to send the codec that want to use in the cmd dial |
| 20:49.16 | *** join/#asterisk Rowter (n=SilverDr@201.135.26.195) |
| 20:50.48 | jlewis | I dont think you can specify a codec in a dial command |
| 20:51.13 | jlewis | the codec would be selected based on the sip or iax peer config |
| 20:51.27 | *** join/#asterisk sigwerk (n=sigwerk@athena.rootednetworks.com) |
| 20:52.13 | _T3_ | i know that |
| 20:52.26 | _T3_ | i have a sip provider with g729 |
| 20:52.59 | _T3_ | my sipphone has 6 different codec to configure |
| 20:53.15 | _T3_ | first option g729 |
| 20:53.19 | _T3_ | second ulaw |
| 20:53.41 | _T3_ | when i call from sip using provider everything ok |
| 20:54.07 | _T3_ | but if i try to call pstn zap channel give an error |
| 20:54.19 | _T3_ | because zap use ulaw |
| 20:54.40 | _T3_ | but my siphone doesnt fall thru the next option |
| 20:54.54 | _T3_ | may i do something to configure from asterisk?? |
| 20:55.53 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost, -- if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
| 20:57.28 | _T3_ | bbz i |
| 20:57.32 | *** join/#asterisk kiwnix (n=egarcia@137.red-82-158-158.user.auna.net) |
| 20:57.41 | _T3_ | want the info |
| 20:57.42 | hypa7ia | bbz: post that on asterisk-biz, maybe? |
| 20:58.03 | jlewis | well...I just upgraded my personal server from cvs-stable to 1.2.0...it was much easier than I expected |
| 20:58.23 | jlewis | haven't had to touch the config...but its pretty basic |
| 20:59.04 | _T3_ | i have running 1.2 2 weeks ago is working well |
| 20:59.23 | wunderkin | damn im having problems getting a budgetone phone registering directly with broadvoice, i had it setup through an asterisk box before |
| 20:59.37 | jlewis | _T3_: are you doing passthrough and you don't have a g729 license on your server, so it can't transcode? |
| 20:59.52 | _T3_ | passthrough |
| 21:00.08 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 21:00.36 | jlewis | I think you're stuck then...if you want to use that sip phone to talk to zap on your local server, you need to set the sip phone to ulaw, or buy a license and let * transcode |
| 21:01.16 | _T3_ | ok, but you know that license is by channels |
| 21:01.26 | _T3_ | or per server?? |
| 21:01.55 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
| 21:05.13 | DrPickles | i have a digium 400 with 2 fxo. i am trying to connect to POTS. i am using asterisk@home |
| 21:05.39 | DrPickles | i can call from sip to sip phone and have AMP installed |
| 21:06.10 | DrPickles | i can not figure out how to tell if the 400B card is recognised aon how to set up outbound calling |
| 21:06.21 | DrPickles | can some on egive me hand |
| 21:06.37 | bweschke | did you load the kernel modules for the card? |
| 21:06.59 | bweschke | it should tell you which physical modules on the card it saw when the sw module loads. you can use 'dmesg' for this |
| 21:07.07 | DrPickles | i issued a command that was told to me in the instructions |
| 21:07.19 | bweschke | what was that command? |
| 21:07.44 | DrPickles | let me reffernce |
| 21:08.27 | jlewis | the license is for a server and sold in blocks of channels...if you're the only user, you need a 1 channel license |
| 21:08.33 | DrPickles | genzaptelconf -s -d |
| 21:09.34 | DrPickles | genzaptelconf -s -d was the command i gave at the root |
| 21:10.29 | bweschke | DrPickles: I'm really not familiar with that command. it's not part of the core Asterisk. You may want to follow up with someone more familiar with A@H. |
| 21:11.29 | L|NUX | hello all |
| 21:11.32 | L|NUX | i have found one thing |
| 21:11.48 | L|NUX | thatmy sip registration failed after few hours |
| 21:11.56 | bweschke | ok - I just looked it up... |
| 21:12.07 | L|NUX | but when i restart then it will register again |
| 21:12.10 | L|NUX | any idea |
| 21:12.16 | bweschke | do a lsmod and pastebin your results from that here |
| 21:12.23 | bweschke | let's see if the right module is loaded |
| 21:12.39 | *** join/#asterisk colloron (n=liran@80.178.101.98.adsl.012.net.il) |
| 21:12.56 | L|NUX | kies |
| 21:13.21 | bweschke | drpickles: did the genzaptelconf generate a zaptel-auto.conf file for you? |
| 21:13.51 | *** join/#asterisk myke420247 (n=myke@69.29.2.130) |
| 21:14.20 | L|NUX | http://pastebin.ca/30380 |
| 21:14.39 | *** join/#asterisk loick (n=loic@APuteaux-151-1-22-105.w82-124.abo.wanadoo.fr) |
| 21:14.40 | myke420247 | anyone using flash operator panel know exactly what should go into the Mailbox entry for a button? the documentation is a bit terse |
| 21:15.15 | bweschke | L|NUX: that was meant for DrPickles |
| 21:16.28 | bweschke | with regard to your sip registration failure... do you get a message about a registration failure or the peer just stops sending you calls? |
| 21:17.10 | chapeaurouge | how do you make the bot learn something? I would like to teach him my link to download my paste2paste.pl script |
| 21:17.24 | L|NUX | bweschke : well when i do sip show registery |
| 21:17.29 | L|NUX | it will show me failed |
| 21:17.33 | L|NUX | request sent |
| 21:17.38 | convey | Anyone know why I would get a "Failure to authenticate on INVITE to "extension#" error for outbound calls? |
| 21:17.39 | L|NUX | but when i restart it will registered |
| 21:17.44 | syzygyBSD | ~help |
| 21:17.49 | L|NUX | but after few hours its expire |
| 21:18.32 | bweschke | you're going to need to enable some more detailed logging in logger.conf and find out what the specific error is at the time the registration is failing to troubleshoot further |
| 21:18.59 | L|NUX | can you help me with thaat |
| 21:19.03 | syzygyBSD | hmm |
| 21:19.05 | syzygyBSD | where can i find x? |
| 21:19.08 | syzygyBSD | nope |
| 21:19.15 | saftsack | what is the difference beetween pri and bri isdn cards? |
| 21:19.50 | bweschke | there's a "full" line in logger.conf that has additional logging options. go ahead and uncomment it.. then, restart asterisk, and do "set verbose 10" and then "set debug 10" and finally "sip debug peer <peername>" from the CLI. |
| 21:19.53 | syzygyBSD | I could be completely off, so take with a grain of salt, but pri = US, bri = europe |
| 21:20.19 | twisted[asteria] | hahahahhaha |
| 21:20.20 | bweschke | then just let things sit until they fail again and then you can dig back into the /var/log/asterisk/full log file to see why the registration failed at the time that it was supposed to re-register |
| 21:20.23 | chapeaurouge | saftsack, pri usually is on T1/E1, 24-30 channels.. bri is smaller |
| 21:20.42 | chapeaurouge | syzygyBSD, true too |
| 21:20.43 | twisted[asteria] | bri is only 3 channels |
| 21:20.47 | L|NUX | full => notice,warning,error,debug,verbose |
| 21:20.58 | bweschke | yes |
| 21:20.59 | bweschke | you've got it |
| 21:21.03 | L|NUX | :) |
| 21:21.11 | saftsack | chapeaurouge, 24 - 30 channels? telephones or s0 channels? |
| 21:21.12 | chapeaurouge | 2B of 64k channels, and 1D of 16kchannels |
| 21:21.22 | chapeaurouge | saftsack, ds0 channels |
| 21:21.51 | saftsack | 24 * 2B channels? |
| 21:22.00 | chapeaurouge | saftsack, 24 in the US (T1), 30 in Europe (E1) |
| 21:22.14 | saftsack | 30 * 2B channels? |
| 21:22.22 | chapeaurouge | saftsack, no. 30 * 1 |
| 21:22.22 | *** join/#asterisk darby_t (i=darby_t@dln70.neoplus.adsl.tpnet.pl) |
| 21:22.27 | chapeaurouge | i _think_ |
| 21:22.56 | saftsack | is it a multiplexer then? |
| 21:23.21 | *** join/#asterisk kiwnix (n=egarcia@137.red-82-158-158.user.auna.net) |
| 21:23.25 | chapeaurouge | what? the t1/e1 ? |
| 21:23.50 | chapeaurouge | if so, i guess so, since it's using TDM. |
| 21:24.01 | saftsack | for example "Wildcard TE110P Single-Span E1 card" |
| 21:24.40 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 21:25.33 | chapeaurouge | well, that card will handle TDM |
| 21:25.53 | chapeaurouge | but this isn't ISDN |
| 21:26.05 | chapeaurouge | which was your original question. |
| 21:26.09 | saftsack | what is TDM? |
| 21:26.15 | chapeaurouge | Time Division Multiplexing |
| 21:26.42 | chapeaurouge | ~TDM |
| 21:26.44 | jbot | hmm... tdm is Time Division Multiplexing. It is a scheme in which numerous signals are combined for transmission on a single communications line or channel. Each signal is broken up into many segments, each having very short duration. |
| 21:26.56 | *** part/#asterisk otwin (n=Use@217.31.78.25) |
| 21:26.58 | saftsack | ok ... |
| 21:27.12 | saftsack | so i can plug 30 telephones on one port of the card? |
| 21:27.23 | chapeaurouge | well... |
| 21:27.37 | chapeaurouge | on that _one_ port.. you'll have to squeeze them really hard :) |
| 21:28.10 | chapeaurouge | but you can get a channel bank (anyone pls correct me if im wrong) |
| 21:28.14 | saftsack | why? there are 30 B channels on this port? |
| 21:28.26 | saftsack | and every channel provides 64k |
| 21:28.38 | chapeaurouge | how do you plug 30 rj11 in 1 ethernet port? |
| 21:28.53 | chapeaurouge | http://www.voipsupply.com/product_info.php?cPath=99_300_305&products_id=1164&osCsid=1943ae3bef030c0f36b35c5f3620a149 |
| 21:28.56 | chapeaurouge | something like that i guess |
| 21:29.17 | chapeaurouge | but it's not what you're looking for in that instance, however. |
| 21:29.24 | *** join/#asterisk simong (n=simong@c-bd13e455.68-0008-74657210.cust.bredbandsbolaget.se) |
| 21:29.30 | saftsack | im looking for a card that provides the following: |
| 21:29.34 | bweschke | cheapearouge: yes. most of the channel banks have a T1/RJ45 input and then an AMP connector that you can run to a 66 block |
| 21:29.56 | chapeaurouge | bweschke, thanks |
| 21:30.06 | saftsack | 30 telephones or more on one pci port |
| 21:30.11 | LostFrog | Dang.. I don't understand it. |
| 21:30.25 | chapeaurouge | saftsack, yes |
| 21:30.37 | LostFrog | A call comes in from my audiocodes box, It get answered, but the incoming call keeps ringing. |
| 21:30.43 | bweschke | saftsack: the only way you can do that now is to an external E1 channel bank as chapeaurouge mentioned |
| 21:30.45 | chapeaurouge | saftsack, but not directly physically connected to it |
| 21:31.17 | bweschke | otherwise, you can do the new Digium TDM2400 series cards and have an AMP connector right on the card and plug up to 24 FXS on a single full size PCI card |
| 21:31.29 | bweschke | it's a cheaper solution than the E1 card and then the channel bank by far |
| 21:31.34 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 21:31.35 | saftsack | chapeaurouge, ok |
| 21:32.00 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:32.25 | saftsack | Wildcard TE411P Quad-Span togglable E1/T1 card (3.3V) |
| 21:32.26 | chapeaurouge | bweschke, well, not sure about the TDM2400P... it handles POTS lines |
| 21:32.35 | saftsack | Voice Modes: |
| 21:32.39 | saftsack | EuroISDN |
| 21:32.45 | chapeaurouge | bweschke, dont think it'll work for E1s |
| 21:32.47 | bweschke | isn't that what an rj11 is |
| 21:32.56 | bweschke | you can't plug an e1 into an rj11 |
| 21:32.59 | bweschke | ? |
| 21:33.26 | chapeaurouge | ? |
| 21:33.27 | saftsack | ? |
| 21:33.34 | chapeaurouge | lol |
| 21:33.45 | chapeaurouge | no you can't. |
| 21:34.02 | saftsack | i couldnt find an entry in the wiki for that topic |
| 21:34.08 | saftsack | do you know one? |
| 21:34.22 | bweschke | if you're getting an e1 from your provider and you actually want to break that out to 30 phones off your asterisk system, you'll need an E1 card.. and then you can do an additional E1 port (on a dual card) to an external channel bank.. OR you can plug in an TDM2400 series card |
| 21:34.48 | chapeaurouge | bweschke, got what ur saying now. |
| 21:35.14 | chapeaurouge | but you'll still need an interface to the E1, which the 2400 doesn't provide |
| 21:35.53 | bweschke | chapeaurouge: yes. absolutely correct. that's an FXO/FXS capable board only.. but a very cost effective replacement to a channel bank |
| 21:36.02 | LostFrog | And.. now, I get static. |
| 21:36.03 | bweschke | with a hardware echo can on board if you want that flavor |
| 21:36.06 | chapeaurouge | still 2000EUR with the mods... |
| 21:36.25 | *** join/#asterisk frenzy (n=frenzy@193.220.82.108) |
| 21:36.30 | chapeaurouge | got a quote today for 1FXO and 4FXS.. 1850EUR. |
| 21:36.31 | saftsack | do you know any webistes on which is talked from that professional solutions? |
| 21:36.45 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 21:36.56 | frenzy | hey alll |
| 21:37.15 | frenzy | can someone tell me what billing solutions are available for * |
| 21:37.19 | frenzy | besides AstBill |
| 21:38.03 | frenzy | ? |
| 21:39.43 | *** join/#asterisk [hC] (n=hardcore@209.153.195.139) |
| 21:39.53 | Dr-Linux | i have TDM card with 4 FXO ports, i wanna redirect all the PSTN calls to the SIP users extension, and body help me? |
| 21:40.34 | bweschke | Dr-Linux: have you configured you zaptel ports already on the server? |
| 21:43.19 | Dr-Linux | bweschke: well, i gonna congiured all activites by tomorrow, right now i came this area to get a quick guide that what to do, and i have some doubts that i wanna clear. |
| 21:43.44 | Dr-Linux | bweschke: can i /msg bweschke ? |
| 21:44.19 | bweschke | sure |
| 21:44.25 | Dr-Linux | Thanks |
| 21:45.30 | convey | what is the correct syntax in the extensions.conf for out bound calls with 1+ dialing, asterisk ver 1.2 |
| 21:47.00 | chapeaurouge | hmm... i have a stupid question... can i send free SMSes if I have a GSM modem connected to my * box (or ser) via SIP? |
| 21:47.02 | *** join/#asterisk [Latre] (n=latre@148.233.19.133) |
| 21:47.41 | azzie | really stupid. What * has to do with SMS pricing? |
| 21:47.47 | chapeaurouge | no idea |
| 21:48.09 | chapeaurouge | i told you it was stupid |
| 21:49.20 | [Latre] | hi people....i have problems with aah, can not make works that one extension recieve 2 or 3 calls at the same time......i want that the user listen beep beep when recive 2 calls at the same time....someone knows about this problem???? |
| 21:49.37 | L|NUX | can i send sms using asteirsk ? |
| 21:50.35 | Berkey | Any recomendations for did voip service that is unlimited usage? I like junction networks but they are 2.9 per min. |
| 21:51.12 | *** join/#asterisk darby_t (i=darby_t@dln70.neoplus.adsl.tpnet.pl) |
| 21:51.19 | bweschke | Berkey: your mileage will vary depending on your ISP connection. I've been happy with Broadvoice |
| 21:51.53 | DrPickles | i have a 402b card and i can't seem to get astrisk to recognise it |
| 21:51.59 | frenzy | i've noticed * 1.2 takes up more cpu.. |
| 21:52.08 | saitech | how can i make a stress test of my zaptel card and asterisk setup ? |
| 21:52.32 | frenzy | in the sense per call resource.. |
| 21:52.56 | Dr-Linux | what is pastebin link ? :S |
| 21:53.21 | DrPickles | is anyone familier with settin gup a digium 400 card? |
| 21:54.26 | DrPickles | i have an astrisk server that can call from sip to sip with voicemail and all the trim i just installed a 400 card |
| 21:56.37 | saitech | does anyone know how many zapcards/spans i could use withoug bad sound quality on a HP DL360 G4. (Dual 3Ghz Xeon, 2GB Ram, 800mHz FSB) |
| 21:57.32 | [Latre] | saitech: check possible problems with irqs |
| 21:57.39 | dudes | I've seem 280 some sip to zap calls on a Dual 3Ghz Xeon 1GB ram |
| 21:58.19 | dudes | and sound was fine |
| 21:58.27 | saitech | dudes really??? did you know what cpu load? |
| 21:58.39 | dudes | It wasn't very much |
| 21:58.42 | saitech | okay |
| 21:58.54 | saitech | do you know how to make a strestest on a zapcard ? |
| 21:59.03 | Berkey | bweschke I see the unlimited in state is only 9.95 do you use that plan? My sprint POTS are 20.75 plus $9 on top for caller ID |
| 22:00.03 | [hC] | Hey for residential people who have alarm systems that dial out in an emergency (mostly modem data i believe) is there a way to allow that to run via voip? ulaw comes to mind but i dont think its reliable ENOUGH. |
| 22:01.10 | saitech | Does anyone have a good idea, on how to make a decent stresstest on a server running with a Zapata Card? |
| 22:01.32 | Berkey | I have 5 POTS and one junction networks for roll over. The POTS are $150 by themself. If I put in another DSL With 700k up I should get 5 to 8 voip lines right? |
| 22:01.41 | test34 | I get a Syntax Error when I try to "make install" the zaptel driver (v1.2.0).. anybody else got this problem ? here's the exact error message-> http://pastebin.ca/30173 |
| 22:02.19 | test34 | Dr-Linux, http://pastebin.ca |
| 22:03.57 | puzzled | sigh this new setgroup & group_count stuff in 1.2 is confusing |
| 22:04.02 | saitech | Does anyone have a good idea, on how to make a decent stresstest on a server running with a Zapata Card? |
| 22:05.42 | puzzled | saitech: already seen your question... be a bit more patient or email the -users list |
| 22:06.30 | Dr-Linux | test34: thanks i got one, but the one you suggest is giving me problems |
| 22:06.36 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 22:08.59 | orlock | lilo: Hey, you dont have any actual control over individual channels/ops do you? |
| 22:10.27 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 22:14.56 | *** join/#asterisk santiago (n=santiago@63.245.87.62) |
| 22:16.47 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
| 22:17.16 | paryl | i don't get it... i compiled zaptel from cvs and i try to insert it, and i get "zaptel: version magic '2.6.12-9-686-smp SMP 686 gcc-4.0' should be '2.6.12-9-686-smp SMP 686 gcc-3.4'" |
| 22:17.21 | paryl | in dmesg |
| 22:17.50 | *** join/#asterisk bugz (n=will@69.15.174.114) |
| 22:17.52 | bugz | hola' |
| 22:18.17 | *** join/#asterisk abatista (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 22:18.18 | bugz | im not sure what broke but i cant get any sound from the pbx over the phone |
| 22:18.44 | bugz | when i let a phone ring to get into voicemail, it hangs right when it starts playing 'vm-theperson' |
| 22:18.51 | bugz | <PROTECTED> |
| 22:18.55 | bugz | right there, thats it |
| 22:18.56 | bugz | no nada |
| 22:19.08 | bugz | its strange, this has me wondering about the disk |
| 22:19.17 | bugz | ive re-hashed all the permissions in the filesystem |
| 22:19.20 | bugz | nothing |
| 22:19.45 | Nivex | bugz: does an echo test work? |
| 22:20.36 | bugz | hmm, cant really do that remotely |
| 22:20.48 | bugz | lemme try to set something up so i can get in through a DID |
| 22:22.47 | crash3m | isnt ulaw g729? |
| 22:23.07 | twisted[asteria] | uhhhhh |
| 22:23.13 | twisted[asteria] | no. |
| 22:23.17 | crash3m | g711? |
| 22:23.18 | bweschke | crash3m: no |
| 22:23.22 | bweschke | yes |
| 22:23.25 | twisted[asteria] | ulaw == g.711u |
| 22:23.30 | crash3m | thanks |
| 22:25.50 | *** join/#asterisk MattB2 (n=Mattb@mail.tricycleinc.com) |
| 22:25.57 | MattB2 | hi all |
| 22:26.03 | fa___ | where can I read about new dialplan managing in asterisk 1.2.0 ? |
| 22:26.27 | MattB2 | has ayone here got experience with SNOM 360s? just looking at professional phones for our office and want to get people's opinions on them |
| 22:26.59 | mog_work | they are all right |
| 22:27.00 | LostFrog | I've been playing with mine for 1.5 weeks, MattB2, and I like them. |
| 22:27.05 | LostFrog | Other than a few quirks. |
| 22:27.05 | mog_work | my screen crashes some times |
| 22:27.09 | MattB2 | :( |
| 22:27.10 | mog_work | but works pretty well |
| 22:27.15 | MattB2 | what other pro-level ip phones are recpommended? |
| 22:27.24 | mog_work | so dont judge it on that |
| 22:28.10 | bweschke | MattB2: I've got a snom360 that I hvae at a few clients... |
| 22:28.14 | bweschke | it works well for the most part |
| 22:28.31 | bweschke | my one complaint is that the UI on the phone is intimidating for someone not "tech savvy" |
| 22:28.47 | MattB2 | hmm.. we currentrly have a bunch of GXP-2000s, and we're fed up of problems with them |
| 22:28.50 | MattB2 | mainly crashing whilst on calls |
| 22:28.52 | MattB2 | and just ranomd hangups |
| 22:28.56 | LostFrog | One of the major things that I wish worked was distinctive ring/custom ringers. |
| 22:28.58 | MattB2 | so we need something very reliable |
| 22:29.05 | bweschke | nah. I've not seen the snom 360's do that at all |
| 22:30.00 | bweschke | However, now that the 601 is out and the firmware is expected shortly to do more than 8 device state presence at a time, I think I'm going to have a preference to put those out because it's just a little more friendly |
| 22:30.35 | LostFrog | bweschke: can it have 54 extensions? (BLFs) |
| 22:30.40 | MattB2 | 601? |
| 22:30.48 | LostFrog | MattB2: polycom |
| 22:31.05 | MattB2 | ah |
| 22:32.29 | bweschke | you're supposed to be able to have 3 sidecars off the 601... |
| 22:32.43 | bweschke | I dunno what that amounts to in total... it's definitely more than 20-30 I think |
| 22:32.48 | MattB2 | When equipped with up to three SoundPoint® IP Expansion Modules, the SoundPoint® IP 601 delivers the advanced call handling capabilities of a high-performance attendant console |
| 22:32.52 | MattB2 | (off the site) |
| 22:35.02 | bweschke | i've got a sidecar in the office here. one sec and i'll dig it out and count the buttons :) |
| 22:35.46 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 22:38.07 | MattB2 | so polycom are reliable then? |
| 22:38.17 | MattB2 | top of my list to investigate are polycom, cisco and snom |
| 22:38.22 | MattB2 | any others I should know about>? |
| 22:38.25 | bweschke | yes. very. |
| 22:38.45 | MattB2 | we have a polycom wireless conf phone - not IP though, but I'm very impressed with it |
| 22:38.47 | bweschke | Ciscos using the SIP firmware are very nice too, but usually, they are more than most clients want to spend |
| 22:38.56 | MattB2 | yeah |
| 22:39.00 | MattB2 | I have a thing against cisco :S |
| 22:39.07 | MattB2 | not sure what, just styarts ringing alarm bells in my head ;) |
| 22:39.13 | bweschke | I probably Polycom in more than 90% of the deployments I've done |
| 22:43.18 | Dr-Linux | if 2 fxo cards(4 port each) are connected to my server, it will be in zeptel.conf fxsks=1-8 |
| 22:43.30 | Dr-Linux | or fxoks=1-8 ? |
| 22:45.28 | *** join/#asterisk lorinc (n=ang@caracas-1419.adsl.interware.hu) |
| 22:46.45 | Dr-Linux | http://paste.uni.cc/8179 |
| 22:47.10 | twisted[asteria] | there is no such file as zeptel.conf |
| 22:47.17 | *** part/#asterisk frenzy (n=frenzy@193.220.82.108) |
| 22:47.49 | Dr-Linux | twisted[asteria]:sorry zaptel.conf |
| 22:48.02 | *** part/#asterisk DrPickles (n=strangeb@sitcomfurniture.net) |
| 22:48.20 | twisted[asteria] | ;) |
| 22:48.33 | twisted[asteria] | but the signalling is always opposite of what your card is |
| 22:48.38 | twisted[asteria] | so if it's fxo card, it's fxs signalling |
| 22:49.35 | Dr-Linux | twisted[asteria]: yes thats in zapata.conf signalling=fxs_ks for fxo cards |
| 22:49.46 | Dr-Linux | but what about zaptel.conf ? |
| 22:50.07 | twisted[asteria] | you're still specifying what signalling to use there too |
| 22:50.17 | twisted[asteria] | zaptel.conf is for the driver, zapata.conf is for asterisk |
| 22:52.17 | *** join/#asterisk phifli (n=mike@189.sub-70-197-169.myvzw.com) |
| 22:52.45 | phifli | how do i make sure bridged calls (that don't need transcoding) don't get relayed to each other? for CDR billing reasons? |
| 22:53.17 | bweschke | reinvite=no ? |
| 22:53.18 | Dr-Linux | twisted[asteria]: so if i have 2 fox(4 port each) in zaptel.conf will be (fxsks=1-8) ... and in zapata.conf (signalling=fxs_ks) |
| 22:53.24 | *** join/#asterisk trixter (i=trixter@65.172.209.246) |
| 22:53.25 | phifli | ah thanks :) |
| 22:53.29 | Dr-Linux | is that right? twisted[asteria] ? |
| 22:53.58 | bweschke | Dr-Linux: yes. twisted is right. |
| 22:54.02 | bweschke | http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf for more info |
| 22:54.50 | Dr-Linux | bweschke: to whome i send my email he will reply me? the one i sent you before? |
| 22:55.12 | MattB2 | does the polycom 601 support some kind of central address book? |
| 22:55.26 | MattB2 | just reading the user guide, doesn't suggest that it does |
| 22:55.28 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 22:55.59 | bweschke | it supports the reading of XML files for directories which can be built from a central address book |
| 22:56.18 | MattB2 | ah |
| 22:56.36 | MattB2 | guess you can force it to re-read the address book at any time remotely? or would that require a reboot? |
| 22:56.52 | *** join/#asterisk MindSpark (n=boo@81.10.32.22) |
| 22:56.57 | MindSpark | hi |
| 22:57.08 | MindSpark | how do I convert my wav files into gsm |
| 22:57.14 | bweschke | ummm... good question. the only time I've done it - I force a reboot. |
| 22:57.20 | MindSpark | ore record gsm files in the first place ? |
| 22:57.50 | MattB2 | ok guys, thanks for all your help. time to go and hassle my cfo ;) |
| 22:57.55 | MattB2 | bfn |
| 22:58.55 | nextime | anyone using native mp3 moh? |
| 22:59.01 | nextime | ( with cvs head ) |
| 22:59.08 | malverian[work] | Anyone written a holiday scheduling agi? |
| 22:59.24 | malverian[work] | Like one that would insert something into the astdb for scheduling holiday, then pull from it to determine if it's within that time range? |
| 22:59.56 | malverian[work] | So for example, the people can dial some extension to set the date/time for a holiday message to play, and it will automatically start/end based on the time they specify. |
| 23:00.16 | trixter | I must say that ITSPs that spam you then complain to your isp that you are bouncing emails back to them are kinda stupid because they are opening themselves up to potential liability once it all comes out.. do they really think that the provider wont investigate anything? |
| 23:01.26 | phifli | still says Releasing IAX2/NuFone and IAX2/OtherProvider |
| 23:01.36 | phifli | does this mean that I can't control it etc, or does it just mean that |
| 23:01.40 | phifli | it won't be using it anymorte? |
| 23:01.42 | MindSpark | does someone know how I get to record or convert gsm files ? |
| 23:01.45 | phifli | like following down context |
| 23:01.49 | MindSpark | I tried using sox without much success |
| 23:01.59 | phifli | MindSpark check out the examples, i think extensions 325 |
| 23:02.00 | phifli | it records |
| 23:02.02 | phifli | 235 |
| 23:02.15 | MindSpark | aight, thanks |
| 23:02.20 | *** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
| 23:02.29 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 23:02.38 | phifli | 205 im sorry |
| 23:02.47 | phifli | it records to /tmp/asterisk-recording:gsm |
| 23:02.51 | *** join/#asterisk Romik_ (n=romik_@1.fix.netvision.net.il) |
| 23:02.56 | phifli | and plays back after your done (push pound to stop) |
| 23:03.04 | phifli | Record(/tmp/asterisk-recording:gsm) |
| 23:03.26 | Romik_ | anyone can advice:i have 4 colors at rj-11 jack, black, red, green, yellow, which 2 of them i need to connect patch panel? |
| 23:03.46 | MindSpark | nice ! |
| 23:03.47 | MindSpark | thanks |
| 23:04.15 | trixter | normally red is ring of the primary green is tip black is ring of secondary yellow is tip (I think I got the yellow black correct) although for most stuff now ring/tip doesnt matter cuase devices use full wave bridge rectifiers which corerct polarity mismatch |
| 23:04.49 | phifli | i did record(file-%d:gsm) |
| 23:04.54 | phifli | so i could do multiple ones for my menus :) |
| 23:04.56 | phifli | just a tip |
| 23:05.06 | Romik_ | trixter: so which 2 of them i need to connect to patch pannel? |
| 23:05.15 | phifli | and i justw wrote down what each num was and renamed them later |
| 23:05.21 | trixter | red/green for line one black/yellow for line two assuming standard wiring |
| 23:05.26 | phifli | (had to record a decent bit of things) |
| 23:06.05 | Romik_ | trixter: thanks |
| 23:08.37 | Romik_ | trixter: just wandering , does it matter which way it should be in patch panel ? http://www.siemon.com/int/download/installation_instructions/us/66_S66M1-50_ii.pdf |
| 23:08.56 | Romik_ | trixter: i mean for red/green color cables. |
| 23:09.38 | Dr-Linux | RHEL AS 3 will recognize the TDM card? or it will need kernel simultineous problem etc ? |
| 23:14.26 | Dr-Linux | anybody answer ? |
| 23:17.32 | *** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net) |
| 23:17.49 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 23:17.53 | kink0 | buenas noches |
| 23:18.24 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 23:18.49 | kink0 | I am intested about install GSM with Asterisk |
| 23:18.57 | Dr-Linux | twisted[asteria]: and what about in zapata.conf >> group=1 << |
| 23:19.13 | phifli | GSM comes with asterisk :) |
| 23:19.17 | phifli | just make sure u put allow=gsm |
| 23:19.33 | kink0 | phifli, sorry, I speak about mobile GSM , not codec |
| 23:20.33 | phifli | oh :) |
| 23:20.35 | phifli | whats zombie gsm? |
| 23:21.59 | kink0 | zombie ? |
| 23:22.23 | test34 | can I use zaptel driver v1.0.10 with asterisk v1.2.0 because zaptel driver v1.2.0 doesnt install succesfully ? |
| 23:22.30 | phifli | Yeah |
| 23:22.41 | phifli | why doesn't it install successfully? |
| 23:22.42 | Dr-Linux | phifli |
| 23:22.48 | Dr-Linux | RHEL AS 3 will recognize the TDM card? or it will need kernel simultineous problem etc ? |
| 23:22.49 | phifli | if its compile issues or whatnot, possibly i can help |
| 23:23.06 | phifli | YOu meana custom recompiled kernel? |
| 23:23.36 | Dr-Linux | phifli: i installed RHEL AS 3 |
| 23:23.38 | phifli | i honestly do not know, but recompiling a custom kernel is easy if you have RH just get the SRPM and undo it and use rpmbuild -bb but you can also load the current config from /boot/config-current_kernel_here and then mod it |
| 23:23.47 | phifli | therefore you wont be changing much if your worried about losing some things |
| 23:23.57 | Dr-Linux | and if i plugged the TDM card to the slot, it will recognized or not, |
| 23:24.05 | Dr-Linux | i have installed zaptel package as well |
| 23:24.05 | phifli | not sure, type dmesg |
| 23:24.21 | test34 | phifli, make works without any problems, but when I do "make install" I get a syntax error |
| 23:24.27 | phifli | if you want e to check it out just setup a normal user and install screen (yum install screen) and type su phifli - then screen |
| 23:24.29 | phifli | and ill screen -x |
| 23:24.32 | phifli | (we'll be on same screen) |
| 23:24.36 | phifli | then you su root and i can help you with it |
| 23:24.45 | phifli | syntax error ah try |
| 23:24.46 | phifli | gmake |
| 23:24.48 | Dr-Linux | phifli: normally what it does? |
| 23:24.54 | phifli | well |
| 23:25.04 | phifli | gmake is a little diff, RH may have upgraded to newer versions |
| 23:25.13 | phifli | prob the syntax error on something that seems completly normal heh |
| 23:25.19 | phifli | just autoconf/automake or whatnot |
| 23:25.21 | test34 | phifli, do I change make for gmake in a config file ? |
| 23:25.24 | phifli | possibly dif vers |
| 23:25.30 | phifli | no |
| 23:25.32 | phifli | just type gmake |
| 23:25.34 | phifli | instead of make |
| 23:25.34 | phifli | :) |
| 23:25.51 | test34 | phifli, ok thanks Im gonna try right now |
| 23:26.09 | phifli | np |
| 23:26.12 | phifli | let me know if you hve any trouble |
| 23:26.28 | Dr-Linux | phifli: what RHEL version your recommend for * successfull setup in both >> RHEL 3 or RHEL 4 ? |
| 23:26.32 | phifli | i dont wanna login anyones boxes alone but if ppl use screen (dont wanna be blamed for anything ;)) ill help you and also you can see what i do exactly |
| 23:26.35 | phifli | hmm |
| 23:26.37 | test34 | phifli, Ive had this problem for 3 days.. I hope this works;) |
| 23:26.38 | phifli | i use fedora core 4 |
| 23:26.44 | phifli | fedora is an offspring of redhat |
| 23:26.48 | phifli | they went diff ways |
| 23:26.51 | test34 | I use slackware |
| 23:26.59 | phifli | ie: redhat has XFree and fedora has Xoreg |
| 23:27.00 | phifli | Xorg |
| 23:27.02 | phifli | i love slack :) |
| 23:27.08 | Dr-Linux | :S |
| 23:27.11 | phifli | too bad lots of companies want the "mainstream" bs |
| 23:27.12 | test34 | it uses xorg |
| 23:27.18 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:27.19 | phifli | I had slack with kernel 2.0.23 :) |
| 23:27.26 | phifli | first time i installed linux |
| 23:27.34 | test34 | yeah I used to have 2.0 too |
| 23:27.41 | phifli | :) |
| 23:27.47 | Dr-Linux | phifli: can you see my pastbin question ? |
| 23:27.55 | phifli | then you've been playing around with it for 10 yrs ;) |
| 23:28.01 | phifli | give 1-2 more actually |
| 23:28.06 | phifli | Dr-Linux: link? |
| 23:28.10 | Dr-Linux | http://paste.uni.cc/8179 |
| 23:28.13 | phifli | I only pay attention 20% of the time |
| 23:28.14 | test34 | like 7 years ago or something;) |
| 23:28.15 | phifli | doing lots of coding |
| 23:28.15 | phifli | sec |
| 23:28.30 | phifli | Oh yeah your right iwas playing with SCO before |
| 23:28.33 | phifli | and HP-UX |
| 23:28.36 | Dr-Linux | phifli rocks ;) |
| 23:28.38 | test34 | phifli, I get the same error with gmake |
| 23:28.47 | *** join/#asterisk digime (n=digime@203.117.181.40) |
| 23:28.51 | phifli | test34: possibly really an error, paste bin that part of the Makefile |
| 23:28.52 | phifli | ill check it out |
| 23:28.56 | phifli | i forgot ppl here use pastbin |
| 23:28.57 | phifli | pastebin |
| 23:29.05 | *** part/#asterisk digime (n=digime@203.117.181.40) |
| 23:29.09 | phifli | or msg or email for all that matter mike-ast@limittech.net |
| 23:29.15 | test34 | phifli, http://pastebin.ca/30173 |
| 23:29.24 | Dr-Linux | phifli: i would like a reply in breif way, if thats possible for you |
| 23:29.32 | phifli | sure sec |
| 23:29.34 | phifli | let me check em both out |
| 23:30.22 | test34 | phifli, oh, gmake is a symlink to make |
| 23:30.45 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 23:31.01 | phifli | ah lol them damn cheaters |
| 23:31.03 | phifli | ok well |
| 23:31.10 | phifli | hoenstly that makefile looks all garbled up |
| 23:31.19 | phifli | sec |
| 23:31.29 | phifli | let me read through the if and loops or whatevr |
| 23:32.42 | phifli | Dr-Linux: I don't have experience with using Digium hardware unfortunately ;( i plan on getting a card soon once i get a rack at a NOC although your going to have to make peer and user settings for both servers in .. oh wait you have a link... Ok, do you VPN? |
| 23:32.42 | test34 | phifli, let me post an easier to read version |
| 23:32.42 | Dr-Linux | phifli: :S you are checking my pastbin :S? |
| 23:32.55 | phifli | Dr-linux: yeah i multitask like a mofo :( |
| 23:33.05 | phifli | i have 3 monitors here with 3 vmware sessions and im coding like 4 things :/ |
| 23:33.15 | phifli | talk about ADHD |
| 23:33.28 | Dr-Linux | phifli: i have only one SERVER |
| 23:33.35 | phifli | ok |
| 23:33.44 | phifli | Are you using any VPN to secure your communication or not too worried? |
| 23:33.52 | Dr-Linux | that Located in US |
| 23:34.01 | Dr-Linux | right now i 'm not using VPN |
| 23:34.11 | Dr-Linux | but for other VOIP solution i'm using VPN |
| 23:34.12 | phifli | an easy solution is openvpn (they have windiows clients, therefore each user in Pak can install, or you could make your router if its wrt54g run it or you could make a gateway box) |
| 23:34.17 | phifli | ok |
| 23:34.18 | Dr-Linux | thats not a problem i can use for it as well |
| 23:34.23 | phifli | then you can just .... |
| 23:34.23 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-14-207.cybersurf.com) |
| 23:34.28 | phifli | want me to paste back to the pastebin how? |
| 23:35.13 | Dr-Linux | phifli: please answer my question acording to my setup in an email (sshah@i2cinc.com) or pastebin |
| 23:35.17 | phifli | u wanna use IAX or SIP Dr-Linux? |
| 23:35.17 | Dr-Linux | if possible for you |
| 23:35.22 | Dr-Linux | SIP |
| 23:35.23 | phifli | ill email |
| 23:35.43 | test34 | phifli, http://pastebin.ca/30410 |
| 23:35.52 | phifli | im no master but i can get what u need done :-P |
| 23:35.52 | Dr-Linux | my all users in pak will use SIP softphones are ip phones |
| 23:35.57 | phifli | cool |
| 23:36.02 | phifli | and here how many ip phones? |
| 23:36.05 | phifli | and how many total? |
| 23:36.18 | Dr-Linux | 100 softphones in PAK |
| 23:36.19 | phifli | too bad theres no quick frontend thats out yet for extensions in mysql :) |
| 23:36.23 | phifli | i just started getting into that today |
| 23:36.24 | phifli | its awesome |
| 23:36.33 | phifli | you bought any licenses for g729 or anything:: |
| 23:36.34 | phifli | ? |
| 23:36.37 | test34 | phifli, I think $(MODCONF) is not set, could that be the problem ? |
| 23:36.38 | phifli | or are you the BW master :) |
| 23:36.39 | Dr-Linux | and 10 IP phones |
| 23:36.39 | phifli | bandwidth |
| 23:36.56 | phifli | sec it loaded just now |
| 23:37.04 | Dr-Linux | phifli: not for g729 yet |
| 23:37.19 | Dr-Linux | bandwidth is 1 MB DSL only for PBX |
| 23:37.27 | phifli | isnt MODCONF defined by something else? |
| 23:37.28 | test34 | if [ -n "" ]; should be if [ -f $(MODCONF) ]; |
| 23:37.30 | phifli | ,maybe an include from Makefile? |
| 23:37.42 | phifli | ah i see |
| 23:38.04 | phifli | do |
| 23:38.08 | phifli | export MODCONF=/etc/modprobe.conf |
| 23:38.13 | phifli | and make |
| 23:38.21 | phifli | you may have to set statically |
| 23:39.00 | phifli | ie in the makefile add MODCONF to the top |
| 23:39.00 | test34 | phifli, thanks a bunch, I think it works |
| 23:39.13 | phifli | MODCONF just means itll detect the card on boot when its processing all hardware |
| 23:39.18 | phifli | good :) |
| 23:39.24 | phifli | 1mb dsl hmm |
| 23:39.37 | phifli | dr-linux: how many users you expect to use this server at once? |
| 23:40.12 | phifli | ill make a lil script to generate all those SIP users etc and show you the script too |
| 23:40.46 | phifli | If you would like to do a user database you may want to check out the mysql thing or do a one time modify to put everyones names for caller id |
| 23:40.53 | phifli | (if your calling from IP->IP) |
| 23:42.02 | tzafrir_laptop | note that modprobe.conf is for kernel 2.4 only |
| 23:42.08 | azzie | Can Cisco 79xx dial a number without picking up the phone first ? |
| 23:42.23 | test34 | tzafrir_laptop, I use 2.4 |
| 23:42.33 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost, -- if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
| 23:42.59 | test34 | why use 2.6 when 2.4 works fine;) |
| 23:43.13 | tzafrir_laptop | because 2.6 has hotplug support |
| 23:43.32 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool139-17.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 23:43.38 | phifli | 2.6 also has a lot of new features |
| 23:43.42 | test34 | tzafrir_laptop, I know that 2.6 has support for a bunch of other stuff, but I dont need it |
| 23:44.14 | phifli | test34: You'd be surprised :) |
| 23:44.17 | phifli | they redid a lot of code |
| 23:44.25 | phifli | security issues in 2.4 were disgusting |
| 23:44.30 | phifli | <- security freak |
| 23:44.44 | test34 | phifli, I tried compiling 2.6.. it's about twice the size |
| 23:44.50 | phifli | well |
| 23:44.53 | test34 | for the same thing |
| 23:44.57 | phifli | dont compile everything into the kernel :) |
| 23:45.06 | phifli | turn some into modules, or turn off the things you won't ever use/need |
| 23:45.15 | test34 | I did |
| 23:45.15 | phifli | lots of ppl don't optimize the configuration for it before they build |
| 23:45.20 | phifli | weird |
| 23:45.23 | phifli | but the kernel being somewhat bigger |
| 23:45.26 | phifli | does it kill your performance? |
| 23:45.45 | phifli | 2.6 really actually handles certain system critical things in a completely different manner especially SMP |
| 23:45.53 | test34 | 1mb for 2.4 and 1.9mb for 2.6 with pretty much the same options |
| 23:45.53 | phifli | i just want kqueue on linux ;) |
| 23:46.07 | test34 | I dont need smp |
| 23:46.14 | phifli | that was an example... |
| 23:46.21 | phifli | hotpluggins i sa very nice feature |
| 23:46.47 | tzafrir_laptop | test34, what Makefile? |
| 23:47.20 | test34 | tzafrir_laptop, that was zaptel's driver Makefile |
| 23:47.24 | Dr-Linux | <phifli> dr-linux: how many users you expect to use this server at once? >> 8 to 10 users |
| 23:48.27 | test34 | there is hotplug in 2.4 btw |
| 23:48.36 | phifli | Dr-Linux: need mailboxes too? |
| 23:48.59 | Dr-Linux | phifli: yes sir |
| 23:49.00 | phifli | test43: just working on 50+ linux boxes for a company ... they kinda get to choose :) |
| 23:49.18 | phifli | when your gone like i am now from my last job before i started my own corp..... the next guy needs to be able to jump on |
| 23:49.24 | phifli | so slack was out of the question |
| 23:49.33 | phifli | and 2.6 was in due to we had all SMP boxes |
| 23:49.43 | phifli | fiber etc |
| 23:49.44 | phifli | hehe |
| 23:49.53 | phifli | Dr-Linux: k |
| 23:50.32 | phifli | the other office, dynamic or static ip? |
| 23:50.35 | test34 | phifli, yeah I dont say that someone might not need it, but I don't;) |
| 23:50.45 | phifli | YOu don't have to tell me it ill comment it for you |
| 23:50.46 | Dr-Linux | phifli: static IP |
| 23:50.51 | phifli | k |
| 23:51.01 | phifli | hmm |
| 23:51.02 | phifli | so |
| 23:51.09 | phifli | you have an office with 100 users using one static? |
| 23:51.10 | phifli | or several? |
| 23:51.17 | phifli | they'll have to swap up ports on their stuff =P |
| 23:51.25 | phifli | unless TCP/IP sip is supported yet lol |
| 23:51.28 | test34 | since I messed up some hardware by upgrading it I think twice about upgrading... sometimes newer isnt better |
| 23:51.34 | phifli | NAT for that is outragous |
| 23:52.21 | phifli | that's one thing i like about VPN... each 100 user could have 10.0.0.1-100 and no address issues |
| 23:52.21 | Dr-Linux | yes one office in pakistan with 100 users having static IPS |
| 23:52.21 | phifli | but if you need to NAT they need to choose source ports wisely |
| 23:52.22 | phifli | nice |
| 23:52.22 | phifli | k |
| 23:52.28 | syzygyBSD | I have to recompile the kernel for HDLC support right? |
| 23:52.34 | Dr-Linux | ok |
| 23:52.50 | phifli | HDLC? |
| 23:53.01 | shido6 | yes |
| 23:53.03 | shido6 | no |
| 23:53.18 | syzygyBSD | HDLC = data |
| 23:53.32 | shido6 | check your kernel for the modules |
| 23:53.41 | shido6 | make menu |
| 23:53.50 | shido6 | config |
| 23:53.51 | syzygyBSD | going to have a hybrid voice/data setup |
| 23:54.05 | shido6 | make menuconfig |
| 23:55.02 | syzygyBSD | I don't even have the source downloaded yet, I need to find out what version I want to use, is 2.6 stable with asterisk? |
| 23:55.18 | trixter | heard there are tdmoe problems with 2.6 |
| 23:55.49 | trixter | and with 2.6 you need to make sure you select 'desktop' in the kernel for timing reasons (server is 2Hz desktop is 1000Hz and you need 1000Hz resolution for asterisk to properly work) |
| 23:55.55 | trixter | other than that afaik yes it is stable |
| 23:57.38 | syzygyBSD | good to know, I was just using a precompiled 2.6 kernel |
| 23:58.27 | trixter | that might be 1000Hz then, but it depends on who compiled it, and I dont know how each distro does it |
| 23:59.05 | kippi | where are digium ? |
| 23:59.31 | file[laptop] | Huntsville, Alabama. |
| 23:59.41 | mog_work | yup |
| 23:59.51 | file[laptop] | that's where mog_work is OMG! |